=== release 0.10.36 ===
2012-02-20 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
releasing 0.10.36, "Better"
2012-02-20 23:19:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/ca.po:
* po/id.po:
po: update translations
2012-02-17 15:08:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiodecoder.c:
* win32/common/libgstaudio.def:
docs: add new audio base class API to docs and .def file
2012-01-30 15:55:26 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: only send new data immediately if there are no queued messages
Even if watch->messages->length is 0 there may still be some
data from a message that was only written partially at the
previous attempt stored in watch->write_data, so check for
that as well. We don't want to write data into the middle
of another message, which could happen when there wasn't
enough bandwidth.
https://bugzilla.gnome.org/show_bug.cgi?id=669039
2012-02-16 12:19:20 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
audiodecoder: add some properties to tweak baseclass behaviour
... so subclass can also rely upon never being bothered with some NULL buffer
it can't do any interesting with, or with any data before it received
any format configuration (and setup properly).
2012-02-16 12:18:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
audioencoder: add some properties to tweak baseclass behaviour
... so subclass can also rely upon never being bothered with less data
than it desires or with some NULL buffer it can't do any interesting with.
2012-02-16 12:15:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: assert some more that subclass parsed frame has proper len
2012-02-14 19:23:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: try harder to obtain a duration if we don't get one right away
If we don't get a duration right away, set the pipeline to playing
and sleep a bit, then try again. This is ugly, but the least worst
we can do right now. The alternative would be to make parsers etc.
return some bogus duration estimate even after only having pushed
a single frame, for example.
Fixes discoverer showing 0 durations for some mp3 and aac files
(e.g. soweto-adts.aac).
2012-02-05 13:55:40 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.35.3 pre-release
2012-02-01 15:28:45 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: fix granpos interpolation violating max keyframe distance
In case many packets fit on a page, we may not see a granpos for
a while, and granpos interpolation can wrap the 'frames since last
keyframe' part of the granpos, generating a granpos which is smaller
than what it should be.
This is fixed by detecting keyframe packets (at least for Theora),
and updating the last keyframe granpos from this.
This may still be generating potentially wrong granpos for streams
which have a Theora like granpos (keyframes, a max keyframe distance
and a count of frames since last keyframe), and which allow implicit
granules on packets. For these streams, a custom keyframe detection
routine should be plugged into their GstOggStream mapper.
https://bugzilla.gnome.org/show_bug.cgi?id=669164
2012-02-01 16:46:13 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/vorbis/gstvorbisparse.c:
vorbisparse: pedantically recognize undefined headers too
2012-02-01 16:32:24 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/vorbis/gstvorbisparse.c:
vorbisparse: fix header detection
It was matching non header packets.
This fixes various leaks, where buffers would be pushed onto a headers
list, but never popped.
Might also fix corruption as those buffers were dropped from the output
silently...
https://bugzilla.gnome.org/show_bug.cgi?id=669167
2012-01-23 09:28:18 -0800 David Schleef <ds@schleef.org>
* gst-libs/gst/interfaces/propertyprobe.c:
propertyprobe: fix documentation
2012-01-18 14:58:08 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: do not try to deactivate an inactive group
A group may have failed to activate due to an error (for instance,
having set the URI to a non existent location in about-to-finish).
https://bugzilla.gnome.org/show_bug.cgi?id=666395
2012-01-17 16:05:41 +0200 Anssi Hannula <anssi.hannula@iki.fi>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: fix state change stall on PAUSED->READY->PAUSED
After a PAUSED->READY change the sink pads are currently not set to
blocking state. When the element is set back to PAUSED, the change will
be done asynchronously, but as the _pad_blocked_cb() callback is now not
called, the state change never completes.
Fix that by setting the sink pads to blocking state on a PAUSED->READY
change, which ensures that the _pad_blocked_cb() is called when needed
on any future READY->PAUSED change. The sink pads are already put to
blocking state on NULL->READY change, so this behavior is consistent.
Fixes bug #668097.
2012-01-19 16:40:22 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: avoid unlikely NULL dereference
2012-01-19 16:35:54 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/videoscale/vs_fill_borders.c:
videoscale: prevent implicit upgrade to integer type and sign extension
2012-01-19 16:35:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tools/gst-discoverer.c:
gst-discoverer: remove extraneous variable
2012-01-19 16:32:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: verify linking to overlay element
2012-01-19 16:32:05 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: avoid finding sink in NULL bin in corner case
2012-01-19 16:29:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: add missing break
2012-01-17 18:19:30 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: initialize variable
... to help out challenged compiler.
2012-01-16 11:43:25 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/alsa/gstalsasink.c:
alsasink: fix high sample rates being rejected
An ALSA sink may select a different rate (as we use the _set_rate_near
API, which is not guaranteed to set the exact target rate).
The rest of the code seems to already handle this well, as output
from a 88200 Hz file seems to have the correct pitch when selecting
a 96 kHz rate.
2012-01-16 11:40:47 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/alsa/gstalsasink.c:
alsasink: fix rate match message mistaking error code for sample rate
2012-01-13 16:57:15 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>
* Android.mk:
Android, Add explicit path for zlib
This change fixes building gst-libs/gst/tag/ code with
the Android buildsystem.
2012-01-13 14:50:49 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>
* ext/vorbis/gstvorbisdec.c:
Fix wrong access to undefined struct member
For the USE_TREMOLO case, GstVorbisDec doesn't have
a vb member. Besides, Tremolo's vorbis_dsp_synthesis()
expects a vorbis_dsp_state to be passed as first
argument. Not a vorbis_block.
2012-01-13 14:47:13 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>
* ext/vorbis/gstvorbisdec.c:
Fix TREMELO -> TREMOLO typo
2012-01-12 16:24:01 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraparse.c:
theoraparse: fix array leak
2012-01-12 14:26:05 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix structure leak
I hit the 'misc' one, but let's also make sure the topology
one get freed as well, though I do not know if this can happen
twice.
2012-01-11 20:47:00 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>
* gst-libs/gst/video/Makefile.am:
Add missing DEFAULT_INCLUDES on androgenizer call
Fix building of the libgstvideo module on Android by adding the
missing and needed $(DEFAULT_INCLUDES) to CFLAGS for the
androgenizer call on gst-libs/gst/video/Makefile.am
Before this change, building was failing due to gst-plugins-base/
and gst-plugins-base/gst-libs/gst/video being left out of the
include path.
2012-01-11 16:17:42 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix push mode chain leak
When I first implemented push mode seeking, I removed the chain
freeing there as it could be used later. The current code does not
seem to do that though, so I'm restoring the previous freeing,
which plugs the leak while apparently not reintroducing use of
freed data with chained and normal files, both with gst-launch
playbin2 and Totem.
2012-01-11 12:52:17 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
discoverer: fix leaks caused by some base class dtors not being called
2012-01-11 12:16:28 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix caps and discoverer object ref leaks
2012-01-11 11:55:59 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: add a few consts where appropriate
2012-01-11 11:55:36 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix pad leak
2012-01-10 18:27:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: use GST_TYPE_TAG_LIST for tag lists
They may not be structures in 0.11/1.0.
2012-01-10 18:07:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix potential tag list leaks
Not that I have ever seen these in practice, but if they
can't happen we may just as well just assign the new tag
list. Merge properly to be on the safe side, and also
avoid a useless tag list copy in the normal case where
there is no tag list yet.
2012-01-10 17:48:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix potential caps leak
in last else chunk.
2012-01-10 16:57:04 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: fix tag list leak
2012-01-10 16:51:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix pad leak
2012-01-10 16:14:29 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix hang on small truncated files
A first hang was happening when trying to locate a page backwards,
where we'd sync forever on the same page.
With that fixed, a second hang would happen after preparing an EOS
event, but with no chain created yet to send it to, the pipeline
would stay idle forever.
An element error is now emitted for this case.
2012-01-09 12:31:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplay-enum.h:
playback: document DEINTERLACE flag
2011-12-16 15:27:24 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: assume live stream if byte size cannot be determined
This prevents trying to seek and failing, then ending up unable
to stream because we can't get back at the headers.
A more robust way would be to find a good place to reinject the
headers when a seek fails, but I can't seem to get this to work.
2012-01-07 20:12:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: make hostname lookup more thread-safe
Don't write IP number string to return into a static
array which is shared amongst all threads (note: of
course a copy is returned).
https://bugzilla.gnome.org/show_bug.cgi?id=666711
2012-01-07 19:39:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: make is_subtitle_caps thread-safe
2011-11-01 17:57:59 +0100 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/tag/tags.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/encoding/gstsmartencoder.c:
* gst/playback/gstplaysink.c:
* tools/gst-discoverer.c:
Fix various unlikely, but still potential memoryleaks in error code paths
https://bugzilla.gnome.org/show_bug.cgi?id=667311
2011-10-22 16:41:23 +0200 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/app/gstappsrc.c:
appsrc: implement get_caps vfunc
This allows downstream elements to query what caps are available.
https://bugzilla.gnome.org/show_bug.cgi?id=667312
2012-01-05 12:23:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tools/gst-discoverer.c:
tools: avoid unportable vararg macro construct in gst-discoverer
https://bugzilla.gnome.org/show_bug.cgi?id=667306
2012-01-01 20:44:08 +0100 Idar Tollefsen <itollefs@cisco.com>
* configure.ac:
build: Run platform check for platform specific configuration.
2011-10-12 11:28:10 +0200 Pascal Buhler <pabuhler@cisco.com>
* gst-libs/gst/rtp/gstrtcpbuffer.c:
rtcpbuffer: prevent overflow of 16bit header length.
RTCP header can be (2^16 + 1) * 4 bytes long, so when validating a bogus
packet it was possible to get a 16bit overflow resulting in a length of 0.
This would put the gst_rtcp_buffer_validate_data function in a endless loop.
https://bugzilla.gnome.org/show_bug.cgi?id=667313
2011-09-24 14:05:42 +0200 Havard Graff <havard.graff@tandberg.com>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: keep the calculation fixed-point
https://bugzilla.gnome.org/show_bug.cgi?id=667315
2011-08-04 11:30:05 +0200 Idar Tollefsen <itollefs@cisco.com>
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttimeoverlay.c:
pango: changes includes from brackets to quotes for local files
https://bugzilla.gnome.org/show_bug.cgi?id=667316
2012-01-04 19:39:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 63d592e to cb5da59
2012-01-03 11:04:23 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: force fallback buffer_alloc when other pad not available
... to avoid unnecessary spurious errors (upon e.g. shutdown).
If a real error is applicable in this unusual circumstance (missing other pad),
other (STREAM_LOCK protected) call paths can take care of that.
2012-01-03 11:02:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: avoid crashing when operating on released pad
2011-12-27 14:37:26 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* ext/ogg/gstoggmux.c:
oggmux: fix leak when initializing pads
Pads are initialized twice: when requesting pads and when
initializing collectpads. Avoid double initialization by
checking if collectpads are still going to be initialized when
creating request pads.
2011-12-23 22:51:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: fix template caps creation on big endian systems
2011-12-23 22:24:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: fix writing of Exif tag payloads <= 4 bytes
When the payload for an Exif tag is less than or equal to 4 bytes,
the data is simply put into the offset field. Fix writing these
kinds of payloads on big endian systems (and possibly also on
little endian systems). The caller will have already formatted
the bytes in memory according to the writer's endianness, so just
write out the bytes as they are in this case. Fixes tags unit test
on big endian systems.
2011-12-22 16:54:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: add a few more debug statements
2011-12-22 16:53:49 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
audiodecoder: tweak documentation
2011-12-22 07:53:39 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Keep compatibility with our old generated xmp
We used to add a trailing \n to the end of generated xmp packets.
Windows viewer was unhappy with it and we fixed it in
96d2120c2bb0b29e1849098198f5fbef81939cdd
The problem is that this caused xmp generated before this fix
to not be recognized and parsed anymore. This patch makes it
recognize xmp with the trailing \n and without, fixing the
regression. Also adds tests for it.
2011-12-14 16:34:39 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/video/video-blend.c:
gstvideo: fix a RGB ordering mixup in colorspace conversion code
2011-12-20 12:42:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
audiodecoder: set a non-zero default maximum tolerated errors
Whereas the previous default 0 was backwards compatible in that it lead
to erroring out immediately upon any error, elements that are really
ported and using the base class error macro can be assumed to intend to
improve behaviour rather than maintaining the old one. So, make it easy
on those and any future one and tolerate some errors by default, as intended.
Fixes #666579.
2011-12-15 11:01:01 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: do not include \0 in size passed to g_convert
When using g_convert, we should only pass the length
of the string content (without the \0) as g_convert will
only parse the real contents when changing formats. Including
the \0 causes it to add another \0, increasing the string
size when not needed.
For example, when writting a North geo location ref entry, that should
be a string with a single N letter, it would write:
"N\0\0", causing the string to have size 3, instead of 2 as expected.
In our case, we can pass -1 and let g_convert calculate the strlen as
we don't use the length anywhere else.
This fixes jifmux's tests on gst-plugins-bad.
2011-10-03 14:51:56 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: tweak chain topology description
... to also properly indicate chain's endpad if no elements are in the
chain (due to the endpad being a raw demuxer pad, or one setup without
decoders since uridecodebin or higher up decided not to need those).
2011-12-13 12:55:45 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: fix late buffer leak
2011-12-12 11:54:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/glib-compat-private.h:
glib-compat: Add license boilerplate for LGPL
2011-12-10 02:08:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/audio-enumtypes.c:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
0.10.35.2 pre-release
2011-12-10 01:36:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/LINGUAS:
* po/cs.po:
* po/eo.po:
* po/es.po:
* po/gl.po:
* po/lv.po:
* po/sr.po:
po: update translations
2011-12-09 15:39:12 +0000 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Add latest header file to spec file
2011-12-09 01:31:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: only typefind text with a BOM as text/utf16 or text/utf32
We added the utf typefinder because the mp3 typefinder was a tad
overzealous when it came to typefinding things as mp3, and replaced
it with even more overzealous utf16/32 typefinders.
Fixes unit test.
2011-12-07 18:45:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video-overlay-composition.c:
* gst-libs/gst/video/video-overlay-composition.h:
video: make composition_blend() return a boolean
Not that anyone will ever check that, and it's not clear what
they're supposed to do if it fails, but at least it's there.
2011-12-07 18:31:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video-overlay-composition.c:
* gst-libs/gst/video/video-overlay-composition.h:
docs: add new API to docs
2011-12-07 17:57:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video-overlay-composition.c:
* gst-libs/gst/video/video-overlay-composition.h:
* tests/check/libs/video.c:
* win32/common/libgstvideo.def:
video: add seqnum getters for overlay compositions and rectangles
API: gst_video_overlay_composition_get_seqnum()
API: gst_video_overlay_rectangle_get_seqnum()
2011-11-23 15:45:57 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* gst-libs/gst/video/video.c:
video: support any type of video in _parse_caps
Slight change in semantics for convenience. Shouldn't cause any
problems since this function is usually only used on pre-filtered
caps and not random caps, and it's hard to imagine a situation
where someone would want to rely on the previous behaviour.
2011-12-06 21:57:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: don't leak previous buffer when shutting down
Implement stop vfunc after port to basetransform, so we
can clean up properly. Fixes make elements/videorate.valgrind
2011-12-06 20:30:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/video.c:
tests: fix calculation of last pixel offset in video unit test
And check the right buffer (pix2) in one case.
2011-12-06 15:01:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/fft/Makefile.am:
examples: fix build of fft example
Should link against our own libgstfft-0.10.
2011-12-06 14:55:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: fix leak in gst_video_format_new_template_caps()
g_value_reset() is not the same as g_value_unset()
2011-11-23 15:43:46 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: add suport for hardware accelerated videos
Don't plug converters for non-raw video.
2011-12-05 15:48:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video-overlay-composition.c:
video: don't use deprecated GStaticMutex with newer glib versions
2011-12-05 15:34:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/Makefile.am:
examples: dist fft sub-directory
2011-11-28 10:05:50 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* ext/pango/gsttextoverlay.c:
textoverlay: unpremultiply text image
The GstVideoOverlayComposition only supports unpremultiplied ARGB
(for now anyway, support for pre-multiplied alpha is planned.)
2011-11-23 12:49:02 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: Attach OverlayComposition to buffers when needed
Add video/x-surface support in the caps
We should then attach it whenever the sink supports it, but this
is working for the time being
2011-11-18 13:22:52 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: Make the text_image data a buffer
This way we won't free data that would be attached to some buffer.
2011-11-18 11:04:47 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* ext/pango/gsttextoverlay.c:
textoverlay: Sync the caps with the new supported formats
Thanks to the use of the new video composition library, we gain support to
more colospaces and formats, let's state it.
2011-11-16 17:54:43 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: Make use of the new video blending utility
2011-11-25 16:46:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/video.c:
tests: add basic unit test for video overlay composition and rectangles
2011-11-12 14:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video-overlay-composition.c:
* gst-libs/gst/video/video-overlay-composition.h:
* win32/common/libgstvideo.def:
video: add video overlay composition API for subtitles
Basic API to attach overlay rectangles to buffers,
or blend them directly onto raw video buffers.
To be used primarily for things like subtitles or
logo overlays, not meant to replace videomixer.
Allows us to associate subtitle overlays with
non-raw video surface buffers, so that subtitles
are not lost and can instead be rendered later
when those surfaces are displayed or converted,
whilst re-using all the existing overlay plugins
and not having to teach them about our special
video surfaces. Could also have been made part
of the surface buffer abstraction of course, but
a secondary goal was to consolidate the blending
code for raw video into libgstvideo, and this
kind of API allows us to do both in a way that's
minimally invasive to existing elements, and at
the same time is fairly intuitive.
More features and extensions like the ability to
pass the source data or text/markup directly will
be added later.
https://bugzilla.gnome.org/show_bug.cgi?id=665080
API: gst_video_buffer_get_overlay_composition()
API: gst_video_buffer_set_overlay_composition()
API: gst_video_overlay_composition_new()
API: gst_video_overlay_composition_add_rectangle()
API: gst_video_overlay_composition_n_rectangles()
API: gst_video_overlay_composition_get_rectangle()
API: gst_video_overlay_composition_make_writable()
API: gst_video_overlay_composition_copy()
API: gst_video_overlay_composition_ref()
API: gst_video_overlay_composition_unref()
API: gst_video_overlay_composition_blend()
API: gst_video_overlay_rectangle_new_argb()
API: gst_video_overlay_rectangle_get_pixels_argb()
API: gst_video_overlay_rectangle_get_pixels_unscaled_argb()
API: gst_video_overlay_rectangle_get_render_rectangle()
API: gst_video_overlay_rectangle_set_render_rectangle()
API: gst_video_overlay_rectangle_copy()
API: gst_video_overlay_rectangle_ref()
API: gst_video_overlay_rectangle_unref()
2011-11-23 00:31:18 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video-blend.h:
video: hide private video-blend.[ch] from gobject-introspection
And remove unused fields from helper structure.
2011-11-15 18:00:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/videoblendorc-dist.c:
* gst-libs/gst/video/videoblendorc-dist.h:
video: add fallbacks for compilation without orc
2011-10-17 17:25:11 +0200 Thibault Saunier <thibault.saunier@collabora.com>
* gst-libs/gst/video/.gitignore:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video-blend.c:
* gst-libs/gst/video/video-blend.h:
* gst-libs/gst/video/videoblendorc.orc:
video: add some internal helper functions for image blending
This could be improved if we decide we don't need it to
be this generic/flexible.
2011-12-05 09:38:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/xoverlay.c:
xoverlay: Fix mistakes in the sample code
Fixes bug #665430.
2011-12-04 20:50:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsamixer.c:
* ext/ogg/gstoggdemux.c:
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gststreamsynchronizer.c:
* gst/tcp/gstmultifdsink.c:
Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
2011-12-04 20:38:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
alsamixer: use GRectMutext instead of GStaticRecMutex with newer glib versions
2011-12-04 20:21:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
alsamixer: embed static mutexes into the mixer structure
instead of allocating them dynamically
2011-12-04 17:02:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/encoding/encoding.c:
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/seek/jsseek.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
* tests/icles/stress-playbin.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
* tools/gst-discoverer.c:
tools, tests: g_thread_init() is deprecated in glib master
It's not needed any longer.
2011-12-04 16:43:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/pango/gsttextoverlay.c:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/glib-compat-private.h:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/video/convertframe.c:
* gst/encoding/gststreamcombiner.c:
* gst/encoding/gststreamsplitter.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysinkconvertbin.c:
* gst/playback/gststreamsynchronizer.c:
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Work around deprecated thread API in glib master
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
Replace g_thread_create() with g_thread_try_new().
2011-12-04 15:23:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/xmpwriter.c:
xmpwriter: update for thread API deprecations in glib master
2011-12-04 13:43:06 +0100 Stefan Sauer <ensonic@users.sf.net>
* tests/examples/fft/Makefile.am:
fft-example: re-add Makefile.am
2011-12-02 23:35:50 +0100 Stefan Sauer <ensonic@users.sf.net>
* configure.ac:
configure: trim trailing whitespace
2011-12-02 23:34:47 +0100 Stefan Sauer <ensonic@users.sf.net>
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/fft/.gitignore:
* tests/examples/fft/fftrange.c:
tests: add a test for fft result value-ranges
Add a small example that uses ffts of various types and parameters and check the
result value ranges.
2011-09-13 21:10:43 +0200 Piotr Fusik <fox@scene.pl>
* docs/design/design-audiosinks.txt:
* docs/design/design-decodebin.txt:
* docs/design/design-encoding.txt:
* docs/design/design-orc-integration.txt:
* docs/design/draft-keyframe-force.txt:
* docs/design/draft-va.txt:
* ext/alsa/gstalsamixer.c:
* ext/libvisual/visual.c:
* ext/ogg/README:
* ext/ogg/gstoggdemux.c:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraparse.c:
* ext/vorbis/gstvorbisdec.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/interfaces/navigation.c:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
* gst-libs/gst/pbutils/descriptions.c:
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-target.h:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtsprange.c:
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/gstxmptag.c:
* gst-libs/gst/tag/id3v2.3.0.txt:
* gst-libs/gst/tag/id3v2.4.0-frames.txt:
* gst-libs/gst/tag/id3v2.4.0-structure.txt:
* gst/adder/gstadder.c:
* gst/audioconvert/audioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c:
* gst/encoding/gststreamsplitter.c:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
* gst/ffmpegcolorspace/mem.c:
* gst/playback/README:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcp.c:
* gst/typefind/gsttypefindfunctions.c:
* gst/videotestsrc/gstvideotestsrc.c:
* m4/freetype2.m4:
* sys/v4l/v4lmjpegsrc_calls.c:
* sys/v4l/videodev_mjpeg.h:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
* tests/check/elements/adder.c:
* tests/check/elements/audioresample.c:
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
* tests/examples/encoding/encoding.c:
various: typo fixes
Fix typos in code and docs. Fixes. #658984
2011-12-01 11:59:17 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: be more graceful in the clipfunction
Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in
0.10 and sending such events in special elements like adder and tee was outvoted
on last attempt, be graceful to the misbehaviour instead.
2011-12-01 01:22:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/audioresample.c:
tests: fix caps leak in audioresample tests
2011-12-01 01:07:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/pipelines/basetime.c:
tests: fix memory leak in basetime test
2011-11-30 23:58:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: tone down debug message about file URIs with spaces
Complain a bit less loudly about URIs that have not been
escaped properly.
2011-11-30 23:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
* ext/alsa/gstalsasrc.h:
Revert "alsasrc: Improve timestamp accuracy"
This reverts commit 0b774e0b7cf7a8ef1780fb6100228ca6e8ca8bcf.
2011-11-30 23:15:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
Revert "alsasrc: Fix some compilation errors"
This reverts commit 2b84f5bd74ddb50f7832917ea8b4dd38d005631b.
2011-11-30 23:15:12 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
Revert "alsa: Remove unused but set variable"
This reverts commit e9aed7f31c7e9e415f733e147140ce3ef2f57a61.
2011-11-30 23:15:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
* ext/alsa/gstalsasrc.h:
Revert "alsasrc: fail gracefully when ALSA does not give timestamps"
This reverts commit c7282a5718c7f31f84fb31b2c38fab0f9a38e2b0.
2011-11-30 23:14:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
Revert "alsasrc: handle the case where the drivers don't supply timestamps"
This reverts commit 8154b69112cdc4830cd6002ec6c1f2917d30437b.
2011-11-28 10:55:39 +0100 Stefan Sauer <ensonic@google.com>
* ext/alsa/gstalsasrc.c:
Revert "alsasrc: style fix"
This reverts commit f70ca6d4cbfd2b672dcc7215814bf6b39ce2c3f8.
2011-11-30 14:25:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Don't send undefined NEWSEGMENT events to the internal elements
This happens when the internal elements are added before any NEWSEGMENT
event arrived and in that case we shouldn't send a NEWSEGMENT event
to the internal elements at all. They will get the NEWSEGMENT event
from upstream later.
2011-11-29 14:15:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Fix decoder-sink compatibility check for raw audio/video formats
If the sink supports raw audio/video, we first check
if the decoder could output any raw audio/video format
and assume it is compatible with the sink then. We don't
do a complete compatibility check here if converters
are plugged between the decoder and the sink because
the converters will convert between raw formats and
even if the decoder format is not supported by the decoder
a converter will convert it.
We assume here that the converters can convert between
any raw format.
Fixes bug #665120.
2011-11-29 09:11:21 +0100 Alessandro Decina <alessandro.d@gmail.com>
* ext/ogg/gstoggdemux.c:
oggdemux: fix compiler warning
2011-11-29 08:49:53 +0100 Alessandro Decina <alessandro.d@gmail.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
* win32/common/libgstvideo.def:
libgstvideo: minor fixes to key unit events
Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
optional, update libgstvideo.def and fix docs a bit.
API: gst_video_event_new_upstream_force_key_unit
API: gst_video_event_new_downstream_force_key_unit
API: gst_video_event_is_force_key_unit
API: gst_video_event_parse_upstream_force_key_unit
API: gst_video_event_parse_downstream_force_key_unit
https://bugzilla.gnome.org/show_bug.cgi?id=607742
2011-06-05 01:49:38 +0200 Andoni Morales Alastruey <ylatuya@gmail.com>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
libgstvideo: Add force key unit events
2011-11-28 20:11:09 +0100 Philippe Normand <philn@igalia.com>
* gst-libs/gst/fft/gstfft.h:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.h:
fft: Bracket public headers
This is especially needed if the gstfftw library is used from C++
code.
Fixes #665074
2011-11-28 20:10:18 +0100 Philippe Normand <phil@base-art.net>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Fix compiler warning
2011-11-28 19:03:50 +0100 Alexey Fisher <bug-track@fisher-privat.net>
* gst/typefind/gsttypefindfunctions.c:
typefind: fix build error
fix build errors:
gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized]
gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized]
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
2011-11-28 19:06:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Fix stupid mistake in last commit
2011-11-28 19:03:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Only return the converter caps if we actually have raw caps
Fixes bug #664818 (hopefully).
2011-11-28 17:59:32 +0100 Kipp Cannon <kcannon@cita.utoronto.ca>
* gst/audioresample/gstaudioresample.c:
audioresample: Don't emit DISCONT buffers if no discontinuity happened
audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output. Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.
Fixes bug #665004.
2011-09-30 20:00:50 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/typefind/Makefile.am:
* gst/typefind/gsttypefindfunctions.c:
typefind: typefind UTF-16 and UTF-32
This avoids the MP3 typefinder from getting the highest score
every time it thinks there's something it might possibly be
able to parse.
https://bugzilla.gnome.org/show_bug.cgi?id=607619
2011-11-28 13:27:29 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoradec.h:
Revert "theoradec: move the QoS logic to libgstvideo"
This reverts commit 149a4ce390a78e21309b210f7daba9db5d42afe6.
*grumble* I managed to merge something I did not mean to.
2011-11-28 13:26:53 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
* win32/common/libgstvideo.def:
Revert "libgstvideo: add a new API to handle QoS events and dropping logic"
This reverts commit eb03323fb683e06ed8e7f557037f13252f150c25.
*grumble* I managed to merge something I did not mean to.
2011-11-28 12:51:22 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/tag/gsttagdemux.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/encoding/gstencodebin.c:
* gst/encoding/gstsmartencoder.c:
* gst/encoding/gststreamcombiner.c:
* gst/encoding/gststreamsplitter.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gststreamselector.c:
* gst/playback/gststreamsynchronizer.c:
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gsturidecodebin.c:
* gst/subparse/gstssaparse.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversrc.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/elements/audiorate.c:
* tests/check/elements/decodebin.c:
* tests/check/elements/decodebin2.c:
* tests/check/elements/playbin.c:
* tests/check/elements/playbin2-compressed.c:
* tests/check/elements/playbin2.c:
* tests/check/elements/videoscale.c:
various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-09-07 16:04:14 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoradec.h:
theoradec: move the QoS logic to libgstvideo
https://bugzilla.gnome.org/show_bug.cgi?id=658241
2011-09-05 13:56:05 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
* win32/common/libgstvideo.def:
libgstvideo: add a new API to handle QoS events and dropping logic
https://bugzilla.gnome.org/show_bug.cgi?id=658241
2011-11-28 11:30:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
audioencoder: elaborate some documentation
2011-11-28 11:28:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
audiodecoder: add some documentation
2011-11-21 14:26:54 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: really discard NULL decoded frame altogether
... including any timestamp, rather than having that one influence base_ts.
2011-11-28 10:55:39 +0100 Stefan Sauer <ensonic@google.com>
* ext/alsa/gstalsasrc.c:
alsasrc: style fix
Use timestamp==0 instead of mixing it with !timestamp style checks.
2011-11-28 09:12:37 +0100 Stefan Sauer <ensonic@users.sf.net>
* ext/alsa/gstalsasrc.c:
alsasrc: handle the case where the drivers don't supply timestamps
If highres-timestamp is 0, try lowres and if that fails fallback to system clock
timestamps.
2011-11-01 15:21:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: set collectpads2 not to wait on sparse streams
https://bugzilla.gnome.org/show_bug.cgi?id=663174
2011-11-25 15:35:39 +0100 Josep Torra <n770galaxy@gmail.com>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: make identiy silent
2011-11-25 13:11:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/vorbis/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
audio: remove unstable API guards from the audio decoder and encoder base classes
2011-11-25 12:58:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
docs: mention explicitly that playbin2 signals are emitted from a streaming thread
2011-11-25 11:11:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Set the multiqueue limits to the playing limits after overrun too
We don't expect any new pads anymore and prerolling is finished now.
2011-11-25 11:08:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits
After preroll the multiqueue limits are still set to the preroll
limits if use-buffering is set to TRUE. In that case we only want
time limits on the multiqueue if upstream is seekable.
2011-11-08 13:55:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: fix prerolling for low bitrate streams from hlsdemux
Such streams were detected as seekable, as the query on the typefind
element was testing the m3u8 file listing the actual streams, and
not going through the demuxer(s).
We now check for seekability for each multiqueue following a demuxer,
so the query will flow through the elements which might prevent seeking.
https://bugzilla.gnome.org/show_bug.cgi?id=647769
2011-10-24 11:46:05 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: minor cleanup
2011-09-27 16:45:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/riff/riff-ids.h:
libgstriff: add a couple tags that need skipping
Found in a sample in the wild, appears to be ID3 tag.
https://bugzilla.gnome.org/show_bug.cgi?id=660249
2011-11-24 14:41:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Rename ARG_ enums to PROP_
This is more consistent with other code and these are
properties anyway, not arguments
2011-11-24 14:29:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: Add property to force an output framerate
API: GstVideoRate:force-fps
Changing the framerate during playback is not possible
with a capsfilter downstream if upstream is not using
gst_pad_alloc_buffer(). In that case there's no way in
0.10 to signal to videorate that the preferred framerate
has changed.
This new property will force the output framerate to
a specific value and can be changed during playback.
2011-11-24 12:38:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps
We might need to add converters and worked in passthrough mode before.
2011-11-24 12:37:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Override acceptcaps function for the two ghostpads
The ghostpad acceptcaps functions are not valid in this case because
we don't only accept the caps accepted by the target but could also
insert converters. Fixes bug #663892.
2011-11-24 11:34:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
playsinkaudioconvert: use-volume and use-converters are no construct-only properties anymore
Fixes bug #663893.
2011-10-22 20:29:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: skip the second bisection when possible
If we already saw the keyframes that we need to find,
we do not need to bisect to find them.
This will always be the case for streams with audio only,
where each frame acts as a keyframe, but will occasionally
also happen for streams with video.
https://bugzilla.gnome.org/show_bug.cgi?id=662475
2011-10-22 20:20:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: improve push time seeking
Various tweaks to improve convergence, in particular for
the worst case, which is now cut in about half.
https://bugzilla.gnome.org/show_bug.cgi?id=662475
2011-10-21 19:38:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: gather some more stats about bisection
https://bugzilla.gnome.org/show_bug.cgi?id=662475
2011-11-23 16:09:13 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/vorbis/gstvorbisenc.c:
vorbisenc: do not accept 256 channels, 255 is the max vorbis supports
2011-11-22 13:29:10 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: extract opus comments if available
2011-11-22 13:15:33 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: recognize opus headers from data, not packet count
Opus streams outside of Ogg may not have headers, and oggstream
may be used by oggmux to mux an Opus stream which does not come
from Ogg - thus without headers.
Determining headerness by packet count would strip the first two
packets from such an Opus stream, leading to a very small amount
of audio being clipped at the beginning of the stream.
2011-11-22 13:01:35 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: add some more debug info when determining start time
2011-11-22 12:55:56 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: fix opus duration calculation
2011-11-22 12:00:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: early out on headers when determining packet duration
2011-11-21 17:03:21 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggstream: account for opus pre-skip in granpos/time mapping
2011-11-22 10:04:12 +0100 René Stadler <rene.stadler@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: avoid removing children from bin twice
GstBin base class removes children in dispose, so we need to do the same.
2011-11-19 16:06:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.c:
ogg: add opus support
2011-11-16 19:00:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisenc.c:
vorbisenc: reset tag setter interface when appropriate
2011-11-16 19:00:30 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: invalidate format info when setup negotiation failed
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
2011-11-15 13:29:31 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: accept dropped buffers before we know the format
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
2011-11-14 12:45:31 +0100 Robert Swain <robert.swain@gmail.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
audio: Remove some unused variables
2011-08-30 18:27:09 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst-libs/gst/rtp/gstrtcpbuffer.h:
rtcpbuffer: Add feedback message types from RFC 5104
These are Codec Control messages (CCM)
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-10-19 16:30:27 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: improve reverse playback
... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.
Fixes #661983.
2011-11-13 13:18:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiodecoder.c:
audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class
API: GST_AUDIO_INFO_IS_VALID
2011-11-12 15:51:52 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
tests: require Gtk+ 3.0 for examples and Gtk-based test apps
The Gtk+ dependency is entirely optional, we're just not
supporting Gtk+ 2.x any longer.
2011-11-07 17:36:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/Makefile.am:
audio: fix order in LIBADD
Local libs must come first.
2011-11-11 13:32:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: fix visualisations again
Make caps writable before merging other caps into them.
2011-11-10 15:55:31 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: make unsigned properties unsigned, not signed
2011-11-09 00:36:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
* configure.ac:
configure: suppress warnings about unused variables if debugging system is disabled in core
https://bugzilla.gnome.org/show_bug.cgi?id=662952
2011-10-27 14:48:52 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: continue processing text when silent
This prevents playback wegding when text buffers are
left to pile up.
https://bugzilla.gnome.org/show_bug.cgi?id=662829
2011-11-08 00:16:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstaudio.def:
win32: update .def file for new audiosink API
API: gst_base_audio_sink_get_alignment_threshold()
API: gst_base_audio_sink_set_alignment_threshold()
API: gst_base_audio_sink_get_discont_wait()
API: gst_base_audio_sink_set_discont_wait()
2011-11-07 23:41:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
examples: sprinkle GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS in seek test utility
https://bugzilla.gnome.org/show_bug.cgi?id=630497
2011-11-07 23:05:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
* gst-libs/gst/audio/gstaudioiec61937.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/video/video.c:
docs: fix up some Since: markers
2011-11-04 10:34:27 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: fix speed level failure test
It was testing the opposite of what it thought it was.
https://bugzilla.gnome.org/show_bug.cgi?id=663390
2011-11-04 10:57:40 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: make logically static const data just so
https://bugzilla.gnome.org/show_bug.cgi?id=663391
2011-11-04 10:58:15 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: use th_packet_iskeyframe instead of peeking at bits
https://bugzilla.gnome.org/show_bug.cgi?id=663391
2011-11-04 10:59:00 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: trivial comment typos fixes
https://bugzilla.gnome.org/show_bug.cgi?id=663391
2011-11-04 10:59:12 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: warn when trying to set an ignored obsolete property
https://bugzilla.gnome.org/show_bug.cgi?id=663391
2011-11-04 11:10:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: refuse to get to READY if the encoder was disabled
https://bugzilla.gnome.org/show_bug.cgi?id=663391
2011-10-18 17:58:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: survive skeleton finding length behind our backs in push mode
In push mode, we determine duration by doing a seek to the end of the
stream. However, a skeleton stream with an index will cause the duration
to be known already, and we end up never setting the push_time_duration
variable which we use to know duration has been determined.
https://bugzilla.gnome.org/show_bug.cgi?id=662049
2011-10-05 15:29:54 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* tests/check/gst-plugins-base.supp:
valgrind: add ALSA leaks fixed by snd_config_update_free_global
If they go when calling snd_config_update_free_global, they're
not really bug leaks, but more like intentional ones we don't
want to get told about.
https://bugzilla.gnome.org/show_bug.cgi?id=615342
2011-05-02 13:05:28 +0300 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
baseaudiosink: make discont-wait configurable
Now we can configure how much time to wait before deciding that a
discont has happened.
Also, adds getter and setter to allow derived implementations to set
this value upon construction.
Suggestions and several improvements by Havard Graff.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-07 11:31:47 +0100 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime
A common problem for audio-playback is that the timestamps might not
be completely linear. This is specially common when doing streaming over
a network, where you can have jittery and/or bursty packettransmission,
which again will often be reflected on the buffertimestamps.
Now, the current implementation have a threshold that says how far the
buffertimestamp is allowed o drift from the ideal aligned time in the
ringbuffer. This was an instant reaction, and ment that if one buffer
arrived with a timestamp that would breach the drift-tolerance, a resync
would take place, and the result would be an audible gap for the
listener.
The annoying thing would be that in the case of a "timestamp-outlier",
you would first resync one way, say +100ms, and then, if the next
timestamp was "back on track", you would end up resyncing the other way
(-100ms) So in fact, when you had only one buffer with slightly off
timestamping, you would end up with *two* audible gaps. This is the
problem this patch addresses.
The way to "fix" this problem with the previous implementation, would
have been to increase the "drift-tolerance" to a value that was greater
than the largest timestamp-outlier one would normally expect. The big
problem with this approach, however, is that it will allow normal
operations with a huge offset timestamp vs running-time, which is
detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
basically means that lip-sync can easily end up being off by that much.
This patch will basically start a timer when the first breach of
drift-tolerance is detected. If any following timestamp for the next n
nanoseconds gets "back on track" within the threshold, it has basically
eliminated the effect of an outlier, and the timer is stopped. If,
however, all timestamps within this time-limit are breaching the
threshold, we are probably facing a more permanent offset in the
timestamps, and a resync is allowed to happen.
So basically this patch offers something as rare as both higher
accuracy, it terms of allowing smaller drift-tolerances, as well as much
smoother, less glitchy playback!
Commit message and improvments by Havard Graff.
Fixes bug #640859.
2011-11-07 11:18:34 +0100 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: rename some variables
2011-05-21 16:16:42 +0300 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: use gst_util_uint64_scale_int when appropriate
It's probably safer this way.
2011-05-21 15:49:20 +0300 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
baseaudiosink: split drift-tolerance into alignment-threshold
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
2011-05-21 16:02:36 +0300 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: trivial comment fixes
Some found by Havard Graff.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-04 10:37:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Use gst_caps_merge() instead of gst_caps_union()
This keeps the caps order and is more efficient.
2011-11-04 10:36:51 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Use gst_caps_merge() instead of gst_caps_union()
This keeps the caps order and is more efficient.
2011-11-03 21:35:38 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>
* gst-libs/gst/tag/Makefile.am:
Add missing default include paths to androgenizer call
Fixes building tag/ with Android's NDK
2011-11-03 14:10:31 +0200 Mart Raudsepp <mart.raudsepp@collabora.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Post all source pads in stream-topology messages as "element-srcpad" values
This allows us to easily get ahold of all pads on a stream-topology message, including
pre-decoder ones, while "pad" only gives us access to the raw pads (as used by discoverer).
2011-10-20 13:04:52 +0300 Mart Raudsepp <mart.raudsepp@collabora.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Use existing "caps" quark for one of the structure sets
2011-11-03 10:07:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Don't add identity multiple times
2011-10-19 14:13:39 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsink: send flush start/stop event when we switch elements
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-19 14:13:30 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkconvertbin.c:
* gst/playback/gstplaysinkconvertbin.h:
playsink: re-add identity where appropriate
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-19 14:12:01 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
playsink: lock the new {set,get}_property functions
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 23:14:54 +0000 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Be more consistent with ghostpad targets
Set up targets on READY->PAUSED state change to passthrough by
default. This prevents the targets from being unset on the
first run, while the 'raw' variable would mean that some
target is set.
2011-10-17 22:41:49 +0000 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: No need to remove the identity
The identity element should be handled by the GstBin's cleanup,
removing it on the remove_elements function might remove it
too soon, as this function can be called directly from playsink
2011-10-17 22:41:11 +0000 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Adding some debug messages
Adds a couple debug messages and some g_assert to make debugging
easier
2011-10-17 22:02:03 +0000 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst/playback/gstplaysinkvideoconvert.c:
playsink-videoconvert: Fix warning on build
Remove unused variable
2011-10-17 21:05:30 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkaudioconvert.h:
* gst/playback/gstplaysinkconvertbin.c:
* gst/playback/gstplaysinkconvertbin.h:
* gst/playback/gstplaysinkvideoconvert.c:
* gst/playback/gstplaysinkvideoconvert.h:
playsink: handle after-the-fact changes in converters/volume booleans
The playsink was nastily poking a boolean in the structure.
Make those booleans properties, so we are told when they change,
and rebuild the conversion bin when they do.
Some cleanup to go with it too.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 18:43:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsink: handle NULL cached caps in getcaps
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 18:06:00 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsink: consider both passthrough and converter caps in getcaps
Since we can switch between both modes.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 17:54:27 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
* gst/playback/gstplaysinkconvertbin.h:
playsink: cache inner converter bin caps
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 17:26:48 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsink: keep both raw and non raw pipelines at all times
and switch between them as needed.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 17:29:50 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsink: only compare against the media type we expect
ie, audio/x-raw- for audio, video/x-raw- for video.
Add a trailing - to be more specific. I doubt there's anything
like audio/x-rawhide or something, but you never know.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 16:55:30 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkaudioconvert.h:
* gst/playback/gstplaysinkconvertbin.c:
* gst/playback/gstplaysinkconvertbin.h:
* gst/playback/gstplaysinkvideoconvert.c:
* gst/playback/gstplaysinkvideoconvert.h:
playsink: refactor the converter bins since they are almost identical
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-17 13:00:05 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkaudioconvert.h:
* gst/playback/gstplaysinkvideoconvert.c:
* gst/playback/gstplaysinkvideoconvert.h:
playsink: fix passthrough mode (hopefully)
The code was doing counterintuitive rewiring of pads when the
bin did not contain any elements. We now add an identity element
in that case, which makes it simpler, and should fix the AC3
passthrough mode when using pulseaudio (but I don't see the bug
here so can't test).
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-10-07 11:16:44 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkvideoconvert.c:
playsink: handle NULL ghost pad target
For the src pad anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
2011-11-03 09:56:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
Revert "playsinkaudioconvert: Fix warning when there is no target pad yet"
This reverts commit f35c51c14915729f0fdf2b348f351ea7e81027cc.
Better patch coming soon.
2011-10-28 10:07:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Remove obsolete #include
2011-11-02 23:33:18 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/design/draft-subtitle-overlays.txt:
docs: add draft for subtitle overlays to design docs
Main purpose is to provide a generic way to make subtitles work on
top of non-raw video (vaapi, vdpau, etc.).
2011-11-02 15:31:11 -0400 Colin Walters <walters@verbum.org>
* common:
* configure.ac:
configure: Allow setting GLIB_EXTRA_CFLAGS
Similar to gstreamer commit bb2020b1e794210cf7d44c6626122f611016a620
2011-10-30 20:00:47 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: don't use soon-to-be-deprecated gst_filter_run()
2011-10-28 13:58:47 +0200 Mersad Jelacic <mersad@axis.com>
* gst-libs/gst/audio/gstaudiosink.c:
audiosink: avoid deadlocking audioringbuffer thread
... when it goes into wait for ringbuffer starting just after such
having been signalled.
Fixes #661738.
2011-04-26 22:20:29 +0200 Philip Jägenstedt <philipj@opera.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: extract SOF marker in jpeg typefinder
The SOF types are defined by http://www.w3.org/Graphics/JPEG/itu-t81.pdf
This is needed to make sure that we plug a jpeg decoder that
can handle the type of JPEG we have (e.g. lossless JPEG)
https://bugzilla.gnome.org/show_bug.cgi?id=556648
2009-08-10 01:48:29 +0000 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: port to gstcollectpads2
2011-10-27 23:39:31 +1100 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/Makefile.am:
build: Fix build for moved volume subdir
2011-10-27 09:51:46 +0200 Stefan Sauer <ensonic@users.sf.net>
* Makefile.am:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/audio/.gitignore:
* tests/examples/audio/Makefile.am:
* tests/examples/audio/volume.c:
* tests/examples/volume/.gitignore:
* tests/examples/volume/Makefile.am:
* tests/examples/volume/volume.c:
volume: move volume example to audio
2011-10-27 09:42:36 +0200 Stefan Sauer <ensonic@users.sf.net>
* tests/examples/audio/Makefile.am:
audio examples. fix the makefile
2011-10-27 09:33:55 +0200 Stefan Sauer <ensonic@users.sf.net>
* tests/examples/volume/volume.c:
volume: make global vars static
2011-10-27 09:33:01 +0200 Stefan Sauer <ensonic@users.sf.net>
* tests/examples/audio/.gitignore:
* tests/examples/audio/Makefile.am:
* tests/examples/audio/audiomix.c:
audiomix: add a simple audiomix example
2011-10-25 20:04:06 +1100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaysinkaudioconvert.c:
playsinkaudioconvert: Fix warning when there is no target pad yet
2011-10-13 11:34:49 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Link elements before testing if they can reach the READY state
This is made possible by filtering errors. This is required to let
harware accelerated element query the video context. The video context
is used to determine if the HW is capable, and thus if the element is
supported or not.
Fixes bug #662330.
2011-10-21 21:57:17 +0200 René Stadler <rene.stadler@collabora.co.uk>
* gst/playback/gstplaybasebin.c:
playbasebin: remove avoidable call to gst_object_set_name
2011-10-21 21:41:03 +0200 René Stadler <rene.stadler@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: remove avoidable call to gst_object_set_name
2011-10-21 21:39:01 +0200 René Stadler <rene.stadler@collabora.co.uk>
* gst/audioconvert/Makefile.am:
* gst/audioconvert/channelmixtest.c:
audioconvert: bury dead test program
2011-10-20 10:13:46 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>
* Android.mk:
Disable ext/vorbis for the android ndk build
It currently makes the build fail. Idea is to enable
it back again once its building problems get sorted
out.
2011-10-19 19:44:06 +0200 René Stadler <rene.stadler@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: fix leaks of pad templates and internal proxy pads
2011-10-19 19:37:07 +0200 René Stadler <rene.stadler@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: fix leak of element reference through pad block
If the pad block never happens because there is no data flow at all, the
callback is never fired and the reference is never released. This causes a
reference cycle between the pad and element, so valgrind is not very vocal
about it (memory is still reachable).
2011-10-18 21:42:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: having gather queue contents implies some draining is in order
... which ensures e.g. processing and sending last fragment of reverse playback
downstream at EOS.
2011-10-19 15:28:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: do not try to read past the buffer array
https://bugzilla.gnome.org/show_bug.cgi?id=662108
2011-10-18 21:40:54 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: only finish header packet frame if received in-stream
... rather than scaring audiodecoder with a frame extracted from caps.
Fixes #662108 (partially).
2011-10-19 10:41:31 +0200 Stefan Sauer <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: make it more clean that "synchronous" props are not for avsync
2011-10-19 00:32:13 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: fix unused variable compiler warning if debugging in core is disabled
https://bugzilla.gnome.org/show_bug.cgi?id=660150
2011-10-18 13:00:29 +0200 René Stadler <rene.stadler@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: fix event unref in (rare) error case
2011-10-07 17:41:32 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: fire drained signal where appropriate
This will allow playbin2 to send its about-to-finish signal.
Taken out (apparently by mistake) by the EOS rewrite in july.
https://bugzilla.gnome.org/show_bug.cgi?id=661202
2011-10-16 11:32:41 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not retry seeking indefinitely
https://bugzilla.gnome.org/show_bug.cgi?id=661897
2011-10-10 13:11:59 +0200 Brian Cameron <brian.cameron@oracle.com>
* gst/videotestsrc/Makefile.am:
videotestsrc: fix LDADD missing GST_LIBS
2011-10-09 21:19:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisenc.h:
vorbisenc: only push header buffers following initial events
2011-10-09 16:48:18 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
audioencoder: fix compile warning
2011-10-08 20:17:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/check/pipelines/vorbisenc.c:
tests: vorbisenc: adjust discontinuity checking to audioencoder behaviour
... which still detects gaps and marks DISCONT, depending on configuration,
but may come up with somewhat different timestamps when crossing the gap.
2011-10-08 20:16:04 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/check/pipelines/vorbisdec.c:
tests: vorbisdec: properly configure audiodecoder when requiring perfect ts
2011-10-08 20:14:27 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/check/elements/vorbisdec.c:
tests: vorbisdec: remove empty header buffer check
... as empty buffers are discarded, and header buffers are now
also optionally retrieved from caps anyway.
2011-10-08 20:13:11 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: only resync to upstream upon discont in perfect ts mode
... as documented, where discont is marked here if tolerance has been
exceeded.
2011-10-08 20:11:22 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: fix timestamp tolerance handling
2011-10-08 20:09:09 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: handle empty input by discarding
2011-10-07 14:52:33 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
vorbisdec: port to audiodecoder
2011-10-07 14:33:04 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: make upstream queries MT-safe
2011-10-07 14:32:33 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: make upstream queries and events MT-safe
2011-10-05 15:43:35 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisenc.h:
vorbisenc: port to audioencoder
2011-10-06 18:21:29 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* tests/check/elements/audiotestsrc.c:
tests: actually test what we said we would
All tests were testing the default sine wave
https://bugzilla.gnome.org/show_bug.cgi?id=661106
2011-10-06 18:20:32 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: add missing break
And make violet noise usable
https://bugzilla.gnome.org/show_bug.cgi?id=661105
2011-10-06 15:38:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkvideoconvert.c:
playsink: fix caps negotiation through the new convenience bins
The bins' getcaps was bypassing the inner elements, and thus
failing to account for the caps transformations they allow,
which caused YUV video pipelines to fail with ximagesink, which
does not support YUV, even though the convenience bin includes
a colorspace converter for just this purpose.
https://bugzilla.gnome.org/show_bug.cgi?id=660816
2011-10-06 11:53:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: fix mismatch between video/ and video/x-dvd-subpicture
The new code was checking for a prefix, and would find video/
first. Check in two passes, first checking for a perfect match,
and falling back to a prefix check if nothing was found.
https://bugzilla.gnome.org/show_bug.cgi?id=657261
2011-10-04 21:17:37 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst/encoding/gstencodebin.c:
encodebin: Re-enable parsers
Re-enable parsers in encodebin to allow more passthrough scenarios
to work. Specially the ones that require changing 'stream formats'.
i.e. h264 in mkv to mpegts.
2011-10-05 12:45:19 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Add audio- and text-sink props
2011-10-04 23:09:42 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
auditestsrc: indent fix
2011-10-04 16:22:55 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Add video-sink property
The video-sink property allows manual specification via g_object_set ()
of the video sink element to be used.
2011-10-03 15:20:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Minor cleanup of decoder-sink compatibility checking code
2011-09-30 12:29:34 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* gst/playback/gstplaybin2.c:
playbin2: Make sure that the decoders we plug are compatible with the fixed sink
The fact that a decoder is not compatible with the fixed sink
is currently happenning in the case where we have hardware accelerated
video decoders on the system (especially vaapi elements that are actually plugged),
and the user is providing a sink that doesn't support the surface.
A simple example that shows how it used to crash on a system where gstreamer-vaapi
is installed:
gst-launch playbin2 video-sink=xvimagesink uri=/codec/supported/by/vaapi
What we are now doing in this case, is avoid using the accelerated
decoder and plug a "normal" decoder instead (if avalaible).
This commit doesn't handle the case where we have hardware accelerated
demuxing.
2011-02-18 11:48:37 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-profile.h:
* win32/common/libgstpbutils.def:
encoding-profile: add a function to create a profile from a discoverer info
Only A/V streams are added at the moment, there does not seem to be
a similar way to add other streams (eg, subtitles).
https://bugzilla.gnome.org/show_bug.cgi?id=642878
2011-09-27 00:26:29 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
* ext/alsa/gstalsasrc.h:
alsasrc: fail gracefully when ALSA does not give timestamps
https://bugzilla.gnome.org/show_bug.cgi?id=660170
2011-10-03 10:55:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Use a TIME limit for pre-rolling in live streams and not in non-live streams
Fixes bug #647769 for real.
2011-10-01 01:05:00 +0100 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
* ext/pango/gsttextoverlay.c:
textoverlay: add YV12 support
Basically the same as I420, just with chroma planes swapped.
https://bugzilla.gnome.org/show_bug.cgi?id=660604
2011-09-30 09:44:12 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Fix typo on formatter adding condition
The condition is if the muxer doesn't have tag setter *and* isn't
a formatter itself. Any of those two conditions makes the muxer
good enough to not need a formatter.
2011-09-28 15:41:16 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: really push pending events
2011-09-28 14:32:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: remove more tags from upstream tag events such as bitrate tags
We want to remove all codec specific tags.
2011-09-28 01:56:42 +0300 Raimo Järvi <raimo.jarvi@gmail.com>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix compiler warning on 64 bit mingw-w64
Fixes bug #660304.
2011-09-28 01:11:30 +0300 Raimo Järvi <raimo.jarvi@gmail.com>
* gst/playback/gstplaybin2.c:
playbin2: Fix compiler warnings on 64 bit mingw-w64
Fixes bug #660301.
2011-09-27 16:18:05 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: only got_data if we really got some
... which avoids going loopy with casual subclass.
2011-09-27 16:57:45 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: really push pending events
2011-09-27 16:16:54 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: send tag event after pending events
... which probably includes a pending newsegment event.
2011-09-27 16:16:29 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: protect pending_events with proper lock
2011-09-27 15:31:20 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: clean up some documentation
2011-09-27 00:32:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: minor docs fix
2011-09-26 16:36:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: Adjust for GstAudioEncoder API changes
2011-09-26 16:36:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstaudio.def:
win32: Adjust for GstAudioEncoder API changes
2011-09-26 16:35:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: Improve set_frame_sample_{min,max} documentation
2011-09-26 16:22:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
audiodecoder: Fix thread safety issues if both pads have different streaming threads
2011-09-26 16:19:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: Delay sending of serialized events to finish_frame()
2011-09-26 16:02:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
Revert "audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code"
This reverts commit 11e375486e07cfa0686a97b5cf6110909b3a828c.
GST_BOILERPLATE() can't define an abstract type and
G_DEFINE_ABSTRACT_TYPE() does not pass the class struct to
the instance_init function and there's no way to get the
class struct of the current type in instance_init().
2011-09-26 15:59:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
audioencoder: Add support for requesting a minimum and maximum number of samples per frame
This extends the special case of a fixed number of samples per frame
that was supported before already.
2011-09-26 15:45:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
audioencoder: Fix thread safety issues if both pads have different streaming threads
2011-09-26 15:42:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: Delay sending of serialized events to finish_frame()
This makes sure that the caps are already set before any serialized
events are sent downstream.
2011-09-26 15:34:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code
2011-09-26 15:14:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
audioencoder: add some tag handling convenience help
2011-09-26 14:48:55 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: provide CODEC/AUDIO_CODEC handling
2011-09-26 13:42:38 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: filter AUDIO_CODEC/CODEC tags from passing tag events
2011-09-25 15:31:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: backport some const-ifications from 0.11 branch
To keep code identical as much as possible between the two branches,
for easier merging.
2011-09-25 15:24:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: fix indentation
2011-09-23 17:50:31 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Avoid unnecessary read only caps copy
2011-09-22 15:38:51 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: proxy some more optional downstream caps fields to upstream
2011-09-22 15:38:22 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: changed is verily the opposite of equal
2011-09-22 15:37:07 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: prevent crashing when comparing to a freshly inited GstAudioInfo
2011-09-22 15:36:22 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/audio.h:
audio: some more accessor macros for GstAudioInfo
2011-09-22 15:34:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: fix documentation typo
2011-09-19 18:32:26 +0100 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* tests/check/elements/videorate.c:
videorate: Add tests for the max-rate case
2011-09-19 18:31:07 +0100 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* tests/check/elements/videorate.c:
videorate: Print which caps didn't match up
2011-09-19 18:26:04 +0100 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: Add a max-rate property
In various use-case you want to dynamically change the framerate (e.g.
live streams where the available network bandwidth changes). Doing this
via capsfilters in the pipeline tends to be very cumbersome and racy,
using this property instead makes it very painless.
2011-09-01 17:05:23 +0100 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* tests/check/elements/videorate.c:
videorate: Add test for caps negotiation
2011-09-01 16:47:49 +0100 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Add more strict caps negotiation
When in drop-only mode we can never provide a framerate that is higher
then the input, so let the caps negotiation reflect this.
2011-09-20 13:35:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: don't unref event we don't own
http://bugzilla.gnome.org/show_bug.cgi?id=659562
2011-09-20 14:04:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Only check if this is a discarded type if we have fixed caps
For unfixed caps we will get here again later when the caps are fixed.
2011-09-20 14:03:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Only call autoplug-continue with fixed caps
With unfixed caps we can't reliably decide if the final caps
are going to be "raw" (e.g. supported by a sink) or not.
We will get here again later when the caps are fixed.
2011-09-20 13:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/decodebin2.c:
decodebin2: Fix unit test by strictly implementing parser behaviour instead of relying on basetransform
2011-01-13 15:35:30 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
oggstream: only use information from skeleton if we have nothing better
The codec setup headers are a lot more likely to have correct information,
especially as it's easy to remux a skeleton in a file where streams don't
have the same parameters (I've even seen a file with two skeletons).
Still, this is useful in the case we have a codec we can't decode, so we
can at least (theoretically) convert granpos to time, so we discard this
information if the codec setup has already provided it.
This fixes playback on (at lesat) the original archive.org encoding of
"The Night of the Living Dead" (now replaced by another encoding).
https://bugzilla.gnome.org/show_bug.cgi?id=612443
2011-09-19 14:16:19 +0200 Age Bosma <agebosma@gmail.com>
* gst-libs/gst/pbutils/gstdiscoverer.h:
discoverer: Don't use gtk-doc /* < ... > */ style comments for signals
The /*< ... >*/ style is only used for public|protected|private,
signal comments use /* signals */. This prevents the some code
parsers/binding generators to be confused by the comment.
2011-09-19 14:02:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Get the target of the video sinkpad, not the target sinkpad in the video setcaps handler
2011-08-18 15:13:23 +0000 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Initialize variable correctly
If subdrained isn't initialized to FALSE then a chain might think
that its group is drained when in fact it's not and this can cause
a switch too early or even cause a deadlock.
2011-07-28 16:44:33 +0000 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Rewrite EOS-handling code
This is now really threadsafe and improves switching
between different groups.
2011-09-19 11:53:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Fix non-prerolling pipelines and not-linked errors if a parser is available but no decoder
Fixes bug #658846.
2011-08-01 07:54:02 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtspdefs: add RTCP-Interval header
2011-09-19 11:24:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Implement support for switching between raw and non-raw video streams
2011-09-19 09:34:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Protect against accessing the NULL parent of the pads during shutdown
Fixes bug #658901.
2011-09-16 20:14:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: remove superfluous check in newsegment event handler
If we get a newsegment event from upstream, we can be quite
sure we're not operating pull-based.
2011-09-16 20:11:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: minor printf format fix
2011-09-14 12:23:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix wedge when seeking twice quickly in push mode
This could happen when testing with navseek, and pressing
right and left at roughly the same time. The current chain
is temporarily moved away, and this caused the flush events
not to be sent to the source pads, which would cause the
data queues downstream to reject incoming data after the
seek, and shut down, wedging the pipeline.
Now, I can't really decide whether this is a nasty steaming
hack or a good fix, but it certainly does fix the issue, and
does not seem to break anything else so far.
https://bugzilla.gnome.org/show_bug.cgi?id=621897
2011-08-13 14:18:56 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: implement push mode seeking
This patch implements seeking in push mode (eg, over the net)
in Ogg, using the double bisection method.
As a side effect, it also fixes duration determination of network
streams, by seeking to the end to check the actual duration.
Known issues:
- Getting an EOS while seeking stops the streaming task, I can't
find a way to prevent this (eg, by issuing a seek in the event
handler).
- Seeking twice in a VERY short succession with playbin2 fails
for streams with subtitles, we end up pushing in a dataqueue
which is flushing. Rare in normal use AFAICT.
- Seeking is slow on slow links - byte ranges guesses could be
made better, decreasing the number of required requests
- If no granule position is found in the last 64 KB of a stream,
duration will be left unknown (should be pretty rare)
https://bugzilla.gnome.org/show_bug.cgi?id=621897
2011-09-15 22:04:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/playback/gstplaybin2.c:
playbin2: fix compiler warning
Remove a check for gchar >= 128
2011-09-15 16:47:26 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: don't access the event after pushing
Fixes valgrind warnings.
2011-09-15 14:27:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
Revert "playbin2: autoplug sink if stream is incompatible to the configured one"
This reverts commit b0b4e286c8cde2e79a959a444a2c68e99c3f29c6.
We agreed that the previous (pre-.35) behaviour is broken and a bug and the
current behaviour is correct, deterministic and allows the application to
handle stuff properly while the old behaviour can't be handled properly by
applications and just worked in some applications by luck.
The solution to the problem that was solved by relying on the old, broken
behaviour would be, to make decodebin2/playbin2 more aware of decoders and
improve the autoplugging of decoders by considering the caps supported by the
sink instead of just using something with the highest rank.
See bug #656923.
2011-09-15 09:23:54 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/playback/gstplaybin2.c:
playbin2: autoplug sink if stream is incompatible to the configured one
Fixes regression since 0.10.33 where sinks that can cope with non raw
caps or custom caps are not autoplugged if there's a sink configured
with the properties video-sink and audio-sink which cannot handle
the stream. This change checks for compatibility on the configured one
and use it if success. Otherwhise it tries with the found factories.
2011-08-13 14:14:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not propagate discontinuities in sparse streams
The first packet of a sparse stream may arrive after an initial
delay in the stream. If ogg_stream_packetout reports a discontinuity
in a sparse stream, do not propagate it to other streams in the
chain unnecessarily.
https://bugzilla.gnome.org/show_bug.cgi?id=621897
2011-09-12 15:48:59 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/playback/gstplaysink.c:
Revert "playsink: only add text overlay if vido sink also accepts raw caps"
This reverts commit a22faad18a73a27a2a0c903748c1a355df4d8c13. Instead
of disabling subtitles completelly when video stream have custom caps,
just let the sutbtileoverlay cope with them as now it's able to.
2011-09-12 15:46:46 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: gracefully handle non raw video streams
Implement handling of non raw video streams by avoiding colorspace
elements and autoplugging a compatible renderer if available. Fallback
to passthrough if no compatible renderer is found.
2011-09-12 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: try to catch malformed URIs
Only log in debug log for now, since the check is a bit
half-hearted, its purpose is mostly to make sure people
use gst_filename_to_uri() or g_filename_to_uri().
https://bugzilla.gnome.org/show_bug.cgi?id=654673
2011-09-12 19:53:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/tag.h:
docs: minor addition to GST_TAG_ID3V2_HEADER_SIZE docs
2011-09-11 14:22:59 -0400 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
* ext/theora/gsttheoraenc.c:
theoraenc: Fix descriptions of properties
2011-09-10 18:30:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: don't try to fixate "width" field for alaw/mulaw
Fixes warning when trying to fixate e.g. pulsesrc ! audio/x-alaw ! fakesink.
2011-09-09 13:10:13 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/design/design-decodebin.txt:
docs: fix some typos in the decodebin design document
2011-09-09 13:07:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/colorbalance.c:
colorbalance: add some guards to interface methods
https://bugzilla.gnome.org/show_bug.cgi?id=658584
2011-09-09 12:07:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: recognize Asylum modules
Note that there is already a AMF detection for a different
magic, I'm not sure if that's a different format with the
same initials or not. AMF is used for a few different formats
(including video), so...
This fixes playbin2 playing Asylum modules.
https://bugzilla.gnome.org/show_bug.cgi?id=658514
2011-08-31 20:51:17 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/subparse/gstsubparse.c:
subparse: Improve subrip type check regex
This patch prevents timestamp like "1 1:00:00", which would have been seen
as hour 101 by our parser, and allow single digit hour, minute and seconds
as it's already supported by the parser, and also by other implementation
like in mplayer. This fixes bug 657872.
https://bugzilla.gnome.org/show_bug.cgi?id=657872
2011-09-08 14:46:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/design/design-decodebin.txt:
decodebin: Update design documentation about how Parser/Converter are handled
2011-09-08 13:25:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
Revert "decodebin2: Do a subset check before actually using a factory"
This reverts commit 50a88396ae6d54a83a10e7d2efd551d39033148e.
See bug #658541.
2011-09-07 16:44:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/decodebin2.c:
decodebin2: Don't use bufferalloc in the test elements
This will cause not-linked errors that usually don't happen
because normal decoders/parsers will set srcpad caps before
allocating buffers from downstream.
2011-09-07 16:43:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Make sure to fixate Parser/Converter caps before continuing autoplugging
2011-09-07 16:04:43 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/playback/gstplaysink.c:
playsink: only add text overlay if vido sink also accepts raw caps
Fixes regression, pipeline fails with not negotiated, on media
containing subtitles when decoder/sink with custom caps is used.
2011-09-07 14:19:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Intersect the factory caps with the current caps for the capsfilter
Otherwise we'll include many incompatible caps in the capsfilter that
will only slow down negotiation.
2011-09-07 14:07:00 +0200 Stefan Sauer <ensonic@users.sf.net>
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
docs: cleanup makefiles
Remove commented out parts that we don't need. Remove "the wingo addition" - no
so useful after all. Narrow down file-globs for plugin docs.
2011-09-07 14:04:10 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.h:
docs: add two mising enum docs
2011-09-07 14:10:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/audiorate.c:
audiorate: Use complete audio caps, including the endianness field
2011-09-07 12:32:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: fix element factory refcounting
g_value_get_object() does not give us our own ref.
Fixes "Trying to dispose object "flacparse", but it still has a parent "registry0".
You need to let the parent manage the object instead of unreffing the object directly."
and similar warnings.
https://bugzilla.gnome.org/show_bug.cgi?id=658416
2011-09-07 11:06:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: do not automatically override quality when using target bitrate
If both quality and bitrate are set, libtheora will try to meet
both constraints, causing it to prefer emitting a smaller number
of good frames, to emitting the full number of frames that would
not meet the requested quality. This causes a slideshow effect
when the bitrate is low and the quality is high. And the default
theoraenc is high (48/63).
So only set quality when it is requested, and leave it unset
otherwise.
https://bugzilla.gnome.org/show_bug.cgi?id=658443
2011-09-06 21:24:33 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From a39eb83 to 11f0cd5
2011-09-06 19:18:27 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Add latest files to spec file
2011-09-06 20:13:30 +0200 Stefan Sauer <ensonic@users.sf.net>
* docs/libs/Makefile.am:
docs: activate overrides file to fix make distcheck
2011-09-06 16:46:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/audio.h:
audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN
2011-09-06 15:46:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/audio.c:
audio: update internal silent sample defines as well to match 0.11
2011-09-06 15:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
audio: update audio format enums to match changes in 0.11
And add new audio format info stuff to docs.
2011-09-06 15:40:02 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 605cd9a to a39eb83
2011-09-06 14:16:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Do a subset check before actually using a factory
This prevents autoplugging if the caps have a non-empty intersection
but are not accepted by the next element's pad.
2011-09-06 14:04:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Use subset check instead of non-empty-intersection check to check if pads are compatible
2011-09-06 14:03:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Use subset check instead of non-empty-intersection check to check if pads are compatible
2011-09-06 13:06:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Fix memory leak
2011-09-06 12:14:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/decodebin2.c:
decodebin2: Add unit test for correct parser/converter negotiation
2011-06-26 15:40:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Correctly negotiate format for parsers that can convert different stream formats
This is done by adding a capsfilter after every parser/converter that contains
all possible caps supported by downstream elements. A capsfilter is necessary
here because the decoder is only selected after the parser selected a format
and the parser can't know what downstream would support otherwise.
2011-09-05 15:19:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: If a audio/video sink was already selected don't check caps of all other possible sinks
2011-09-06 08:25:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/decodebin2.c:
decodebin2: Add Tim as author for the parser test
2011-09-06 10:07:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.h:
docs: more docs clean-ups
2011-09-05 23:00:30 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: don't take the object lock twice in {set,get}_property
https://bugzilla.gnome.org/show_bug.cgi?id=658294
2011-09-05 22:51:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/audio.h:
audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean
2011-09-05 21:40:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.h:
docs: some docs love
2011-09-05 20:45:22 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
docs: add GstAudioDecoder and GstAudioEncoder to documentation
2011-09-05 15:01:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
* win32/common/libgstaudio.def:
audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder
API: gst_gst_audio_decoder_finish_frame()
API: gst_gst_audio_decoder_get_audio_info()
API: gst_gst_audio_decoder_get_byte_time()
API: gst_gst_audio_decoder_get_delay()
API: gst_gst_audio_decoder_get_latency()
API: gst_gst_audio_decoder_get_max_errors()
API: gst_gst_audio_decoder_get_min_latenc()y
API: gst_gst_audio_decoder_get_parse_state()
API: gst_gst_audio_decoder_get_plc()
API: gst_gst_audio_decoder_get_plc_aware()
API: gst_gst_audio_decoder_get_tolerance()
API: gst_gst_audio_decoder_get_type()
API: gst_gst_audio_decoder_set_byte_time()
API: gst_gst_audio_decoder_set_latency()
API: gst_gst_audio_decoder_set_max_errors()
API: gst_gst_audio_decoder_set_min_latency()
API: gst_gst_audio_decoder_set_plc()
API: gst_gst_audio_decoder_set_plc_aware()
API: gst_gst_audio_decoder_set_tolerance()
API: gst_gst_audio_encoder_finish_frame()
API: gst_gst_audio_encoder_get_audio_info()
API: gst_gst_audio_encoder_get_frame_max()
API: gst_gst_audio_encoder_get_frame_samples()
API: gst_gst_audio_encoder_get_hard_resync()
API: gst_gst_audio_encoder_get_latency()
API: gst_gst_audio_encoder_get_lookahead()
API: gst_gst_audio_encoder_get_mark_granule()
API: gst_gst_audio_encoder_get_perfect_timestamp()
API: gst_gst_audio_encoder_get_tolerance()
API: gst_gst_audio_encoder_get_type()
API: gst_gst_audio_encoder_proxy_getcaps()
API: gst_gst_audio_encoder_set_frame_max()
API: gst_gst_audio_encoder_set_frame_samples()
API: gst_gst_audio_encoder_set_hard_resync()
API: gst_gst_audio_encoder_set_latency()
API: gst_gst_audio_encoder_set_lookahead()
API: gst_gst_audio_encoder_set_mark_granule()
API: gst_gst_audio_encoder_set_perfect_timestamp()
API: gst_gst_audio_encoder_set_tolerance()
https://bugzilla.gnome.org/show_bug.cgi?id=642690
2011-08-03 13:31:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Select muxer further
Sort muxers based on their caps and ranking before iterating to
find one that fits the profile.
Sorting is done by putting the elements that have a pad template
that can produce the exact caps that is on the profile. For example:
when asking for "video/quicktime, variant=iso", muxers that
have this exact caps on their pad templates will be put first on
the list than ones that have only "video/quicktime".
https://bugzilla.gnome.org/show_bug.cgi?id=651496
2011-09-05 20:31:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Actually iterate over the factories instead of only taking the first one
2011-09-05 15:51:25 +0200 Stefan Sauer <ensonic@users.sf.net>
* tests/check/libs/profile.c:
* tests/check/libs/tag.c:
* tests/check/libs/video.c:
tests: supress ERROR log output for some tests
Be nice when we tests for correct error handling and don't spam stdout.
2011-09-05 14:40:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
Revert "playsink: Try include 'pitch', if no other sink is provided"
This reverts commit 105814e2c78f9867c61531b9e8166e4ae994296f.
The general consensus seems to be that we should revert this for
now. If such behaviour is desired, we should probably enable it
via a flag. And maybe use the scaletempo plugin instead.
2011-09-05 12:02:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Don't leak the videochain ts-offset element
Also don't leak the audiochain ts-offset element if one is
found but the sink doesn't support volume settings.
2011-09-05 11:55:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Use gst_object_unref() instead of g_object_unref() for better debugging
2011-03-17 19:13:58 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/Makefile.am:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_lanczos.c:
videoscale: Add modified Lanczos scaling method
Adds a Lanczos-derived scaling method, which is rather slow, but very
high quality. Adds a few properties that can be used to tune various
scaling properties: sharpness, sharpen, envelope, dither. Not currently
Orcified, but was designed with that in mind.
2011-05-16 14:46:52 -0700 David Schleef <ds@schleef.org>
* gst/playback/Makefile.am:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysinkvideoconvert.c:
* gst/playback/gstsubtitleoverlay.c:
playback: Add define for colorspace element
Single point of change if you want to switch from ffmpegcolorspace
to colorspace.
2011-08-25 15:14:58 +0100 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: fix dynamically changing average period
The average_period_set variable can be accessed in different threads, so
always lock it when reading. Furthermore when switching to averaging
mode we should make sure we don't have cached buffers that aren't used
in that mode. And any modeswitch will cause the latency to change, so we
should post a NewLatency message
2011-08-23 10:11:52 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* gst/videorate/Makefile.am:
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: Port to basetransform
2011-08-22 15:52:57 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* gst/videorate/gstvideorate.c:
Correct added versions
2011-08-31 14:45:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Only unref ts_offset elements if they're not NULL
2011-08-31 12:39:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Keep the chain mutex locked while connecting to the notify::caps signal
2011-08-30 18:21:31 +1000 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/seek/seek.c:
seek: Accept pipeline descriptions for audiosink/videosink
Make the element_factory_make_or_warn utility function try parsing
the input string as a bin if element_factory_make() fails. This makes
the --audiosink/--videosink commandline options accept a pipeline
string.
2011-08-30 18:21:31 +1000 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaysink.c:
playsink: Try include 'pitch', if no other sink is provided
As a default, try the pipeline 'pitch ! audioconvert ! autoaudiosink'
before trying plain autoaudiosink
2011-08-27 14:57:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstdiscoverer.c:
pbutils: don't depend on libgstvideo just to parse some caps
Let's extract those ints and fractions ourselves and not depend
on libgstvideo.
2011-08-27 13:31:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* win32/common/libgstaudio.def:
audio: add GstBaseAudioDecoder and GstBaseAudioEncoder to build
However, libgstaudio now depends on libgstvideo (via pbutils).
https://bugzilla.gnome.org/show_bug.cgi?id=642690
API: gst_audio_info_clear()
API: gst_audio_info_convert()
API: gst_audio_info_copy()
API: gst_audio_info_free()
API: gst_audio_info_from_caps()
API: gst_audio_info_init()
API: gst_audio_info_to_caps()
API: gst_base_audio_decoder_finish_frame()
API: gst_base_audio_decoder_get_audio_info()
API: gst_base_audio_decoder_get_byte_time()
API: gst_base_audio_decoder_get_delay()
API: gst_base_audio_decoder_get_latency()
API: gst_base_audio_decoder_get_max_errors()
API: gst_base_audio_decoder_get_min_latency()
API: gst_base_audio_decoder_get_parse_state()
API: gst_base_audio_decoder_get_plc()
API: gst_base_audio_decoder_get_plc_aware()
API: gst_base_audio_decoder_get_tolerance()
API: gst_base_audio_decoder_get_type()
API: gst_base_audio_decoder_set_byte_time()
API: gst_base_audio_decoder_set_latency()
API: gst_base_audio_decoder_set_max_errors()
API: gst_base_audio_decoder_set_min_latency()
API: gst_base_audio_decoder_set_plc()
API: gst_base_audio_decoder_set_plc_aware()
API: gst_base_audio_decoder_set_tolerance()
API: gst_base_audio_encoder_finish_frame()
API: gst_base_audio_encoder_get_audio_info()
API: gst_base_audio_encoder_get_frame_max()
API: gst_base_audio_encoder_get_frame_samples()
API: gst_base_audio_encoder_get_hard_resync()
API: gst_base_audio_encoder_get_latency()
API: gst_base_audio_encoder_get_lookahead()
API: gst_base_audio_encoder_get_mark_granule()
API: gst_base_audio_encoder_get_perfect_timestamp()
API: gst_base_audio_encoder_get_tolerance()
API: gst_base_audio_encoder_get_type()
API: gst_base_audio_encoder_proxy_getcaps()
API: gst_base_audio_encoder_set_frame_max()
API: gst_base_audio_encoder_set_frame_samples()
API: gst_base_audio_encoder_set_hard_resync()
API: gst_base_audio_encoder_set_latency()
API: gst_base_audio_encoder_set_lookahead()
API: gst_base_audio_encoder_set_mark_granule()
API: gst_base_audio_encoder_set_perfect_timestamp()
API: gst_base_audio_encoder_set_tolerance()
2011-08-27 13:15:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
docs: add since markers to baseaudio{decoder,encoder} documentation
2011-08-27 12:47:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudiodecoder, baseaudioencoder: fix some compiler warnings
Leaving the GST_USE_UNSTABLE_API guards in until some of the
ported decoders have been updated and it's clear that I didn't
mess up anywhere porting things to the new audio API.
2011-08-27 12:41:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioutils.c:
* gst-libs/gst/audio/gstbaseaudioutils.h:
baseaudioutils: remove, merged into or superseded by audio.c
2011-08-27 12:39:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: port to new GstAudioInfo API
2011-08-27 12:37:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiodecoder: port to GstAudioInfo API
2011-08-27 11:43:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
audio: add gst_audio_info_{init,clear} and gst_audio_info_{copy,free}
2011-08-22 20:15:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/multichannel.c:
* gst-libs/gst/audio/multichannel.h:
audio: add GstAudioFormat, GstAudioFormatInfo and GstAudioInfo
Same as in 0.11, but with caps parsing/serialising for 0.10 style
caps. Add setting default channel positions.
2011-08-17 18:48:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: remove leftover experimental code
2011-08-17 18:32:54 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioutils.c:
* gst-libs/gst/audio/gstbaseaudioutils.h:
audioutils: modify _parse, add GType support functions
2011-08-16 21:11:42 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiodecoder: move properties to private storage and add _get/_set
2011-08-16 21:11:52 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: rename property
2011-08-16 20:39:07 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiodecoder: replace context helper structure by various _get/_set
2011-08-16 18:59:13 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: move properties to private storage and add _get/_set
2011-08-16 18:25:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: rename some properties
2011-08-16 18:23:14 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: replace context helper structure by various _get/_set
2011-08-16 17:27:07 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
* gst-libs/gst/audio/gstbaseaudioutils.c:
* gst-libs/gst/audio/gstbaseaudioutils.h:
baseaudio: rename GstAudioState to GstAudioFormatInfo
2011-06-17 11:54:08 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: TEMP; avoid some imperfect ts jitter ?
... even when not in perfect mode ?
2011-04-28 12:01:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: debug format fixes
2011-04-28 12:01:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: debug format fix
2011-03-31 14:03:11 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: fixup documentation
2011-03-29 15:51:40 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: fix FLUSH_STOP actions
2011-03-28 13:16:27 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: preserve upstream seek event seqnum
2011-03-22 11:09:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: use buffer running time for granule calculation
2011-03-22 10:45:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: minor fix in ts resync
2011-03-21 11:40:31 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiodecoder: improve glitch resilience
Provide a replacement for GST_ELEMENT_ERROR to avoid aborting at the first
atom out of place, while on the other hand not failing indefinitely.
2011-03-17 12:09:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiodecoder: add limited legacy seeking support
2011-03-16 14:41:40 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: cater for audio-codec tag
2011-03-10 16:01:05 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiodecoder: initial version
2011-03-16 18:41:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: misc fixes
2011-03-15 17:27:42 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
* gst-libs/gst/audio/gstbaseaudioutils.c:
* gst-libs/gst/audio/gstbaseaudioutils.h:
baseaudio: add audioutils for caps and query handling helper utils
2011-03-14 12:39:49 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: mark unstable API
2011-03-10 15:12:54 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: fix clearing context
2011-03-10 15:12:19 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: simplify latency variable handling
2011-03-10 14:28:48 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: minor fixes and code simplifications
Also modify and elaborate a bit on pre_push (though currently unused to no harm).
2011-03-09 12:44:36 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: additional documentation on granule semantics and configuration
2011-03-09 12:24:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: elaborate property names
2011-03-09 12:22:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: rename state field xint to is_int
2011-03-09 12:18:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
baseaudioencoder: gtk-doc syntax fixes
2011-03-09 12:17:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
baseaudioencoder: minor fix and cleanup
2011-03-01 14:08:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
baseaudiocodec: ... and also rename to baseaudiodecoder
2011-03-01 13:58:31 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
gst-libs/gst/audio: Remove baseaudiodecoder
Adds little beyond baseaudiocodec (seeking, bit of query), and what it adds
is mainly out-of-scope (e.g. decoder seeking, should be done by upstream
demuxer/parser) and/or based on non-prime example (mad).
2009-09-17 13:26:28 +0200 Iago Toral <itoral@igalia.com>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
baseaudiodecoder: Return TRUE if we run into special conversion cases.
2009-09-01 14:17:53 +0200 Iago Toral <itoral@igalia.com>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
audio: initial version of GstBaseAudioCodec
Moved most of the code to GstBaseAudioCodec, GstBaseAudioDecode is
now really small, maybe we do not really need it (or its encoder
counterpart). Added more API for subclasses and documentation.
2009-08-14 09:45:52 +0200 Iago Toral <itoral@igalia.com>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
Added src_queries to decoder class. Added handle_discont to decoder class. Reworked reset. Various other minor fixes.
2009-08-06 15:28:00 +0200 Iago Toral <itoral@igalia.com>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
Added a draft implementation of gstbaseaudiodecoder
2011-03-01 11:56:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
Added audio directory for audio codec base classes
2011-02-18 16:38:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
audioencoders: add streamheader helper utility
2011-01-27 16:52:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
audioencoders: baseaudioencoder and ported encoders
2011-08-26 10:03:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstpbutils.def:
win32: Add new discoverer API
2011-08-26 10:03:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: Add new discoverer API
2011-08-24 16:29:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* gst-libs/gst/pbutils/pbutils-private.h:
* tools/gst-discoverer.c:
discoverer: retrieve audio track language from tags too
https://bugzilla.gnome.org/show_bug.cgi?id=657257
2011-08-24 15:09:47 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: consider subtitles as raw
Otherwise, discoverer will generated an "inner" codec
where there can be a tranformation (eg, kate -> DVD SPU,
and various ->text/x-pango-markup).
https://bugzilla.gnome.org/show_bug.cgi?id=639055
2011-08-24 15:05:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: add application/x-kate to subtitles caps
https://bugzilla.gnome.org/show_bug.cgi?id=639055
2011-08-24 14:59:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: get language from other tags if we did not get it already
https://bugzilla.gnome.org/show_bug.cgi?id=639055
2011-08-24 15:04:50 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* gst-libs/gst/pbutils/pbutils-private.h:
* tools/gst-discoverer.c:
discoverer: add subtitles API
https://bugzilla.gnome.org/show_bug.cgi?id=639055
2011-08-21 14:51:45 -0700 David Schleef <ds@schleef.org>
* gst/playback/gstplaysink.c:
playback: reference count ts_offset
Apparently this object is being used after it's freed. This is one
way to fix it, although perhaps not the best way. Fixes: #656715.
2011-08-25 14:55:14 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: fix caps leak
https://bugzilla.gnome.org/show_bug.cgi?id=657333
2011-07-08 23:06:46 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppayload: Make perfect timestamps reproducible across element restart
Without the perfect timestamp machinery, the RTP timestamp can be
computed directly from the running time of a buffer, but the perfect
timestamp patch broke that assumption. This patch restores it by
having the first perfect timestamp be the running time of that buffer
and counting from there.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=654434
2011-08-24 17:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: fix leaks in skeleton writing
https://bugzilla.gnome.org/show_bug.cgi?id=563251
2011-08-18 16:36:23 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: generate message headers from received tags
Some message headers can be deduced from tags (eg, "Language").
https://bugzilla.gnome.org/show_bug.cgi?id=563251
2011-08-18 10:05:17 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggparse.c:
ogg: use memory slices where appropriate
While there, avoid zeroing newly allocated memory where unnecessary
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-24 14:05:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkvideoconvert.c:
playsink{audio,video}convert: Send NEWSEGMENT events to sinkpads instead of pushing them
2011-08-23 11:12:10 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not warn when reaching EOS while scanning for the end chain
After all, we were asking for it.
This gets rid of the last warning-about-expected-condition.
w00t.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 11:08:25 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: add media type to chain information reports
One more little step in making logs a little less abstruse.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 11:05:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: correctly identify skeleton EOS packet
It is 0 byte, and was triggering the "bad packet" logic.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 10:58:20 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not warn about expected occurences
In this case, finding a skeleton packet.
Once upon a time, it used to be rare indeed, but no more.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 10:47:53 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not warn when finding a non BOS page
After all, we do hope to find actual data for these streams.
However, warn if we could not set up a chain when we find a
non BOS page, as that means we don't have a valid Ogg stream.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 10:40:12 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: rename local variable for clarity
While the casual reader might end up bewildered by just why this
change might increase clarity, it just happens than, in the libogg
and associated sources, op is the canonical name for an ogg_packet
whlie og is the canonical name for an ogg_page, and reading this
code confuses me.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 10:32:36 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not try to determine duration of header packets
Headers are inherently durationless.
Instead, set duration to 0 to avoid increasing tracked granpos,
and do not warn about it, since it is totally expected.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 10:29:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: include stream type in warnings
It makes it easier to work out what's going on.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-23 10:28:33 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: set skeleton stream media type to application/x-ogg-skeleton
This is to match the typefinder, and to make logs clearer.
https://bugzilla.gnome.org/show_bug.cgi?id=657151
2011-08-17 17:09:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: add skeleton write support
Version written is 3.0
Base times are left empty for now.
Content-Type should be the MIME type of the stream. It is set to
the GStreamer media type for now, which is probably the same for
the streams oggmux supports.
https://bugzilla.gnome.org/show_bug.cgi?id=563251
2011-08-22 14:56:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not skip sparse streams when determining start times
This fixes demuxing of streams containing only sparse streams,
which would cause an infinite loop in _read_end_chain.
https://bugzilla.gnome.org/show_bug.cgi?id=657062
2011-08-22 14:55:59 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do not ignore sparse streams' start time
But do not wait for them either, if we don't have a packet for them.
https://bugzilla.gnome.org/show_bug.cgi?id=657062
2011-07-21 17:16:26 -0400 Monty Montgomery <cmontgom@redhat.com>
* ext/vorbis/gstvorbisenc.c:
vorbisenc: Relax overly-tight jitter tolerances in gstvobisenc
vorbisenc currently reacts in a rater draconian fashion if input
timestamps are more than 1/2 sample off what it considers ideal. If data
is 'too late' it truncates buffers, if it is 'too soon' it completely
shuts down encode and restarts it. This is causingvorbisenc to produce
corrupt output when encoding data produced by sources with bugs that
produce a smple or two of jitter (eg, flacdec)
2011-08-22 09:06:53 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: fix text buffer leak
Make sure to always unref the input text buffer.
Reported by bcxa.sz@gmail.com.
https://bugzilla.gnome.org/show_bug.cgi?id=657049
2011-08-20 19:46:31 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/gstvideosink.h:
docs: fix xref for the property
2011-08-20 19:16:42 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/navigation.c:
* gst-libs/gst/interfaces/streamvolume.h:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/install-plugins.h:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst-libs/gst/video/gstvideosink.h:
docs: handle warnings emitted by gtk-doc
This is useful and in most cases someone had put arbitrary markup into the docs,
misspelled xref'ed symbols, forgot to add stuff to the docs etc..
2011-08-20 17:53:11 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: partially revert my last commit
Somehow this was already there, but I missed that commit.
2011-08-20 14:11:11 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/licenses.c:
docs: add new taglicense docs and clean them up
Avoid ugly docbook tags unless needed.
2011-08-20 12:37:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update for new translatable string
2011-08-20 12:36:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
tag: fix distcheck issue
Dist licenses dict.
2011-08-18 16:20:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggparse.c:
ogg: do not use 32 bit modifiers to print serial numbers
If ints are 64 bits, 32 bits should get promoted in varargs anyway,
and we don't care about 16 bit ints.
This makes the code a lot more readable, and still gets us nice
hexadecimal 32 bit serialnos.
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-07-27 11:05:31 +0000 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Reconfigure when pads are added later
Instead of just assuming all pads are created at the same time,
remember which ones are actually new (via ->pending_blocked_pads).
This allows the following use-case to properly work:
* Upstream starts with audio-only
* Only that pad gets data, blocks and a real audio sink is created
* Upstream laters adds a video stream
* A new pad is requested, blocks and reconfiguration kicks in in
order to add a new real video sink
2011-08-18 09:37:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/README:
ogg: get the operator precedence right, even if only a doc
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-18 09:30:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: vorbis has a preroll of 2
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-17 19:40:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggstream: new convenience function to get a stream's media type
This will make logging a lot clearer, both in code and in output.
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-17 18:48:54 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
ogg: move the "always flush page" to oggstream
It avoids checking for specific media types in the muxer.
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-17 18:38:39 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: use oggstream to decide which BOS packets to place first
Ogg recommends video BOS packets to be first.
Use the "is_video" flag in oggstream to select those, rather than
check for known mime types.
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-17 18:03:16 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.h:
ogg: rationalize serialno type to guint32
It is a 32 bit unsigned number.
Sure, the libogg API uses a long, but that's an unfortunate oversight.
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-17 17:39:18 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: factor the header packet creation code
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-17 17:18:47 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: headers should always have granpos 0
https://bugzilla.gnome.org/show_bug.cgi?id=656775
2011-08-18 09:48:16 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/audioresample/resample.c:
audioresample: fix build without orc
https://bugzilla.gnome.org/show_bug.cgi?id=656781
2011-08-15 01:22:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstid3tag.c:
* tests/check/libs/tag.c:
tag: id3: avoid some more relocations in genre table
2011-08-12 12:07:32 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/elements/audioresample.c:
audioresample: add FFT based checks
Send a few simple tones through audioresample and check
that the main frequency spot is the same for the input and
the resampled output.
https://bugzilla.gnome.org/show_bug.cgi?id=656392
2011-08-15 23:41:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: add OSX specific hack to detect when a connection is refused
Unlike linux, OSX wakes up select with POLLOUT (instead of POLLERR) when
connect() is done async and the connection is refused. Therefore always check
for the socket error state using getsockopt (..., SO_ERROR, ...) after a
connection attempt.
2011-08-15 00:17:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: add new license API to docs
2011-08-15 00:03:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: try pkg-config first when looking for zlib
2011-08-14 20:44:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.3.0.txt:
* gst-libs/gst/tag/id3v2.4.0-frames.txt:
* gst-libs/gst/tag/id3v2.4.0-structure.txt:
tag: id3v2: add specs to git for reference
2011-08-14 13:32:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: avoid some relocations, make table static
2011-08-14 01:47:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.c:
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: add debug category for ID3 tag parsing
2011-07-18 18:09:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* configure.ac:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/id3v2.c:
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
* win32/common/libgsttag.def:
tag: id3v2: add id3v2 tag parsing helpers
https://bugzilla.gnome.org/show_bug.cgi?id=654388
2011-02-22 15:19:00 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: return ID3TAGS_BROKEN_TAG for unsupported versions
This prevents us for trying to work with a NULL taglist.
2011-01-02 19:23:51 +0000 Erich Schubert <erich@debian.org>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: fix parsing of ID3v2.4 genre frames with multiple genres
We'd only extract the first genre (multiple times) instead of all
genres.
https://bugzilla.gnome.org/show_bug.cgi?id=638535
2010-09-24 15:19:15 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: Sanitize id3 frame names
This is similar to what is done in qtdemux. Avoids providing invalid
structure/tags names
2010-03-30 01:50:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: fix parsing of unsynced frames with data length indicator
Fixes bug #614158.
2010-03-20 00:54:14 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/tag/id3v2.c:
Add -Wwrite-strings to the configure flags
... and fix all warnings
2009-12-13 13:19:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: prefer two letter ISO 639-1 code for extended comment
2009-10-09 15:59:25 +0200 Josep Torra <n770galaxy@gmail.com>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: fixes warnings building on macosx
Another round on the formating of that debug line.
2009-10-09 14:44:02 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: cast pointer math results to glong
2009-10-09 13:38:17 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: don't cast, but use the right format specified instead
This correct some of the previous macos fixes.
2009-10-09 11:42:36 +0200 Josep Torra <n770galaxy@gmail.com>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: fix printf warnings on macosx
2009-10-07 14:03:20 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: fprintf, sprintf, sscanf need stdio.h
2009-09-22 15:03:20 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: Fix compile warnings with gcc 4.0.1.
2009-08-09 12:52:17 +0200 LoneStar <lone@auvtech.com>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8
Fixes bug #499242.
2009-08-07 16:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: sizes in ID3 v2.3 are unlikely to be sync-safe integers
In ID3 v2.3 compressed frames will have a 4-byte data length indicator
after the frame header to indicate the size of the decompressed data.
This integer is unlikely to be a sync-safe integer for v2.3 tags,
only in v2.4 it's sync-safe.
2009-08-07 16:36:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: fix typo in debug message
2009-08-07 16:02:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.c:
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c:
tag: id3v2: fix parsing of unsync'ed ID3 v2.4 tags and frames
Reversing the unsynchronisation seems to work slightly differently
for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
sizes in the frame header, so the unsynchronisation is applied to
the whole frame data including all the frame headers. v2.4 frames
have sync-safe sizes, however, so the unsynchronisation only needs
to be applied to the actual frame data, and it seems that's what's
being done as well. So we need to undo the unsynchronisation on a
per-frame basis for v2.4 tags for things to work properly.
Fixes extraction of coverart/images from APIC frames in ID3 v2.4
tags (#588148).
Add unit test for this as well.
2009-04-24 01:51:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: parse unsynchronised tags properly
We didn't handle unsynchronization at all up to now, which might have
caused frames to not be extracted - esp. frames after an APIC picture
frame. Fixes #577468.
2009-04-24 01:01:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/id3v2.c:
tag: id3v2: pass the right size value for size of all frames to the parser
Frame data size is tag size adjusted for size of the tag header and
footer, not tag size including header and footer.
2008-06-04 10:42:46 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Use new utility functions in libgsttag to process coverart (#512333).
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
Use new utility functions in libgsttag to process coverart (#512333).
2008-01-11 21:08:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Generate the image-type values correctly. Leave them out of the caps when outputting a "preview image" tag, since it ...
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
Generate the image-type values correctly. Leave them out of the caps
when outputting a "preview image" tag, since it only makes sense
to have one of those - the type is irrelevant.
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_open):
If we can, mark the mixer multiple open when we use it, in case
(for some reason) the process wants to open it again elsewhere.
2008-01-09 15:20:19 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
tag: id3v2: Make sure the ISO 639-X language code in ID3v2 COMM frames so we don't end up with non-UT...
Original commit message from CVS:
Based on patch by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
* gst-libs/gst/tag/id3v2frames.c: (parse_comment_frame):
Make sure the ISO 639-X language code in ID3v2 COMM frames
is actually valid UTF-8 (or rather: ASCII), so we don't end
up with non-UTF8 strings in tags if there's garbage in the
language field. Also make sure the language code is always
lower case. Fixes: #508291.
2007-12-14 10:17:10 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Parse WOAF frames and put the result into GST_TAG_CONTACT, which is where it would end up...
Original commit message from CVS:
* tag: id3v2: (parse_url_link_frame):
Parse WOAF frames and put the result into GST_TAG_CONTACT,
which is where it would end up if the same information was
put in a vorbis comment (don't think it's worth adding a
new URI tag for this). Fixes #488112.
2007-11-14 21:39:47 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: We don't want the same string multiple times in a tag list for the same tag ever, for any tag, not jus...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c:
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c: (id3v2_tag_to_taglist):
We don't want the same string multiple times in a tag list for the
same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure
this doesn't happen and remove special-case code for GST_TAG_GENRE.
2007-10-11 17:55:29 +0000 Jason Kivlighn <jkivlighn@gmail.com>
tag: id3v2: Extract license/copyright URIs from ID3v2 WCOP frames (Fixes #447000).
Original commit message from CVS:
Based on patch by: Jason Kivlighn <jkivlighn gmail com>
* gst-libs/gst/tag/id3v2frames.c:
Extract license/copyright URIs from ID3v2 WCOP frames
(Fixes #447000).
* tests/check/elements/id3demux.c:
* tests/files/Makefile.am:
* tests/files/id3-447000-wcop.tag:
Add simple unit test.
2007-10-06 16:13:14 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Port ID3 tag demuxer over to the new GstTagDemux in -base (now would be a good time to test re-importi...
Original commit message from CVS:
* gst-libs/gst/tag/gstid3demux.c:
* gst-libs/gst/tag/gstid3demux.h:
* gst-libs/gst/tag/id3v2.c:
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c:
Port ID3 tag demuxer over to the new GstTagDemux in -base
(now would be a good time to test re-importing your music
collection).
2007-03-12 13:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is the image format a vari...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
the image format a variable-length NUL-terminated string; in
versions before that the image format is a fixed-length string of
3 characters (see #348644 for a sample tag).
Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.
2007-03-06 18:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interp...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_obsolete_tdat_frame):
Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
the four-digit number will be interpreted as a year, whereas it is
month and day in DDMM format. Instead, parse TDAT frames and fix up
the date in the GST_TAG_DATE tag later if we also extracted a year.
Fixes #407349.
2006-11-19 13:41:53 +0000 René Stadler <mail@renestadler.de>
tag: id3v2: Make sure that g_free always gets called on the same pointer that was returned by g_mallo...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame):
Make sure that g_free always gets called on the same pointer that was
returned by g_malloc. Fixes #376594.
Do not leak memory if decompressed size is wrong.
Remove unneeded check of return value of g_malloc.
Patch by: René Stadler <mail@renestadler.de>
2006-11-01 13:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: We require a -base more recent than 0.10.9, so it's safe to use
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
We require a -base more recent than 0.10.9, so it's safe to use
GST_TYPE_TAG_IMAGE_TYPE unconditionally now.
* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
Use _newsegment_full() now that we depend on a recent enough core.
* gst/wavparse/gstwavparse.c:
Remove cruft that we don't need any longer now that we depend on
a recent enough -base.
2006-10-05 16:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Printf format fixes.
Original commit message from CVS:
* ext/cairo/gsttimeoverlay.c:
(gst_cairo_time_overlay_update_font_height):
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
* ext/libpng/gstpngdec.c: (user_endrow_callback):
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_data):
* gst/cutter/gstcutter.c: (gst_cutter_chain):
* gst/debug/efence.c: (gst_efence_buffer_alloc),
(gst_fenced_buffer_copy):
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame):
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
(gst_rtspsrc_handle_message):
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
Printf format fixes.
2006-08-22 13:53:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: If strings in text fields are marked ISO8859-1, but contain valid UTF-8 already, then han...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_text_identification_frame),
(parse_insert_string_field):
If strings in text fields are marked ISO8859-1, but contain
valid UTF-8 already, then handle them as UTF-8 and ignore
the encoding. (#351794)
2006-08-16 13:01:32 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Require CVS of GStreamer core and -base (for
Original commit message from CVS:
* configure.ac:
Require CVS of GStreamer core and -base (for
GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
* ext/taglib/gstid3v2mux.cc:
Write extended comment tags properly (#348762).
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame):
Extract COMM frames into extended comments, which makes it
easier to properly retain the description bit of the tag
and maintain this information when re-tagging (#348762).
2006-07-25 16:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as well, and add the version to...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c:
(id3demux_add_id3v2_frame_blob_to_taglist):
Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
well, and add the version to the blob's buffer caps, since that
information will be needed for deserialisation later on (#348644).
2006-07-23 11:33:54 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: On second thought, it might be wiser and more efficient not to do tag registration from a streaming th...
Original commit message from CVS:
* gst-libs/gst/tag/gstid3demux.c: (plugin_init):
* gst-libs/gst/tag/id3v2.c:
(id3demux_add_id3v2_frame_blob_to_taglist):
* gst-libs/gst/tag/id3v2.h:
On second thought, it might be wiser and more efficient
not to do tag registration from a streaming thread.
2006-07-23 10:56:27 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Put ID3v2 frames we can't parse as binary blobs into private tags, so that they are not lost ...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c:
(id3demux_add_id3v2_frame_blob_to_taglist),
(id3demux_id3v2_frames_to_tag_list):
Put ID3v2 frames we can't parse as binary blobs into private
tags, so that they are not lost when retagging, at least once
id3v2mux has been taught to re-inject those frames again.
See bug #334375.
2006-07-21 10:57:00 +0000 Wim Taymans <wim.taymans@gmail.com>
tag: id3v2: Don't use \n in debug lines
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_process_next_entry):
Fix some leaks.
* gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
Don't use \n in debug lines.
2006-06-22 12:17:13 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Set image type from APIC frame as "image-type" field of GST_TAG_IMAGE buffer caps (#344605).
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
Set image type from APIC frame as "image-type" field
of GST_TAG_IMAGE buffer caps (#344605).
2006-06-11 19:31:10 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Extract images from ID3v2 tags (APIC frames). Fixes #339704.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(scan_encoded_string), (parse_picture_frame):
Extract images from ID3v2 tags (APIC frames). Fixes #339704.
* configure.ac:
Require core >= 0.10.8 (for GST_TAG_IMAGE and
GST_TAG_PPEVIEW_IMAGE used in the patch above).
2006-05-28 10:05:47 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: A track/volume number or count of 0 does not make sense, just ignore it along with negati...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3v2_tag_to_taglist):
A track/volume number or count of 0 does not make sense,
just ignore it along with negative numbers (a tag might
only contain a track count without a track number).
2006-05-19 14:05:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Don't output any tag when we encounter a negative track number - the tag type is uint, so...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3v2_tag_to_taglist):
Don't output any tag when we encounter a negative track number - the
tag type is uint, so we end up outputting huge positive numbers
instead. (Fixes: #342029)
2006-05-16 14:07:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Rework string parsing to always walk over BOM markers in UTF16 strings, using the endianness indicated by the innermost one ...
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_find_best):
Make the name of the child element be based on the name of the
parent, so that debug output is more useful.
* gst-libs/gst/tag/id3v2frames.c: (find_utf16_bom),
(parse_insert_string_field), (parse_split_strings):
Rework string parsing to always walk over BOM markers in UTF16
strings, using the endianness indicated by the innermost one,
then trying the opposite endianness if that fails to convert
to valid UTF-8. Fixes #341774
2006-05-12 08:21:37 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Some more debug info. No need to check whether the string returned by g_convert() is real...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_insert_string_field):
Some more debug info. No need to check whether the string
returned by g_convert() is really UTF-8 - either it is or
we get NULL returned.
2006-05-10 13:51:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Fix parsing of numeric genre strings some more, by ensuring that we only try and parse st...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3v2_genre_fields_to_taglist):
Fix parsing of numeric genre strings some more, by ensuring that
we only try and parse strings that a) Start with '(' and b) Consist
only of digits.
Also, when finding an escaping '((' sequence, bust it back to '(' by
swallowing the first parenthesis
2006-04-28 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Recognise and skip any byte order marker (BOM) in
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (has_utf16_bom),
(parse_split_strings):
Recognise and skip any byte order marker (BOM) in
UTF-16 strings.
2006-04-17 10:01:51 +0000 Alex Lancaster <alexlan@fedoraproject.org>
tag: id3v2: Recognise TCO (Genre) tags in ID3v2.2
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c:
Recognise TCO (Genre) tags in ID3v2.2. Patch by Alex Lancaster
(Fixes #338713)
2006-03-30 23:37:16 +0000 Sébastien Moutte <sebastien@moutte.net>
tag: id3v2: use of GST_DEBUG instead of DEBUG(a...) for WIN32
Original commit message from CVS:
* ext\jpeg\smokecodec.c:
use of GST_DEBUG instead of DEBUG(a...) for WIN32
* ext\speex\gstspeexenc.c: (gst_speexenc_set_header_on_caps):
move first instruction after all variables declarations
* gst\alpha\gstalpha.c:
* gst\effectv\gstshagadelic.c:
* gst\smpte\paint.c:
* gst\videofilter\gstvideobalance.c:
define M_PI if it's not defined (it's not defined on WIN32)
* gst\cutter\gstcutter.c: (gst_cutter_chain):
* gst\id3demux\id3v2frames.c: (parse_relative_volume_adjustment_two):
* gst\level\gstlevel.c: (gst_level_set_property), (gst_level_transform_ip):
* gst\matroska\matroska-demux.c: (gst_matroska_demux_parse_info),
(gst_matroska_demux_video_caps):
* gst\matroska\matroska-mux.c: (gst_matroska_mux_start), (gst_matroska_mux_finish):
* gst\wavparse\gstwavparse.c: (gst_wavparse_stream_data):
use gst_guint64_to_gdouble for conversions
* gst\goom\filters.c: (setPixelRGB_):
fix a debug which was using undefined variable
* gst\level\gstlevel.c: (gst_level_set_caps), (gst_level_transform_ip):
* gst\matroska\ebml-read.c: (gst_ebml_read_sint):
replace LL suffix with L suffix (LL isn't supported by MSVC6.0)
* win32/vs6:
add vs6 projects files for most of plugins-good
2006-03-22 13:00:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Don't attempt typefinding on too-short buffers that have been completely trimmed away.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
* gst-libs/gst/tag/gstid3demux.c: (gst_id3demux_chain):
Don't attempt typefinding on too-short buffers that have been
completely trimmed away.
* gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v2_tag):
Improve the debug output
2006-03-16 16:06:22 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: We only care about gain and peak data for the master volume.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c:
(parse_relative_volume_adjustment_two):
We only care about gain and peak data for the master volume.
2006-03-16 13:22:28 +0000 Tim-Philipp Müller <tim@centricular.net>
tag: id3v2: Read replay gain tags
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_id_string), (parse_unique_file_identifier),
(parse_relative_volume_adjustment_two), (id3v2_tag_to_taglist):
Read replay gain tags (#323721).
2006-03-14 17:56:02 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(), used by id3demux.
Original commit message from CVS:
* configure.ac:
Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(),
used by id3demux.
* gst-libs/gst/tag/gstid3demux.c: (plugin_init):
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_user_text_identification_frame),
(parse_unique_file_identifier):
Add support for UFID and TXXX frames and extract musicbrainz tags.
2006-02-18 20:48:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Handle 0 data size in otherwise valid frames.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
* gst-libs/gst/tag/id3v2frames.c: (id3v2_genre_fields_to_taglist):
Handle 0 data size in otherwise valid frames.
Handle numeric strings in 2.4.0 even when not in parentheses
2006-02-16 10:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: 3 2.3.0 used synch-safe integers for the tag size, but not for the frame size. (Fixes #331368)
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
ID3 2.3.0 used synch-safe integers for the tag size, but not for the
frame size. (Fixes #331368)
2006-02-13 12:00:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Add more validation to ensure that a char encoding conversion produced a valid UTF-8 string.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_insert_string_field),
(parse_split_strings):
Add more validation to ensure that a char encoding conversion
produced a valid UTF-8 string.
2006-02-04 13:30:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Adjust for data length indicators when parsing (Fixes #329810)
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_split_strings):
Adjust for data length indicators when parsing (Fixes #329810)
Fix stupid bug parsing UTF-8 tag text.
Output tag strings with multiple fields as multiple tags, so the
app gets all the data.
2006-02-03 13:06:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Never output a tag with a null contents string.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (parse_text_identification_frame),
(id3v2_tag_to_taglist), (id3v2_genre_string_to_taglist),
(id3v2_genre_fields_to_taglist):
Never output a tag with a null contents string.
2006-01-30 23:13:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Someone should kick my butt. Remove ID3v1 tags from the end of the file.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3demux.c: (gst_id3demux_chain),
(gst_id3demux_read_id3v1), (gst_id3demux_sink_activate),
(gst_id3demux_send_tag_event):
* gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v1_tag):
Someone should kick my butt. Remove ID3v1 tags from the end of the
file.
Improve error messages. Send the TAG message as soon as we complete
typefinding, instead of waiting until we send the first buffer.
Downstream tag event is still sent before the first buffer.
2006-01-25 18:23:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Never trust ANY information encoded in a media file, especially when it's giving you size...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame):
Never trust ANY information encoded in a media file, especially
when it's giving you sizes. (Fixes #328452)
2006-01-23 14:32:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Remove errant break statement, and fix compilation with older GCC.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3v2_tag_to_taglist):
Remove errant break statement, and fix compilation with
older GCC.
2006-01-23 09:22:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: Rewrite parsing of text tags to handle multiple NULL terminated strings. Parse numeric genre strings a...
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v2_tag):
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame), (parse_text_identification_frame),
(id3v2_tag_to_taglist), (id3v2_are_digits),
(id3v2_genre_string_to_taglist), (id3v2_genre_fields_to_taglist),
(parse_split_strings), (free_tag_strings):
Rewrite parsing of text tags to handle multiple NULL terminated
strings. Parse numeric genre strings and ID3v2 type
"(3)(6)Alternative" style genre strings.
Parse dates that are only YYYY or YYYY-mm format.
2006-01-15 20:21:48 +0000 Sergey Scobich <sergey.scobich@gmail.com>
tag: id3v2: Fix compilation of id3demux when zlib is not present.
Original commit message from CVS:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame):
Fix compilation of id3demux when zlib is not present.
(Fixes #326602; patch by: Sergey Scobich)
2006-01-06 11:46:53 +0000 Edward Hervey <bilboed@bilboed.com>
tag: id3v2: Add gst_element_no_more_pads() for proper decodebin behaviour.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3demux.c: (gst_id3demux_add_srcpad):
Add gst_element_no_more_pads() for proper decodebin behaviour.
* gst-libs/gst/tag/id3v2frames.c: (parse_comment_frame),
(parse_text_identification_frame), (parse_split_strings):
Failure to decode some tags is not a GST_ERROR() but a
GST_WARNING()
When iterating over a chunk of text, check that we haven't gone too
far.
2005-12-28 18:55:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: If a broken tag has 0 bytes payload, at least still skip the 10 byte header
Original commit message from CVS:
* gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v2_tag):
If a broken tag has 0 bytes payload, at least still skip
the 10 byte header
2005-12-18 15:14:44 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tag: id3v2: all new LGPL id3 demuxer, can use zlib for compressed frames
Original commit message from CVS:
* configure.ac:
Check for optional dependency on zlib for id3demux
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstid3demux.c: (gst_gst_id3demux_get_type),
(gst_id3demux_base_init), (gst_id3demux_class_init),
(gst_id3demux_reset), (gst_id3demux_init), (gst_id3demux_dispose),
(gst_id3demux_add_srcpad), (gst_id3demux_remove_srcpad),
(gst_id3demux_trim_buffer), (gst_id3demux_chain),
(gst_id3demux_set_property), (gst_id3demux_get_property),
(id3demux_get_upstream_size), (gst_id3demux_srcpad_event),
(gst_id3demux_read_id3v1), (gst_id3demux_read_id3v2),
(gst_id3demux_sink_activate), (gst_id3demux_src_activate_pull),
(gst_id3demux_src_checkgetrange), (gst_id3demux_read_range),
(gst_id3demux_src_getrange), (gst_id3demux_change_state),
(gst_id3demux_pad_query), (gst_id3demux_get_query_types),
(simple_find_peek), (simple_find_suggest),
(gst_id3demux_do_typefind), (gst_id3demux_send_tag_event),
(plugin_init):
* gst-libs/gst/tag/gstid3demux.h:
* gst-libs/gst/tag/id3v2.c: (read_synch_uint),
(id3demux_read_id3v1_tag), (id3demux_read_id3v2_tag),
(id3demux_id3v2_frame_hdr_size), (convert_fid_to_v240),
(id3demux_id3v2_frames_to_tag_list):
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/id3v2.4.0-frames.txt:
* gst-libs/gst/tag/id3v2.4.0-structure.txt:
* gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame), (parse_text_identification_frame),
(id3v2_tag_to_taglist), (parse_split_strings):
All new LGPL id3 demuxer. Can use zlib for compressed frames,
otherwise it discards them. Works on my test files.
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop):
Don't send EOS to a non-existing srcpad
The debug category can be static
2011-08-11 18:50:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
audioresample: fix quality setting being ignored by the resampler state
https://bugzilla.gnome.org/show_bug.cgi?id=636562
2011-08-11 15:54:15 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* configure.ac:
* gst/audioresample/resample.c:
* gst/audioresample/resample_sse.h:
* gst/audioresample/speex_resampler_double.c:
* gst/audioresample/speex_resampler_float.c:
audioresample: use SSE/SSE2 when possible
Compile in the code on i386 and x86_64, and use ORC to determine
when the runtime platform can run the code.
https://bugzilla.gnome.org/show_bug.cgi?id=636562
2011-08-11 19:23:42 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/audioresample/resample_sse.h:
audioresample: fix SSE2 building with double precision
The full double implementation was missing.
https://bugzilla.gnome.org/show_bug.cgi?id=636562
2011-08-11 12:12:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Check for utf8 before trying to convert
If the string is already on utf8, there is no need to
try to convert it, because it is useless and it might garble
the string.
2011-08-10 13:16:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/tag.c:
tests: tag: exif: Add tests for 'non-trivial' chars
Adds two new cases to check that characters are properly
converted to ascii when writen to exif and parsed correctly
back to utf8 when read.
2011-08-09 16:02:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Exif strings should be ascii
Use g_convert to turn all strings into extended ascii before writing
to the exif buffer and converting back from ascii to utf8 when
reading them.
2011-08-10 15:57:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgsttag.def:
win32: update libgsttag.def for new API
2011-08-10 15:21:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
tag: don't build helper programs that generate/update data by default
No point building these by default. Also, these generated files
should go into the srcdir, not the builddir in this case, since
they're version controlled.
2011-08-10 15:20:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/mklicensestables.c:
tag: fix stray printf in mklicensestables
Don't dump debug output to stdout.
2011-08-10 15:06:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/licenses.c:
tag: fix compilation of new licenses code with GLib versions < 2.28
Add local g_variant_lookup_value() fallback for now when compiling
against older GLib versions.
2011-08-10 14:57:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/licenses.c:
* gst-libs/gst/tag/tag.h:
tag: add GType for GstTagLicenseFlags
API: gst_tag_license_flags_get_type()
2011-08-10 10:49:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: fix runtime warnings when doing position query
Add missing 'break'.
2011-07-15 13:19:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/libs/tag.c:
* tests/files/Makefile.am:
* tests/files/license-uris:
tag: add unit test for new license API
https://bugzilla.gnome.org/show_bug.cgi?id=646868
2011-07-15 13:14:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/mklicensestables.c:
tag: add mklicensestables utility
Add (uninstalled) tool to create licenses-table.dat from liblicense's
RDF files. It's not very pretty and makes loats of assumptions about
the input, but should work. If things change, we can fix it then.
https://bugzilla.gnome.org/show_bug.cgi?id=646868
2011-07-15 13:07:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/license-translations.dict:
* gst-libs/gst/tag/licenses-tables.dat:
* gst-libs/gst/tag/licenses.c:
* gst-libs/gst/tag/tag.h:
tag: add convenience API to handle creative commons licenses
Based on liblicense's RDF files.
API: GstTagLicenseFlags
API: gst_tag_get_licenses()
API: gst_tag_get_license_flags()
API: gst_tag_get_license_nick()
API: gst_tag_get_license_title()
API: gst_tag_get_license_version()
API: gst_tag_get_license_description()
API: gst_tag_get_license_jurisdiction()
https://bugzilla.gnome.org/show_bug.cgi?id=646868
2011-08-08 10:00:40 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: bump probability if all frames we found are similar
Similar meaning same layer, same bitrate, and same number of channels
This fixes misdetection of (some MP3 files that have zero padding
between the ID3 tag and the MP3 stream) as H.264 video.
https://bugzilla.gnome.org/show_bug.cgi?id=656018
2011-08-05 16:53:47 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
gstvorbistag: map ENCODER Vorbis comment to application-name
What GStreamer calls encoder ("encoder used to encode this stream") is
stored in the vendor string in Vorbis/Theora/Kate and possibly others.
The Vorbis comment packet used in those streams uses ENCODER as the name
of the encoding program, which GStreamer calls application-name.
https://bugzilla.gnome.org/show_bug.cgi?id=656034
2011-08-05 11:32:09 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: fix sample depth typo
https://bugzilla.gnome.org/show_bug.cgi?id=656022
2011-08-05 13:05:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc-dist.c:
volume: Update disted ORC files
2011-08-03 14:14:55 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Set queues to silent=true
As encodebin doesn't connect to the queue signals, it can set
queues to silent mode to make queue not emit them.
Check https://bugzilla.gnome.org/show_bug.cgi?id=621299 for
more info on queue's silent property.
2011-08-03 13:40:19 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Fix typo on installing properties
queue buffers and bytes properties have ids swapped, fix it.
2011-08-03 10:18:29 +0200 Jonathan Liu <net147@gmail.com>
* ext/ogg/gstoggstream.c:
oggstream: Fix crashes with 0-byte vorbis packets
Fixes bug #655574.
2011-07-28 14:43:53 +0200 Jens Georg <jensg@openismus.com>
* gst-libs/gst/pbutils/codec-utils.c:
pbutils: Add SP levels 4a, 5 and 6
https://bugzilla.gnome.org/show_bug.cgi?id=655503
2011-07-26 16:10:17 +0200 Philip Jägenstedt <philipj@opera.com>
* ext/theora/gsttheoradec.c:
theoradec: segfault on 0-byte ogg_packet in _chain_reverse
2011-07-29 10:23:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* win32/common/libgsttag.def:
Add new GstTagMux base class
Hook up new tag muxing base class to build system.
https://bugzilla.gnome.org/show_bug.cgi?id=555437
API: GstTagMux
2011-07-29 10:22:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gsttagmux.c:
* gst-libs/gst/tag/gsttagmux.h:
docs: add documentation for GstTagMux
2011-07-28 20:38:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gsttagmux.c:
tagmux: require subclass to install sink pad template
Require the subclass to install both source and sink pad
templates. Also, print some warnings if the subclass doesn't
do that.
https://bugzilla.gnome.org/show_bug.cgi?id=555437
2011-07-15 20:57:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gsttagmux.h:
tagmux: const-ify GstTagList argument of render vfuncs
2011-07-15 20:39:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gsttagmux.c:
* gst-libs/gst/tag/gsttagmux.h:
tagmux: fix up private base class header so it can be made public
Move private bits into a private struct, add some padding.
https://bugzilla.gnome.org/show_bug.cgi?id=555437
2011-07-28 23:31:03 +0100 Michael Smith <msmith@songbirdnest.com>
* gst-libs/gst/tag/gsttagmux.c:
* gst-libs/gst/tag/gsttagmux.h:
tagmux: add support for end tags
Originally "id3tag: Add new id3 tagging plugin, supports v1, v2.3,
and v2.4." from gst-plugins-bad. This is an artificial bridge commit.
2010-06-06 18:00:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gsttagmux.c:
ext: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs
2007-11-20 11:41:13 +0000 Julien Moutte <julien@moutte.net>
Fix build on Mac OS X 10.5
Original commit message from CVS:
2007-11-20 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/tag/gsttagmux.c: (gst_tag_lib_mux_render_tag),
(gst_tag_lib_mux_adjust_event_offsets):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
* sys/osxaudio/Makefile.am:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5
2007-09-13 15:04:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Update my mail address.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstapev2mux.h:
* gst-libs/gst/tag/gsttagmux.c:
* tests/check/elements/apev2mux.c:
Update my mail address.
2006-05-30 14:35:18 +0000 Sebastian Dröge <mail@slomosnail.de>
Add apev2mux element (#343122).
Original commit message from CVS:
Patch by: Sebastian Dröge <mail at slomosnail de >
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* ext/taglib/Makefile.am:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstapev2mux.h:
* ext/taglib/gstid3v2mux.cc:
* gst-libs/gst/tag/gsttagmux.c: (plugin_init):
* gst-libs/gst/tag/gsttagmux.h:
Add apev2mux element (#343122).
* tests/check/Makefile.am:
* tests/check/elements/apev2mux.c:
(test_taglib_apev2mux_create_tags),
(test_taglib_apev2mux_check_tags), (fill_mp3_buffer), (got_buffer),
(demux_pad_added), (test_taglib_apev2mux_check_output_buffer),
(test_taglib_apev2mux_with_tags), (GST_START_TEST),
(apev2mux_suite), (main):
Add unit test for apev2mux element.
2006-05-18 12:46:08 +0000 James Doc Livingston <doclivingston@gmail.com>
gst-libs/gst/tag/gsttagmux.c: Merge event tags and tag setter tags correctly (#339918). Also, don't leak taglist in case...
Original commit message from CVS:
Patch by: James "Doc" Livingston <doclivingston gmail com>
* gst-libs/gst/tag/gsttagmux.c: (gst_tag_lib_mux_render_tag):
Merge event tags and tag setter tags correctly (#339918). Also,
don't leak taglist in case of an error.
2006-05-01 11:46:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
docs/plugins/Makefile.am: also check .cc files for gtk-doc markup
Original commit message from CVS:
* docs/plugins/Makefile.am:
also check .cc files for gtk-doc markup
* configure.ac:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* tests/check/Makefile.am:
* tests/check/elements/id3v2mux.c: (id3v2mux_suite), (main):
* ext/Makefile.am:
* ext/taglib/Makefile.am:
* ext/taglib/gstid3v2mux.h:
* gst-libs/gst/tag/gsttagmux.c:
* gst-libs/gst/tag/gsttagmux.h:
move taglib-based id3v2muxer to -good. Fixes #336110.
2006-04-30 16:16:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-libs/gst/tag/gsttagmux.c:
small cleanups
Original commit message from CVS:
small cleanups
2006-04-29 18:46:36 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/gsttaglib.cc: Post an error message on the bus in the (extremely unlikely) case of an error.
Original commit message from CVS:
* ext/taglib/gsttaglib.cc:
Post an error message on the bus in the (extremely unlikely)
case of an error.
2006-04-29 18:18:24 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/: Split the actual ID3v2 tag rendering code into its own subclass.
Original commit message from CVS:
* ext/taglib/Makefile.am:
* ext/taglib/gstid3v2mux.cc:
* ext/taglib/gstid3v2mux.h:
* ext/taglib/gsttaglib.cc:
* ext/taglib/gsttaglib.h:
Split the actual ID3v2 tag rendering code into
its own subclass.
2006-04-28 15:33:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-libs/gst/tag/gsttagmux.c:
* gst-libs/gst/tag/gsttagmux.h:
pedantic cleanups
Original commit message from CVS:
pedantic cleanups
2006-04-01 16:50:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-libs/gst/tag/gsttagmux.c:
add taglib checks and docs
Original commit message from CVS:
add taglib checks and docs
2006-03-26 19:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/gsttaglib.*: Fix newsegment event handling a bit. We need to cache the first newsegment event, because we ...
Original commit message from CVS:
* ext/taglib/gsttaglib.cc:
* ext/taglib/gsttaglib.h:
Fix newsegment event handling a bit. We need to
cache the first newsegment event, because we can't
adjust offsets yet when we get it, as we don't
know the size of the tag yet for sure at that point.
Also do some minor cleaning up here and there and add
some debug statements.
2006-03-25 21:57:24 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/gsttaglib.cc: We do not want to proxy the caps on the sink pad; our source pad should have application/x-i...
Original commit message from CVS:
* ext/taglib/gsttaglib.cc:
We do not want to proxy the caps on the sink pad; our
source pad should have application/x-id3 caps; also,
don't use already-freed strings in debug messages;
finally, adjust buffer offsets on buffers sent out.
2006-03-20 08:59:29 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/gsttaglib.h: Fix left-over gst_my_filter_get_type.
Original commit message from CVS:
* ext/taglib/gsttaglib.h:
Fix left-over gst_my_filter_get_type.
2006-03-13 17:22:19 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/gsttaglib.cc: Add gtk-doc blurb (unused for the time being); match registered plugin name to the filename ...
Original commit message from CVS:
* ext/taglib/gsttaglib.cc:
Add gtk-doc blurb (unused for the time being); match registered
plugin name to the filename of the plugin (taglibmux => taglib)
2006-03-12 15:02:02 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/taglib/: Add support for writing MusicBrainz IDs.
Original commit message from CVS:
* ext/taglib/Makefile.am:
* ext/taglib/gsttaglib.cc:
* ext/taglib/gsttaglib.h:
Add support for writing MusicBrainz IDs.
2006-03-11 10:58:08 +0000 Alex Lancaster <alexlan@fedoraproject.org>
ext/taglib/gsttaglib.cc: and add support for TCOP (copyright)
Original commit message from CVS:
2006-03-11 Christophe Fergeau <teuf@gnome.org>
Patch by: Alex Lancaster
* ext/taglib/gsttaglib.cc: fix writing of TPOS tags (album number),
and add support for TCOP (copyright)
2006-03-09 17:44:17 +0000 Christophe Fergeau <teuf@gnome.org>
new id3v2 muxer based on TagLib
Original commit message from CVS:
2006-03-09 Christophe Fergeau <teuf@gnome.org>
reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* configure.ac:
* ext/Makefile.am:
* ext/taglib/Makefile.am:
* ext/taglib/gsttaglib.cc:
* ext/taglib/gsttaglib.h: new id3v2 muxer based on TagLib
2011-07-28 11:21:26 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: rename flags names
Rename flags names from native-audio/-video to
no-audio/video-conversion to be more explicit on what it does
2011-07-20 18:10:57 +0200 Stefan Sauer <ensonic@google.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: fix latency calculation for live elements
Max_latency was computed on already adjusted min_latency. Introduce a new
variable for clarity. Spotted by Blaise Gassend.
Fixes #644284
2011-07-28 11:44:20 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: fix max latency calculation
... to allow infinite max, as also claimed by comment.
2011-06-01 10:21:39 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: drop samples that are too late
... rather than having all of them rendered at 0 or subsequently aligned,
likely inevitably leading to repeated resyncing.
2011-07-26 13:51:31 +0200 Stefan Sauer <ensonic@google.com>
* tests/check/pipelines/basetime.c:
basetime: fix failing test
Always use audiotestsrc as it seems to have been the intention according to the
comment header. The test does not work with live-audiosources.
2011-07-25 19:51:24 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/check/elements/playbin2-compressed.c:
tests: rename the test suite to match the binary
This unbreaks determining the name for make elements/playbin2-compressed.check
from the test output.
2011-07-25 19:39:55 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
adder: rework pending event handling
Use atomic ops on pending flags. Rename the segment_pending to
new_segment_pending. Set new_segment_pending not when we received seek, but
when we received the first upstream new_segment.
2011-07-25 19:11:59 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: more debug logging for events
2011-07-26 12:33:56 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Allow all EOS to go through if we don't have a next group
Only drop them if the current group isn't drained .. AND there is a
next group to switch to.
Should Fix #655268
2011-07-25 18:37:15 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Avoid resetting playsink when not needed
When we don't have specific {audio|video|text}-sink properties, don't
set them on playsink when reconfiguring.
If we do that, we end up setting the previous configured sink to
GST_STATE_NULL resulting in any potentially pending push being returned
with GST_FLOW_WRONG_STATE which will cause the upstream elements to
silently stop.
https://bugzilla.gnome.org/show_bug.cgi?id=655279
2011-07-25 12:04:02 +0200 Stefan Sauer <ensonic@google.com>
* ext/pango/gsttextoverlay.c:
textoverlay: improve the example
Mentioned that this is not ment to be used with subtitles and suggest alternatives.
2011-07-25 10:41:04 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Properly handle multi-stream chains
When we have a multi-stream (i.e. audio and video) input and the demuxer
adds/removes pads for a new stream (common in a mpeg-ts stream when the
program stream mapping is updated), the algorithm for EOS handling was
previously wrong (it would only drop the EOS of the *last* pad but would
let the EOS on the other pads go through).
The logic has only been changed a tiny bit for EOS handling resulting in:
* If there is no next group, let the EOS go through
* If there is a next group, but not all pads are drained in the active
group, drop the EOS event
* If there is a next group and all pads are drained, then the ghostpads
will be removed and the EOS event will be dropped automatically.
2011-07-23 14:21:27 +0200 Stefan Sauer <ensonic@google.com>
* ext/pango/gsttextoverlay.c:
textoverlay: add example for feeding from stdin
2011-07-23 13:46:31 +0200 Stefan Sauer <ensonic@google.com>
* tests/check/pipelines/basetime.c:
test: print actual timestamp on failure
2011-07-20 13:46:31 +0200 Stefan Sauer <ensonic@google.com>
* ext/pango/gsttextoverlay.c:
textoverlay: keep untimestamped textbuffer until next one
Instead of discarding untimestamped text-buffers immeditely after rendering,
keep them until we receive the next text buffer.
Fixes #654959
2011-07-15 16:46:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/decodebin2.c:
tests: add decodebin2 test for parser autoplugging
Make sure decodebin2 doesn't try to plug the same parser twice
in a row.
2011-07-06 19:40:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/decodebin.c:
* tests/files/Makefile.am:
* tests/files/test.mp3:
tests: add decodebin1 test for parser autoplugging
Make sure decodebin1 doesn't try to plug the same parser twice
in a row (so we can change all parsers to accept parsed input as
well without breaking applications still using the old decodebin1
element).
2011-07-07 15:02:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstdecodebin.c:
decodebin: don't plug the same parser multiple times in a row
This allows us to make parsers accept both parsed and unparsed input
without decodebin plugging them in a loop until things blow up, ie.
without affecting applications that still use the old playbin or the
old decodebin.
(Making parsers accept parsed input is useful for later when we want
to use parsers to convert the stream-format into something the decoder
can handle. It's also much more convenient for application authors
who can plug parsers unconditionally in transcoding pipelines, for
example).
2011-07-14 13:56:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
* win32/common/libgstpbutils.def:
docs: add Since marker to gtk-doc chunk for new codec utils API
And add new API to .def file.
API: gst_codec_utils_h264_get_level_idc()
2011-03-07 17:55:48 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/codec-utils.c:
* gst-libs/gst/pbutils/codec-utils.h:
codec-utils: Add method to convert H.264 text level in a level_idc
2011-07-09 18:33:38 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggmux.c:
oggmux: check for EOS on both current and best pad
Oops, need both. Fixes #654270.
2011-07-09 18:24:26 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggmux.c:
oggmux: check for EOS on current pad, not best
Fixes #654270.
2011-07-09 11:59:42 +0200 Piotr Fusik <fox@scene.pl>
* gst/typefind/gsttypefindfunctions.c:
typefind: fixed detection of audio/x-sap
Fixes: #654295.
Signed-off-by: David Schleef <ds@schleef.org>
2011-06-30 20:33:36 +0200 Luis de Bethencourt <luis@debethencourt.com>
* gst/encoding/gstencodebin.c:
encodebin: fix compiler warning
cspace and cspace2 may run uninitialized.
2011-06-29 13:12:49 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Add flags to disable conversion elements
Add a flags property and two flags to allow one to disable the
conversion elements within encodebin. Doing so insists that the
uncompressed input to encodebin for the appropriate stream type is
sufficient to meet the caps requirements of the encoders, muxers and
encodebin target.
This is mostly beneficial to bypass slow caps negotiations in the
conversion elements.
2011-06-29 09:59:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Remove extra chars from end of xmp packet
Windows picture viewer is unhappy with extra trailing chars at the
end of the xmppacket footer. So remove them as they aren't needed.
2011-06-29 11:30:51 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/encoding/gststreamsplitter.c:
streamsplitter: Fix getcaps src pad caps merge
Caps returned from gst_pad_peer_get_caps_reffed () may not be writable.
If they are not is should cause an assertion in gst_caps_merge (),
however, sometimes assertions are disabled in binary builds of -base and
it's safer to just be sure the caps are writable. Also, check that the
reffed caps pointer is not NULL.
2011-06-15 13:51:31 +0200 Philip Jägenstedt <philipj@opera.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: NULL check in degas_type_find
The length check isn't sufficient, an source might
report the correct length, but then still fail to
read the requested number of bytes for some reason.
https://bugzilla.gnome.org/show_bug.cgi?id=652642
2011-06-26 01:06:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/design/design-decodebin.txt:
docs: minor addition to decodebin2 design doc
2011-06-26 01:06:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/navigation.c:
tests: the navigation interface isn't GstImplementsInterface-wrapped
2011-06-26 00:49:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/streamvolume.h:
interfaces: GstStreamVolume isn't wrapped by GstImplementsInterface
This interface depends on properties and isn't per-instance.
2011-06-26 00:40:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspextension.h:
rtsp: GstRTSPExtension isn't wrapped by GstImplementsInterface
Fix copy'n'paste error in headers, GstRTSPExtension isn't
something that's per-instance.
2011-06-26 00:36:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/xmpwriter.h:
tag: GstXmpWriter doesn't use the GstImplementsInterface
No need for per-instance checking of interface implementation here,
presumably just a copy'n'paste issue.
2011-06-11 19:03:57 +1000 Jonathan Matthew <jonathan@d14n.org>
* gst-libs/gst/pbutils/encoding-target.c:
encoding-target: set names on audio and video profiles
https://bugzilla.gnome.org/show_bug.cgi?id=652342
2011-06-23 11:28:04 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From 69b981f to 605cd9a
2011-06-18 13:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
Bump git version after unplanned 0.10.35 release
Merge branch '0.10.35'
Conflicts:
configure.ac
docs/plugins/inspect/plugin-adder.xml
docs/plugins/inspect/plugin-alsa.xml
docs/plugins/inspect/plugin-app.xml
docs/plugins/inspect/plugin-audioconvert.xml
docs/plugins/inspect/plugin-audiorate.xml
docs/plugins/inspect/plugin-audioresample.xml
docs/plugins/inspect/plugin-audiotestsrc.xml
docs/plugins/inspect/plugin-cdparanoia.xml
docs/plugins/inspect/plugin-decodebin.xml
docs/plugins/inspect/plugin-encoding.xml
docs/plugins/inspect/plugin-ffmpegcolorspace.xml
docs/plugins/inspect/plugin-gdp.xml
docs/plugins/inspect/plugin-gio.xml
docs/plugins/inspect/plugin-gnomevfs.xml
docs/plugins/inspect/plugin-libvisual.xml
docs/plugins/inspect/plugin-ogg.xml
docs/plugins/inspect/plugin-pango.xml
docs/plugins/inspect/plugin-playback.xml
docs/plugins/inspect/plugin-subparse.xml
docs/plugins/inspect/plugin-tcp.xml
docs/plugins/inspect/plugin-theora.xml
docs/plugins/inspect/plugin-typefindfunctions.xml
docs/plugins/inspect/plugin-uridecodebin.xml
docs/plugins/inspect/plugin-videorate.xml
docs/plugins/inspect/plugin-videoscale.xml
docs/plugins/inspect/plugin-videotestsrc.xml
docs/plugins/inspect/plugin-volume.xml
docs/plugins/inspect/plugin-vorbis.xml
docs/plugins/inspect/plugin-ximagesink.xml
docs/plugins/inspect/plugin-xvimagesink.xml
gst-libs/gst/audio/Makefile.am
gst/subparse/gstsubparse.c
win32/common/_stdint.h
win32/common/config.h
2011-06-18 11:16:19 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Allow GError* argument to be NULL
This is how other methods taking GError* arguments behave.
Fixes #652838
=== release 0.10.35 ===
2011-06-15 19:29:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
Release 0.10.35
This is an ad-hoc release that is almost identical to 0.10.34:
* work around GLib atomic ops API change
* don't use G_CONST_RETURN in public headers
* subparse: typefinding fixes for subtitles in non-UTF8 charsets
2011-06-15 15:08:32 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Add gobject introspection files to spec
2011-06-15 14:53:56 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
remove old v4l plugin from spec file
2011-06-15 14:49:41 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* tests/examples/Makefile.am:
Add missing dist subdir
2011-06-15 14:21:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/Makefile.am:
audio: link test program against libgstaudio
2011-06-14 10:31:18 +0530 Debarshi Ray <rishi@gnu.org>
* gst-libs/gst/pbutils/codec-utils.c:
codec-utils: restore 7350 as a valid sampling frequency for AAC
This was lost during c77f88cac675a1dbb89e40da8e3c28320523bfca.
2011-06-09 18:30:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tag.h:
libs: replace G_CONST_RETURN with 'const'
G_CONST_RETURN will be deprecated soon.
https://bugzilla.gnome.org/show_bug.cgi?id=652211
2011-05-31 22:14:09 -0700 David Schleef <ds@schleef.org>
* gst/audioresample/resample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/videoscale/vs_4tap.c:
* gst/videotestsrc/generate_sine_table.c:
* gst/videotestsrc/videotestsrc.c:
* tests/icles/test-xoverlay.c:
convert M_PI to G_PI, for msvc
2011-06-06 14:41:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: no input data implies no type can be found
... and posting a proper error message to this effect is appropriately
informative and prevents auto-plugging otherwise stalling.
2011-06-04 13:36:55 -0700 David Schleef <ds@schleef.org>
* gst/adder/gstadder.c:
adder: Work around changes in g_atomic API
See #651514 for details.
2011-05-31 20:38:56 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix c99-ism
2011-05-23 16:02:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Try to typefind even if conversion to UTF8 failed
Fixes bug #600043.
2011-05-23 15:51:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Interprete typefind strings passed to GRegex as raw bytes instead of valid UTF8
2011-05-20 10:48:39 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/lang.c:
lang: fix possible array overrun
We where checking for i<G_N_ELEMENTS, but where accessing i+1.
2011-06-14 10:31:18 +0530 Debarshi Ray <rishi@gnu.org>
* gst-libs/gst/pbutils/codec-utils.c:
codec-utils: restore 7350 as a valid sampling frequency for AAC
This was lost during c77f88cac675a1dbb89e40da8e3c28320523bfca.
2011-05-31 22:14:09 -0700 David Schleef <ds@schleef.org>
* gst/audioresample/resample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/videoscale/vs_4tap.c:
* gst/videotestsrc/generate_sine_table.c:
* gst/videotestsrc/videotestsrc.c:
* tests/icles/test-xoverlay.c:
convert M_PI to G_PI, for msvc
2011-06-07 21:30:18 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: refactor how EOS is determined
This decreases the number of buffers held on each pad by one,
eliminating next_buffer. Simplifies the logic by relying solely
on CollectPads to let us know when a pad is in EOS. As a side
benefit, the collect pads related code is structured more like
other CollectPad users.
The previous code would occasionally mark the wrong pad as EOS,
causing the code to get in a state where all the streams were
finished, but EOS hadn't been sent to the source pad.
2011-06-09 18:30:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tag.h:
libs: replace G_CONST_RETURN with 'const'
G_CONST_RETURN will be deprecated soon.
https://bugzilla.gnome.org/show_bug.cgi?id=652211
2011-06-09 00:02:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Fix LocationShown syntax
According to the specification, the LocationShown requires its
struct fields to be inside a Bag type.
2011-06-08 14:21:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Add room for extra namespace definitions
Adds an extra field to the namespace definitions of the schemas
so they can add the namespace of any array/struct fields they
might use internally.
2011-06-08 12:21:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/pango/Makefile.am:
* gst/audioresample/Makefile.am:
* tests/check/Makefile.am:
* tests/examples/v4l/Makefile.am:
GST_PLUGINS_BASE_LIBS is not defined in -base.
2011-06-08 11:33:07 +0200 Christophe Fergeau <cfergeau@redhat.com>
* tests/examples/audio/Makefile.am:
examples: don't link testchannels example with system libgstaudio
The testchannels audio test program is using -lgstaudio-0.10 to link
with libgstaudio which won't use the gstaudio library that was just
built but the one from the system. This is an issue since it means
we won't be testing the code from the current source tree, and it
also breaks the build when building on a system which don't have
a libgstaudio yet.
https://bugzilla.gnome.org/show_bug.cgi?id=652100
2011-06-08 11:11:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/design/design-decodebin.txt:
docs: add some text about parser/decoder autoplugging issues
2011-06-06 14:41:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: no input data implies no type can be found
... and posting a proper error message to this effect is appropriately
informative and prevents auto-plugging otherwise stalling.
2011-06-06 12:48:23 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: determine granulepos metadata using stream mapper whenever possible
... which unfortunately is not the case for all types, but at least so for
most common ones.
2011-06-06 12:46:05 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: convert incoming buffer timestamp to running time
... so all subsequent manipulation can take place in the proper timeline
without further ado.
2011-06-01 20:48:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: remove superfluous code
... since there is nothing in oggstream that cares (or even should)
about granulepos for what is being asked from it.
2011-06-04 13:36:55 -0700 David Schleef <ds@schleef.org>
* gst/adder/gstadder.c:
adder: Work around changes in g_atomic API
See #651514 for details.
2011-05-31 20:38:56 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: fix c99-ism
2011-06-03 16:29:00 +0200 Luis de Bethencourt <luis.debethencourt@collabora.como>
* ext/theora/gsttheoraenc.c:
theora: separate encode and push block in chain, into own function.
2011-06-02 19:08:41 +0200 Luis de Bethencourt <luis.debethencourt@collabora.como>
* ext/theora/gsttheoraenc.c:
theora: use fixed src cap pads
2011-06-02 18:57:05 +0200 Luis de Bethencourt <luis.debethencourt@collabora.como>
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraenc.h:
theora: set the width/height/par on the srcpad caps
2011-06-02 17:29:53 +0200 Luis de Bethencourt <luis.debethencourt@collabora.como>
* ext/theora/gsttheoraenc.c:
theora: get sink caps info from downstream element pad
https://bugzilla.gnome.org/show_bug.cgi?id=651564
2011-05-27 14:41:39 -0700 Patrick McCarty <patrick.mccarty@intel.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for wbmp images.
https://bugzilla.gnome.org/show_bug.cgi?id=651294
2011-06-02 00:55:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: add typefinder for WAP WBMP bitmaps
https://bugzilla.gnome.org/show_bug.cgi?id=651294
2011-06-02 11:53:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkvideoconvert.c:
playsink: Fix deadlock in the audio/video converter bins when linking fails
2011-06-01 17:31:35 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* tests/check/Makefile.am:
check: ... and don't forget to add the new arm header
Forgot it in my previous commit
2011-06-01 17:24:30 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* tests/check/libs/libsabi.c:
* tests/check/libs/struct_arm.h:
libsabi: Add structure sizes for arm
2011-05-31 19:57:57 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts32.c:
fft: s/M_PI/G_PI/ for MSVC
2011-05-31 11:05:03 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
* tests/check/elements/volume.c:
volume: Fix handling of volume>=4.0 for 8 and 16 bit integer formats
Also add a unit test for this. Previously volumes bigger than 4.0
would have resulted in overflows in the fixed point processing.
Fixes bug #649642.
2011-05-29 13:32:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/adder.c:
* tests/check/elements/ffmpegcolorspace.c:
* tests/check/elements/vorbistag.c:
* tests/check/libs/rtp.c:
* tests/check/pipelines/theoraenc.c:
tests: fix some more unused-but-set-variable warnings with gcc 4.6
2011-05-28 16:14:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstvideo.def:
win32: update .def file for new API
2011-05-28 12:39:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* Makefile.am:
* tests/check/elements/.gitignore:
Ignore new playbin2-compress test binary
And add old testchannels binary to CRUFT_FILES.
2011-05-27 23:31:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.h:
video: sprinkle some G_GNUC_CONST
Mark functions that have no effect besides their return value and
only inspect their input arguments with G_GNUC_CONST. (We just
ignore the g_return_val_if_fail() guards for this)
2011-05-27 23:25:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.h:
video: clean up header file
Sprinkle some spaces and newlines here and there.
2011-05-27 15:03:19 +0300 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* gst-libs/gst/audio/.gitignore:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/testchannels.c:
* tests/examples/Makefile.am:
* tests/examples/audio/.gitignore:
* tests/examples/audio/Makefile.am:
* tests/examples/audio/testchannels.c:
audio: move testchannels example to 'tests/examples' dir
Also fix it up a little to not include 'c' file but link to the libs instead.
2011-05-27 11:39:21 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/codec-utils.c:
code-utile: fix level descriptions for fgs
fgs levels range from 8-13 and are mapped to 0-5.
2011-05-25 14:38:21 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/codec-utils.c:
codec-utils: fix mpeg4 level verification
The current condition would never be true. As levels<6 are asp and levels>7 and
<14 are fgs, we should return NULL for cases 6,7,14,15.
2011-05-26 12:33:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
xvimagesink: Fallback to non-XShm mode if allocating the XShm image failed
Fixes bug #630442.
2011-05-26 12:30:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/ximage/ximagesink.c:
ximagesink: Fallback to non-XShm mode if allocating the XShm image failed
Fixes bug #630442.
2011-05-26 11:41:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Let the input-selectors sync all streams to the running time
This is especially needed when switching between a non-sparse and sparse
video stream, see bug #537382. It also lowers the time needed for switching
between streams a bit.
2011-01-20 00:52:50 -0700 Lane Brooks <dirjud@gmail.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: added 'outline-color' parameter to control whether text gets a shadow
2011-01-20 00:42:39 -0700 Lane Brooks <dirjud@gmail.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: added 'shadow' option to control whether text gets a shadow
2011-05-26 10:48:05 +0200 Jindrich Makovicka <makovick@gmail.com>
* ext/pango/gsttextrender.c:
textrender: Correctly negotiate with downstream instead of just using random caps
Fixes bug #638897.
2011-05-26 10:43:51 +0200 Jindrich Makovicka <makovick@gmail.com>
* ext/pango/gsttextrender.c:
textrender: Add bound checks to not write outside the image area
2011-05-26 10:42:46 +0200 Jindrich Makovicka <makovick@gmail.com>
* ext/pango/gsttextrender.c:
textrender: Prevent double unref of caps if the caps can't be set on the srcpad
2011-05-26 10:31:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gnomevfs/gstgnomevfssrc.c:
gnomevfssrc: Keep track of interruptions during read with a flag
2010-09-03 09:11:30 -0400 American Dynamics <GStreamer-Bugs@tycosp.com>
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
gnomevfssrc: Add support for cancelling the read operations
This allows the state change from PAUSED to READY to be faster.
Fixes bug #628337.
2011-05-25 14:14:46 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
* sys/ximage/ximagesink.c:
ximagesink: Remove g_assert from interface query
2011-05-25 14:08:43 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
* sys/xvimage/xvimagesink.c:
xvimagesink: Remove the g_assert from interface query
2011-05-26 00:17:40 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
audiotestsrc: add blue and violet noise by using spectral inversion
Add blue and violet noise by spectral inversion of pink and red noise.
Fixes #649969
2011-05-25 23:40:26 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
audiotestsrc: add red (brownian) noise generator
Add another noise generator which produces a quite dark noise color.
Fixes parts of #649969.
2010-09-27 13:32:31 +0400 Vladimir Eremeev <eremeev@atlantis.ru>
* tests/examples/seek/seek.c:
seek: set selected/default audio/video sinks on playbin and playbin2
https://bugzilla.gnome.org/show_bug.cgi?id=630322
2011-05-25 19:03:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: add --audiosink and --videosink command line options
2011-05-25 18:50:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: use the right GDK defines to differentiate between the backends
2011-05-25 18:45:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: use gst_filename_to_uri() to convert a filename to a uri
2010-09-27 12:46:54 +0400 Vladimir Eremeev <eremeev@atlantis.ru>
* tests/examples/seek/seek.c:
seek: make seek example work in win32
https://bugzilla.gnome.org/show_bug.cgi?id=630322
2011-05-25 16:08:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: update GLib requirement to >= 2.24
Same as core (make implicit requirement explicit).
http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
2011-05-25 15:24:33 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/theora/gsttheoraenc.c:
theoraenc: remove bogus <0 check for unsigned var
bytes_written is a gsize which is unsigned and thus never < 0.
2011-05-25 15:23:13 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/theora/gsttheoraenc.c:
theoraenc: fix variable type for bytes_consumed
th_encode_ctl() returns an int. Using a gsize result in bogus <0 checks.
2011-05-25 15:04:20 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/riff/riff-read.c:
riff: remove the g_return_if_fail as we test it below
We don't want to return without setting taglist=NULL if asserts are on and with
setting taglist=NULL otherwise.
2011-05-25 14:28:18 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/volume/gstvolume.c:
volume: use a flag for 'mute' using the controller
Previously we checked mute_csource to determine wheter we need to premultiply
volumes and mute values. That fails as we unrefs mute_csource and set it to
NULL after. Use an extra flag instead.
2011-05-25 14:12:50 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/gstexiftag.c:
exiftag: reflow the code
Move the warning on unsupported units to the swicth-case. Move fetching the
pending tags down to where we use them.
2011-05-25 13:59:57 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/gstexiftag.c:
exiftag: set value=1 if we found the token
Otherwise we never write the tag. This would also be consistent with the code in
deserialize_scene_type().
2011-05-25 12:30:51 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: run gst-indent
2011-05-25 12:29:21 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: remove unneded !=NULL checks
We check for matching_attr!=NULL right before already.
2011-05-24 00:13:04 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: massage the section file more
Add more symbols (from unused.txt). Move the whole bunch of riff-fourcc defines
to std section too (no one is hoing to document them, right).
2011-05-24 00:12:26 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/video.c:
docs: add missing parameter docs
2011-05-23 23:53:38 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: move the riff structure to std-section
If someone intents to document them and the fields we can move them back.
2011-05-23 23:53:06 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/gstpluginsbaseversion.c:
docs: move pluginbaseversion to separate section as we have section docs
2011-05-23 23:51:15 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspdefs.h:
docs: add minimal docblobs for status code and headers
Use a trick to avoid documenting all 100 enums.
2011-05-23 23:41:56 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
docs: update xoverlay docs for api addition and deprecation
2011-05-23 23:12:50 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
docs: rtp library docs update
2011-05-23 22:58:22 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/navigation.h:
* gst-libs/gst/interfaces/tuner.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
docs: add missing documentation for various pieces
2010-02-19 12:54:18 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: recalibrate clock on setcaps
Because the spec for the ringbuffer can change when changing
the caps, we must recalibrate the clock.
https://bugzilla.gnome.org/show_bug.cgi?id=610443
2011-05-23 16:02:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Try to typefind even if conversion to UTF8 failed
Fixes bug #600043.
2011-05-23 16:02:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Compile the typefind regex with optimization to speed up matching
2011-05-23 15:51:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Interprete typefind strings passed to GRegex as raw bytes instead of valid UTF8
2011-05-23 15:21:59 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/convertframe.c:
convertframe: fix docs
Fixup paramter mismatch between func and prototype. Add missing parameter docs.
2011-05-23 15:08:24 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/multichannel.h:
docs: fixup audio-library docs
2011-05-23 15:02:27 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/app/gstapp.c:
docs: fixup appsrc/sink api docs
2011-05-23 14:53:26 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstaudioiec61937.c:
* gst-libs/gst/audio/gstaudioiec61937.h:
docs: fix docs for new api
Some parameters where wrong, first line missed the ':' and return docs where
broken.
2011-05-23 14:45:23 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: update xmp api docs
Add missing section. Add new section to main-sgml. Add missing function.
2011-05-23 14:07:38 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: remove late check
We deref the pointer two lines before already and besides this internal function
should not be called with this parameter=NULL.
2011-05-23 14:01:29 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: have the default branch as the last one
2011-05-23 14:00:04 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: an uint value can't be <0
2011-05-23 13:53:06 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/video.c:
whitespace: trim trailing whitespace
2011-05-23 13:50:59 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/video.c:
video.c: use a break and a final warning instead of early returns
Use breaks for case branches instead of return 0. We don't expect these to
happen anyway. Thus have a warning before the final return to make it easier to
see when things go out of sync.
2011-05-23 13:49:01 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/video.c:
video.c: use g_assert_not_reached() for logical error here.
This will help to detect them closer to the source if they ever happen.
2011-05-20 10:48:39 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/lang.c:
lang: fix possible array overrun
We where checking for i<G_N_ELEMENTS, but where accessing i+1.
2011-05-19 23:41:08 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: cleanup helper code
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
2011-05-19 23:25:24 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
docs: update plugin introspection data
Now more files are merged and produced in a canonical fashion, which hopefully
creates less or no delta in the future.
2011-05-19 22:56:53 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 9e5bbd5 to 69b981f
2011-05-19 13:40:29 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Add new header file
2011-05-19 08:30:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Autoplug formatters
Autoplug formatters for streams if a formatter with secondary or
higher rank is found. Formatters are autoplugged when there is no
muxer or when the muxer doesn't implement the tagsetter interface.
Currently only the first formatter found is plugged, this might
help in lots of cases, but it doesn't solve the
'lamemp3 ! xingmux ! id3mux'
case.
https://bugzilla.gnome.org/show_bug.cgi?id=649841
2011-05-19 08:27:29 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: fix typos
2011-05-18 22:07:58 +0200 Aleix Conchillo Flaque <aleix@oblong.com>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: Handle headers in caps
2011-05-18 16:09:47 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From fd35073 to 9e5bbd5
2011-05-18 13:18:15 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
* tests/check/libs/video.c:
gstvideo: Add gst_video_get_size_from_caps function
gst_video_get_size_from_caps () allows easy calculation of the raw video
buffer size from some fixed video caps.
API: gst_video_get_size_from_caps()
2011-05-18 12:24:02 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 46dfcea to fd35073
2011-05-18 09:34:52 +0200 Robert Swain <robert.swain@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
alsa: Remove unused but set variable
Unused but set variables cause warnings in GCC 4.6.x and newer.
2011-05-17 10:20:36 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Fix typo which broke the build
2011-05-16 15:35:50 +0200 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: not enter in not controllable state unless it is necessary
When closing rtspsrc the state change blocks until the polling in the
connection timeouts. This is because the second time we loop to read a
full message controllable is set to FALSE in the poll group, even though no
message is half read.
This can be avoided by not setting controllable to FALSE the poll group
unless we had begin to read a message.
Fixes #610916
2010-05-30 13:21:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.h:
cdparanoiasrc: fix build on OSX by #undef-ing VERSION before including system headers
On OSX the cdparanoia headers include IOKit framework headers (in particular
SCSICmds_INQUIRY_Definitions.h) which define a structure that has a member
named VERSION, so we must #undef VERSION before including those for things
to compile on OSX.
Fixes #609918.
2011-05-02 11:43:38 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: optionally ensure maximum average output frame rate
See #628764.
2011-04-29 14:58:02 +0200 Alexey Fisher <bug-track@fisher-privat.net>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: optionally only drop frames to ensure maximum frame rate
This adds option to arrange for maximal allowed variable frame rate.
Fixes #628764.
2011-04-26 13:37:51 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: use bitrate to configure streaming buffer-duration default case
In particular, in audio only cases whose (estimated) metadata provides bitrate
information, the buffer-size based on such bitrate (and buffer-duration)
will be much more reasonable than queue2 default buffer-size.
2011-04-26 11:27:40 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: remove some dead code
... which was dead as pads were never added to the list, and need not be added,
since removing them is handled by a pad callback.
2011-04-29 11:48:02 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/examples/encoding/Makefile.am:
encodebin: examples: Add missing base libs to makefile
2011-04-28 10:58:15 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Check for missing converters
Adds checks for missing video and audio converter elements
2011-04-27 22:05:55 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/xmpwriter.c:
tag: xmpwriter: Rename documentation headers
Fix some wrong documentation headers from the first name
given to this interface.
2011-04-19 08:41:53 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/tag.c:
tests: xmp: New tests for the Iptc4xmpExt tags
2011-04-18 23:28:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Add Iptc4xmpExt schema support
Adds Iptc4xmpExt schema with country, city and sublocation
tags mapped
2011-04-19 11:00:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Add support for reading struct tags
Adds a context variable that controls if the parsing is on
'top level' tags or inside a struct tag.
2011-04-18 16:54:54 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Add struct xmp tag type support
Adds support for writing the xmp struct tag type, it is a compound tag
that has inner tags.
2011-04-18 23:16:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Fixing schema maps
Do not forget to create a new schema for every supported schema
instead of reusing the same object
2011-04-18 10:20:00 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Write the same tag to all schemas
Instead of writing only the xmp tag for the first found entry
that matches the gstreamer tag, look for all mappings to write
the tag to different schemas.
The rationale here is that some reader application might only
be interested on a particular schema tags, so we should try
to write as many tags for all schemas.
2011-05-15 13:39:18 +0200 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstaudio.def:
win32: Update libgstaudio.def for new symbols
2011-05-14 17:27:30 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
baseaudiosink: Use g_str_equal() instead of strncmp()
The strncmp is unnecessary anyway since one of the strings is a const
string.
2011-05-14 16:49:53 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: Fix trivial indentation problems
2011-03-07 20:49:16 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudioiec61937.c:
* gst-libs/gst/audio/gstaudioiec61937.h:
audio: Add an IEC 61937 payloading library
This can be used by sinks to take compressed formats, correctly payload
these in IEC 61937 frames and feed these to sinks that support
passthrough output over IEC 60958 (S/PDIF) or, in the case of MP3, over
Bluetooth.
Initial implementation includes AC3, E-AC3, MPEG-1, MPEG-2 (non-AAC),
and DTS (type-I/II/II) payloading. More formats can be added as needed.
API: gst_audio_iec61937_frame_size()
API: gst_audio_iec61937_payload()
https://bugzilla.gnome.org/show_bug.cgi?id=642730
2011-03-09 11:12:39 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
baseaudiosink: Allow subclasses to provide payloaders
This allows subclasses to provide a "payload" function to prepare
buffers for consumption. The immediate use for this is for sinks that
can handle compressed formats - parsers are directly connected to the
sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading
might be used.
API: GstBaseAudioSinkClass:payload()
https://bugzilla.gnome.org/show_bug.cgi?id=642730
2011-04-09 09:49:10 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: Add support for E-AC3
Adds support for pushing E-AC3 buffers and doing bytes-to-ms conversion
correctly. The assumption (as with other formats) is that something like
IEC 61937 payloading will be used. Correspondingly the ringbuffer spec
is populated so that the data rate is 4x normal AC3.
https://bugzilla.gnome.org/show_bug.cgi?id=642730
2011-03-14 15:51:40 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: Add support for MPEG audio buffers
2011-03-14 15:49:57 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.h:
ringbuffer: Add AAC format types
These are meant to be used for buffers containing AAC data. Nothing uses
this yet, but for now it serves to distinguish from GST_BUFTYPE_MPEG
which represents non-AAC MPEG audio.
API: GST_BUFTYPE_MPEG2_AAC
API: GST_BUFTYPE_MPEG4_AAC
2011-03-09 22:57:00 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: Add support for DTS buffers
2011-05-14 11:42:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Require core 0.10.34.1 for the new ghostpad API
2011-05-09 22:20:23 +0200 Andoni Morales Alastruey <ylatuya@gmail.com>
* gst/playback/gstdecodebin2.c:
decodebin2: fix preroll for streams at low bitrates
For streams at low bitrates we need to set a limit in time because the limit
in bytes might not reached too late, sometimes more than 30 seconds.
This limit can only be set if upstream is seekable (see #584104)
Closes #647769
2011-05-09 13:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Use new ghostpad/proxypad API to get the internal pad
2011-05-09 12:59:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkaudioconvert.h:
* gst/playback/gstplaysinkvideoconvert.c:
* gst/playback/gstplaysinkvideoconvert.h:
playsink: Use new ghostpad/proxypad API
2011-05-09 12:50:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/playbin2-compressed.c:
playbin2: Disable some compressed stream tests that are racy without a stream-activate event
2011-03-29 19:15:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/playbin2-compressed.c:
playbin2: Reset buffer counter in playbin2-compressed tests every time when going to READY
2011-03-25 08:26:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysinkaudioconvert.c:
* gst/playback/gstplaysinkaudioconvert.h:
* gst/playback/gstplaysinkvideoconvert.c:
* gst/playback/gstplaysinkvideoconvert.h:
playsink: Add audio and video converter convenience bins
These reconfigure based on the caps and plugin in converters if
necessary. This also makes switching between compressed and raw
streams work flawlessly without loosing the states of any element
somewhere or having running time problems.
2011-03-15 12:51:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2/playsink: Decide if A/V caps are raw only inside playsink
Before playbin2 would use different selectors for raw audio and
compressed audio (and the same for video) and used different
pads from playsink. This made the involved logic much more
complex and was not implemented completely in playsink, which
made it impossible to support files with a compressed and
uncompressed stream that is support by the sink.
playbin2 handles raw/non-raw streams the same now and the
decision is left to playsink, which now can also handle
caps changes from raw to non-raw and the other way around.
Fixes bug #632788.
2011-03-15 11:41:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/elements/playbin2-compressed.c:
playbin2: Add unit test for compressed stream support in playbin2/playsink
2011-05-09 12:56:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
alsasrc: Fix some compilation errors
2011-05-09 11:50:05 +0200 Pontus Oldberg <pontus.oldberg@invector.se>
* ext/alsa/gstalsasrc.c:
* ext/alsa/gstalsasrc.h:
alsasrc: Improve timestamp accuracy
Fixes bug #635256.
2011-05-06 17:01:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Use new, public ghostpad functions
2011-05-03 11:26:32 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* sys/xvimage/xvimagesink.c:
xvimagesink: Use GST_BOILERPLATE
2011-05-14 09:41:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development
=== release 0.10.34 ===
2011-05-14 01:00:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.34
=== release 0.10.33 ===
2011-05-10 09:32:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.33
Highlights:
- support for 16-bit-per-component video formats
- playbin2 fixes and improvements for custom and non-raw sinks
- oggmux muxes based on running time now
- many other fixes and improvements
2011-04-30 17:35:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* po/da.po:
* po/de.po:
* po/fr.po:
* po/uk.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.32.4 pre-release
2011-04-30 17:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/volume/gstvolumeorc-dist.c:
gst: update orc-generated disted C backup code to orc 0.4.14
2011-04-27 12:09:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* po/bg.po:
* po/nl.po:
* po/pl.po:
* po/ru.po:
* po/sl.po:
* po/tr.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.32.3 pre-release
2011-04-24 18:45:40 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/vs_image.c:
videoscale: Fix off-by-one error in previous commit
Fix for 7c0b702e. It helps to get your j+1's right.
2011-04-24 18:16:20 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/vs_image.c:
videoscale: Fix ARGB bilinear scaling
Fixes #648548. Orc generates bad code for
gst_videoscale_orc_resample_merge_bilinear_u32, so we'll use the
slightly slower two-stage process. I'd fix Orc, but it's hard to
get excited about fixing a feature that I'm planning to deprecate
and replace.
2011-04-23 13:42:23 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/vs_image.c:
videoscale: hack to fix invalid reads in linear
https://bugzilla.gnome.org/show_bug.cgi?id=633837
2011-04-23 12:46:09 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/vs_4tap.c:
videoscale: protect 4tap from out-of-bounds reads
https://bugzilla.gnome.org/show_bug.cgi?id=633837
2011-04-24 14:03:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From c3cafe1 to 46dfcea
2011-04-23 12:44:50 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/gstvideoscale.c:
videoscale: use simpler scaling method for small images
https://bugzilla.gnome.org/show_bug.cgi?id=633837
2011-04-14 09:32:19 +0200 Marc Plano-Lesay <marc.planolesay@gmail.com>
* gst/audioresample/gstaudioresample.c:
audioresample: fix unused-but-set-variable warnings with gcc 4.6
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-22 13:55:20 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
tag: exif: register common tags from tag library
Exif uses tags like image-vertical-ppi or image-horizontal-ppi which are
registered in gst_tag_register_musicbrainz_tags(), but neither GstExifReader
nor GstExifWriter register them.
https://bugzilla.gnome.org/show_bug.cgi?id=648459
2011-04-24 12:16:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
tag: update some FIXMEs for 0.11
2011-04-21 14:11:49 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videoscale.c:
tests: add unit test for basetransform/videoscale negotiation regression
Turn Rene's test pipeline into a unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=648220
2010-11-25 17:01:53 +0100 Håvard Graff <havard.graff@.eu.tandberg.int>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: make sure to not start if the may_start flag is FALSE
Fixes #635784
2011-04-18 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: If NULL caps are passed to buffer_alloc() do fallback allocation
Fixes bug #647857.
2011-04-18 10:19:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/pipelines/oggmux.c:
oggmux: Remove bus GSource to prevent a valgrind warning
2011-04-18 09:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/pipelines/gio.c:
gio: Remove the bus GSource from the main context
Prevents a valgrind warning about possibly leaked memory,
see bug #647763.
2011-04-17 19:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/sdp/Makefile.am:
sdp: remove gst_init() for g-i scanner here again as well to avoid problems with -Wl,--as-needed
2011-04-17 17:59:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/fft/Makefile.am:
fft: remove gst_init() for g-i scanner again
libgstfft doesn't actually use any symbols from libgstreamer, so when
compiling with -Wl,--as-needed it won't even link to it, which can
cause failures with older versions of g-i that ignore the --pkg
arguments.
Should fix PPA build failure on Ubuntu Maverick
2011-04-16 16:31:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Always hold the class-global pango mutex when using pango API
2011-04-16 16:23:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttimeoverlay.c:
{time,clock}overlay: Hold the class-global pango mutex when changing the pango context
2011-04-16 16:21:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttimeoverlay.c:
{clock,time}overlay: Only set the global pango context options once in class_init
Instead of doing it over and over again when instantiating a new instance.
2011-04-16 16:18:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
pango: Create a new pango context for every subclass
timeoverlay/clockoverlay are setting some global options
on the context that shouldn't be used for the generic textoverlay.
2011-04-16 16:03:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/video-enumtypes.c:
0.10.32.2 pre-release
2011-04-16 15:58:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/adder/gstadderorc-dist.c:
* gst/adder/gstadderorc-dist.h:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.h:
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
gst: update disted orc backup code
2011-04-16 15:50:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
docs: update documentation
2011-04-16 15:42:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translations
2011-03-31 17:56:00 +0000 Thibault Saunier <thibault.saunier@collabora.co.uk>
* Android.mk:
* configure.ac:
* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/gstvorbisdeclib.h:
vorbis: add support for using tremolo on android
Tremolo is an ARM-optimised version of xiph's tremor library.
2011-04-15 13:36:39 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggmux: prefer headers from caps to determine stream type
Ogg mandates the first header packet must determine a stream's type.
However, some streams (such as VP8) do not include such a header
when muxed in other containers, and thus do not include this header
as a buffer, but only in caps. We thus use headers from caps when
available to determine a new stream's type.
https://bugzilla.gnome.org/show_bug.cgi?id=647856
2011-04-16 11:00:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
libs: gobject-introspection scanner doesn't need to scan or update plugin info
Make sure the scanner doesn't load or introspect or check any plugins,
(especially not outside the build directory).
2011-04-15 21:09:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
tests: list libs/struct*h files explicitly in Makefile.am
Hopefully makes the gentoo buildbot happy again.
2011-04-15 11:11:52 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: avoid foregoing READY_TO_NULL when appropriate
2011-04-14 22:13:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: ensure proper PAUSED_TO_READY cleanup
... since going async to PAUSED might fail, and never making it to PAUSED
subsequently skips going down to READY.
Fixes #647781.
2011-04-14 12:42:20 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
Revert "video: Remove the extensive checkings from switch"
This reverts commit 500d14c35c656890686574e1c041fb556df17056.
2011-04-14 13:15:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/encodebin.c:
encodebin: Unref encoding profiles after usage in the test
2011-04-14 12:55:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/encodebin.c:
encodebin: Release pads after setting the state to NULL in the unit test
See bug #647756.
2011-04-14 12:23:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Set all elements to NULL and remove them from the bin when removing a source group
2011-04-14 00:26:34 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* gst-libs/gst/video/video.c:
video: Remove the extensive checkings from switch
The default case handles them already
2011-04-13 23:17:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/tag.c:
tests: tag: Fix typo
2011-04-13 23:17:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION for xmp
library.
Includes unit tests.
2011-04-13 23:16:02 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION for exif
library.
Includes unit tests.
2011-04-13 23:13:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
tag: Adds GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
Adds a new tag for indicating the used exposure compensation
level in EV used when capturing an image.
API: GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
2011-04-14 00:24:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/encoding/gstcapslist.c:
* tests/examples/gio/giosrc-mounting.c:
* tests/examples/playrec/playrec.c:
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
tests: fix unused-but-set-variable warnings with gcc 4.6
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-13 23:57:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.c:
ogg: fix unused-but-set-variable warnings with gcc 4.6
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-13 23:19:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: fix unused-but-set-variable warnings with gcc 4.6
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-13 22:59:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: fix unused-but-set-variable warning with gcc 4.6
We don't compare the bitrates of consecutive mp3 frames on purpose
here.
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-13 09:10:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.h:
docs: fix typo in video format docs
2011-04-12 12:41:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: fix uninitialised variable usage and element leak
gcc on OSX complains about ret being used uninitialized in
this function, and it is right. Don't leak element ref
when returning early because newsegment event is not in
TIME format.
2011-04-12 12:20:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/tcp/gstmultifdsink.c:
multifdsink: do check return values of fcntl() and fstat()
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-09 19:15:23 +0200 Marc Plano-Lesay <marc.planolesay@gmail.com>
* gst/playback/gstplaybasebin.c:
* gst/subparse/tmplayerparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/videoscale/vs_image.c:
fix unused-but-set-variable warnings with gcc 4.6
https://bugzilla.gnome.org/show_bug.cgi?id=647294
2011-04-06 22:57:41 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* gst-libs/gst/rtsp/gstrtsptransport.c:
rtsptranport: ensure valid int result when parsing ranges
Specifically, make sure that the return value of strtol is falling in
between the range of G_MININT and G_MAXINT.
Fixes #646952.
2011-04-06 16:27:54 +0100 Bastien Nocera <hadess@hadess.net>
* gst-libs/gst/pbutils/encoding-target.c:
encoding-profile: fix unused-but-set-variable warnings with gcc 4.6
Top-level profiles don't have restrictions, only stream profiles,
so no need to serialise that here.
https://bugzilla.gnome.org/show_bug.cgi?id=646925
2011-04-11 14:29:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
tests: dist all struct_*.h files for libs ABI test
Should fix distcheck on x86_64.
2011-04-11 15:02:38 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: empty caps have no structure to pick
2011-04-11 10:06:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstid3tag.c:
tag: fix typo in ID3 genres
psychadelic -> psychedelic. Spotted by Sébastien Wilmet.
https://bugzilla.gnome.org/show_bug.cgi?id=647399
2011-01-27 17:28:51 +0100 Alessandro Decina <alessandro.d@gmail.com>
* Android.mk:
* android/alsa.mk:
* android/app.mk:
* android/app_plugin.mk:
* android/audio.mk:
* android/audioconvert.mk:
* android/audioresample.mk:
* android/audiotestsrc.mk:
* android/decodebin.mk:
* android/decodebin2.mk:
* android/ffmpegcolorspace.mk:
* android/gdp.mk:
* android/gst-libs/gst/app/gstapp-marshal.c:
* android/gst-libs/gst/app/gstapp-marshal.h:
* android/gst-libs/gst/audio/audio-enumtypes.c:
* android/gst-libs/gst/audio/audio-enumtypes.h:
* android/gst-libs/gst/interfaces/interfaces-enumtypes.c:
* android/gst-libs/gst/interfaces/interfaces-enumtypes.h:
* android/gst-libs/gst/interfaces/interfaces-marshal.c:
* android/gst-libs/gst/interfaces/interfaces-marshal.h:
* android/gst-libs/gst/pbutils/pbutils-enumtypes.c:
* android/gst-libs/gst/pbutils/pbutils-enumtypes.h:
* android/gst-libs/gst/rtsp/gstrtsp-enumtypes.c:
* android/gst-libs/gst/rtsp/gstrtsp-enumtypes.h:
* android/gst-libs/gst/rtsp/gstrtsp-marshal.c:
* android/gst-libs/gst/rtsp/gstrtsp-marshal.h:
* android/gst-libs/gst/video/video-enumtypes.c:
* android/gst-libs/gst/video/video-enumtypes.h:
* android/gst/playback/gstplay-marshal.c:
* android/gst/playback/gstplay-marshal.h:
* android/gst/tcp/gsttcp-enumtypes.c:
* android/gst/tcp/gsttcp-enumtypes.h:
* android/gst/tcp/gsttcp-marshal.c:
* android/gst/tcp/gsttcp-marshal.h:
* android/interfaces.mk:
* android/netbuffer.mk:
* android/pbutils.mk:
* android/playbin.mk:
* android/queue2.mk:
* android/riff.mk:
* android/rtp.mk:
* android/rtsp.mk:
* android/sdp.mk:
* android/tag.mk:
* android/tcp.mk:
* android/typefindfunctions.mk:
* android/video.mk:
* android/videoscale.mk:
* android/videotestsrc.mk:
* ext/ogg/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst/adder/Makefile.am:
* gst/app/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/encoding/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
* gst/gdp/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* tools/Makefile.am:
android: make it ready for androgenizer
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-09 02:01:08 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Add new header file to spec file
2011-04-08 15:10:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
rtp: Unref events if the parent element disappeared or has no event handler implemented
2011-01-06 18:20:58 +0100 Ole André Vadla Ravnås <oravnas@cisco.com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
rtp: fix pad callbacks so they handle when parent goes away
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
2011-03-20 08:59:33 +0100 Havard Graff <havard.graff@tandberg.com>
* gst/audioresample/gstaudioresample.c:
audioresample: Make src query MT-safe
It is possible that the element might be going down while the event arrives
2011-04-08 15:00:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: Unref events if the parent element disappeared
2011-03-21 16:03:16 +0100 Havard Graff <havard.graff@tandberg.com>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: make upstream queries and events MT-safe
2011-04-06 16:25:37 +0100 Bastien Nocera <hadess@hadess.net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtp: Remove unused variables
https://bugzilla.gnome.org/show_bug.cgi?id=646924
2011-04-07 10:06:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Fix creation of grayscale caps
The endianness was not set correctly before.
Fixes bug #646923.
2011-04-06 16:11:02 +0200 Robert Swain <robert.swain@collabora.co.uk>
* docs/design/part-interlaced-video.txt:
docs: Update interlaced video design document
The RFF flag is to be reused for buffers in the telecine state to
indicate that the buffer contains only unneeded repeated fields that are
present in other buffers and as such this buffer can be dropped.
2011-03-25 16:59:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: refactor multipass file writing
2011-02-08 14:02:20 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
audioresample: minor simplification
... which avoids crashing in the off-chance that structure == NULL.
2011-04-05 18:14:49 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/check/Makefile.am:
* tests/check/libs/.gitignore:
* tests/check/libs/discoverer.c:
tests: add basic unit tests for discoverer
2010-08-24 13:14:33 +0200 Pascal Buhler <pascal.buhler@tandberg.com>
* gst-libs/gst/rtp/gstrtcpbuffer.c:
rtcpbuffer: Round to next 32bit word, not current 32bit word at end of SDES chunk
2011-04-05 11:32:52 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: don't paint the window black when going to NULL
Leave dealing with the appearance of the window when we are not playing to the
applications. We anyway want to go to NULL as quickly as possible.
Fixes #635800
2011-04-04 16:00:30 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* tests/check/libs/video.c:
video: Fix YUV9 and YVU9 again
2011-04-04 23:41:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
tag: fix compiler warning on OSX
gstvorbistag.c: In function 'gst_tag_list_from_vorbiscomment_buffer':
gstvorbistag.c:371: warning: 'data' may be used uninitialized in this function
2011-04-04 23:23:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/.gitignore:
tests: ignore xmpwriter unit test binary
2011-04-04 17:21:45 +0200 Haakon Sporsheim <haakon.sporsheim@gmail.com>
* gst-libs/gst/tag/gstexiftag.c:
tag: use gst/math-compat.h header.
https://bugzilla.gnome.org/show_bug.cgi?id=646744
2011-04-04 17:23:53 +0200 Haakon Sporsheim <haakon.sporsheim@gmail.com>
* gst-libs/gst/tag/xmpwriter.c:
tag: Remove constness to silence MS compiler.
https://bugzilla.gnome.org/show_bug.cgi?id=646744
2011-04-04 17:23:13 +0200 Haakon Sporsheim <haakon.sporsheim@gmail.com>
* gst-libs/gst/tag/gstxmptag.c:
tag: Explicit cast to GThreadFunc to silence MS compiler.
https://bugzilla.gnome.org/show_bug.cgi?id=646744
2011-04-04 15:56:50 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 1ccbe09 to c3cafe1
2011-03-11 10:41:11 +0100 Trond Andersen <trondand@cisco.com>
* gst-libs/gst/rtp/gstrtcpbuffer.c:
rtcpbuffer: fix invalid read in validation of padding in rtcp packet
2011-02-23 10:55:12 +0100 Stian Johansen <stian.johansen@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: Add src object lock around call to ringbuffer parse caps.
A race was observed between query() and setcaps() where the latter would
change the ringbuffer spec while the former was performing operations
based this data.
2011-01-22 23:09:32 +0100 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: protect against ringbuffer disappearing while in a query
Observed a case where the src went to null-state during the query,
hence the spec pointer was no longer valid, and
gst_util_unit64_scale_int crashed (assertion `denom > 0´failed)
Add locking to make sure the ringbuffer can't disappear.
2011-02-08 18:27:43 +0100 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: don't allow aligning behind the read-segment
Given a large enough drift-tolerance, one could end up in a situation
where one would keep aligning the written buffers behind the current
read-segment position. The result for the reader would be complete
silence, possible preceded by very choppy audio.
By checking the available headroom, one can determine if there is
room to do alignment, or if one should resort to a resync instead to get
the pointers back on track.
Also refactor the alignment-logic out of the render function for cleaner
code.
2011-04-01 13:55:26 -0700 David Schleef <ds@schleef.org>
* gst/encoding/Makefile.am:
* gst/playback/Makefile.am:
Remove setting of plugindir from Makefiles
2011-03-23 23:10:51 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* tests/check/libs/video.c:
video: Fix height calculation for YUV9/YVU9
2011-04-01 15:34:30 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/ogg/gstoggmux.c:
oggmux: fix warning building in mac os x
2011-04-01 15:33:42 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/pango/gsttextoverlay.c:
textoverlay: fix comparison is always false due to limited range of data type
Perform calculation in a temp var with enough room as there's guarantee that
ret will be able to hold the result for example in _blit_AYUV.
2011-04-01 12:52:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: Write GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE as METADATA_BLOCK_PICTURE
This is the official, standardized way of embedding images into
vorbiscomments now.
2011-04-01 12:28:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: Add support for METADATA_BLOCK_PICTURE tags
This is the official, standardized way of embedding pictures
inside vorbiscomments now. Parsing code taken from flacparse
and slightly changed.
Fixes bug #635669.
2011-04-01 12:09:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: Use g_base64_decode_inplace()
Instead of using the GLib base64 decoding functions manually to
do inplace base64 decoding. This makes the code easier to understand.
2011-04-01 11:00:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: Store the segment directly inside the pad
Also initialize it always in TIME format. We require TIME segments
in oggmux anyway and drop newsegment events in other formats and
assume an open-ended segment starting at 0.
2011-04-01 10:57:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Reset the segment on flush-stop events and when going back to READY
2011-03-03 08:45:15 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Use running time instead of timestamps
Theora and vorbis use running time (which is correct) for calculating
the granulepos for their ogg packets. Oggmux, however, used
timestamps to order the received buffers.
This patch makes it use the running time to compare buffer times
and also to timestamp pushed buffers.
Some bits of the code still use timestamps, but they are only
used to calculate durations, so it should be fine.
https://bugzilla.gnome.org/show_bug.cgi?id=643775
2011-02-16 16:07:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: Keep track of pad's segments
https://bugzilla.gnome.org/show_bug.cgi?id=643775
2011-04-01 10:39:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Add support for xBGR and RGBx
Now all RGB variants are supported.
2011-01-17 21:12:18 -0700 Lane Brooks <dirjud@gmail.com>
* ext/pango/gsttextoverlay.c:
textoverlay: Added support for ARGB and other RGB alpha variants
2011-01-11 10:34:33 -0700 Lane Brooks <dirjud@gmail.com>
* ext/pango/gsttextoverlay.c:
textoverlay: converted AYUV to use 'A OVER B' alpha compositing
'A OVER B' compositing is explained at
http://en.wikipedia.org/wiki/Alpha_compositing.
Previously, overlaying text on a transparent background image left the
text overlay also transparent. This pipeline shows such an example:
gst-launch videotestsrc pattern=white ! video/x-raw-yuv,format=\(fourcc\)AYUV ! alpha alpha=0.0 ! textoverlay text=Testing auto-resize=False font-desc=60px ! videomixer ! ffmpegcolorspace ! autovideosink
With this patch, text is composited "OVER" the background image and
thus is visible regardless of the alpha of the background image. The
overlay in the above pipeline works after applying this patch.
2011-03-28 22:00:25 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
baseaudiosink: arrange for running clock when rendering eos
Commit ba2e500bd992d8ad7db0da923801964964835967 ensured to provide
a running clock when EOS had finished rendering. However,
other measures are needed (and were in place before) to ensure a
running clock when EOS still needs rendering (i.e. waiting).
So, specifically, re-introduce eos_rendering removed in aforementioned commit,
this time as a public variable so subclasses can be aware of the situation.
Fixes (part of) #645961.
API: GstBaseAudioSink:eos_rendering
2011-03-31 12:37:32 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* tests/check/libs/libsabi.c:
* tests/check/libs/struct_i386_osx.h:
tests: Fixes libsabi for MacOSX/32bit.
GStaticRecMutex is 60bytes on macosx/32bit (As opposed to 40).
Fixes #644996
2011-03-31 10:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/libsabi.c:
* tests/check/libs/struct_x86_64.h:
libsabi: Add structure sizes for x86-64
2011-03-09 11:51:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
libs: make sure gobject-introspection scanner calls gst_init()
Cherry-picked from 0.11, since it's the right thing to do (we
now silently rely on various _get_type() working without
gst_init() having been called).
2011-03-30 20:57:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
libs: replace 0.10 with @GST_MAJORMINOR@ in Makefile.am
For easier cherry-picking/merging later.
2011-03-30 15:47:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/gst/typefindfunctions.c:
* tests/files/Makefile.am:
* tests/files/hls.m3u8:
tests: add typefind test for application/x-hls
To make sure we don't break detection when we add typefinding
for normal m3u8 playlists.
2011-03-30 15:44:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: rename type playlist/m3u8 to application/x-hls
We should keep playlist/m3u8 available for normal m3u8 playlists,
which we we'll likely support some day. Also, we probably don't
want this handled like other playlists, so application/* seems
more appropriate in this case, even if it's really just a playlist.
2011-03-30 09:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: Fix comment typo and add a link the the HTTP live streaming spec
2011-03-30 09:12:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: Use the DataScanCtx for the m3u8 typefinder
2011-02-14 19:05:09 +0100 Andoni Morales Alastruey <amorales@flumotion.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: add m3u8 playlists
2011-03-21 15:34:09 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/libs/xmpwriter.c:
tagxmpwriter: Add check tests
https://bugzilla.gnome.org/show_bug.cgi?id=645167
2011-03-17 15:42:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstxmptag.c:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/xmpwriter.c:
* gst-libs/gst/tag/xmpwriter.h:
* win32/common/libgsttag.def:
tagxmpwriter: Adds a new GstTagXmpWriter interface
The GstTagXmpWriter interface is to be implemented on elements that
provide xmp serialization. It allows users to select which
xmp schemas should be used on serialization.
API: GstTagXmpWriter
https://bugzilla.gnome.org/show_bug.cgi?id=645167
2011-03-18 09:28:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* gst-libs/gst/tag/tag.h:
* win32/common/libgsttag.def:
tag: xmp: Add function to list the available schemas
Adds a function to list the available schemas in our xmp lib
https://bugzilla.gnome.org/show_bug.cgi?id=645167
2011-03-29 15:41:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/encodebin.c:
encodebin: Requesting a pad again now gives a g_return_val_if_fail()
Before the behaviour was undefined and implemented differently by elements,
now core checks for this (and other problems) and returns NULL and an assertion.
2011-03-26 19:36:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggparse.c:
oggparse: fix list iteration code
Not that it really matters, but let's fix it before someone
notices and makes fun of us.
2011-03-26 12:01:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/.gitignore:
tests: ignore new libsabi test binary
2011-03-26 11:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggparse.c:
oggparse: make sure buffer metadata is writable before setting caps on buffers
2011-03-25 22:14:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 193b717 to 1ccbe09
2011-03-25 14:55:52 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From b77e2bf to 193b717
2011-03-25 11:06:35 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/Makefile.am:
docs: do xrefs for non installed books too
Get the xrefs from the builddir for the books in the same package. This fixes
the cross references if one does not have the docs already installed.
2011-02-25 16:46:29 +0100 Robert Swain <robert.swain@collabora.co.uk>
* docs/design/part-interlaced-video.txt:
docs: Add an interlaced video design document
2011-03-25 09:29:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From d8814b6 to b77e2bf
2011-03-25 09:03:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 6aaa286 to d8814b6
2011-03-24 18:48:59 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 6aec6b9 to 6aaa286
2011-03-24 14:22:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Update comment about why an audio queue is needed
2011-03-24 14:21:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
Revert "playsink: Only add a queue before the audio sink if visualizations are enabled"
This reverts commit df886c0622257bb8635e5bd0fc7fc3da20bfc3be.
2011-03-24 14:03:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Only add a queue before the audio sink if visualizations are enabled
The queue is not needed otherwise and will add some delay to track
switches.
2011-03-23 12:42:04 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/video.c:
tests: video: Uncommenting test
Pushed a commented test by accident, uncommenting it.
2011-03-23 12:02:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* win32/common/libgstvideo.def:
video: adds missing function to win32 def
2011-03-23 12:02:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Getting component offsets without dimensions is fine if it is not YUV
This fixes a regression that an assertion would happen if
gst_video_get_component_offset would be called with width or
height as 0.
Calling it with 0 is fine if the format isn't yuv and this
was already being used in some other places of video.c
2011-03-23 11:13:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/video.c:
tests: video: Add a test for checking rgb caps creation
This new test for checking rgb caps creation exposes a regression
2011-03-15 14:45:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Remember automatically created sinks for future reconfigures
Also allow reuse of sink elements in error cases.
2011-03-16 15:27:51 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Check if an already existing sink supports the non-raw format too
Before we were assuming that a sink will always support all non-raw formats
in a single stream.
2011-03-10 19:04:51 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Check if an element accepts requisite caps before selecting
In addition to ensuring that an element we want to select in
autoplug-select can enter the READY state, we also now check if it can
accept the caps we wish to plug it for. This is handy for sinks that
need to perform a probe to figure out whether they can actually handle a
given format.
2011-03-16 15:56:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Set sinks to READY before checking if it accept caps
Fixes bug #642732.
2011-03-16 15:56:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Always prefer the custom set sink and also set it back to NULL in all cases.
2011-03-17 13:47:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Only consider the audio/video sinks in autoplug_continue for the normal uridecodebin
Considering them for the subtitle uridecodebin will add audio/video
streams that might be in a file used as subtitle file.
2011-03-22 11:59:40 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add gst_video_format_new_template_caps()
2011-02-24 08:42:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Fix assertion on caps fixation
When fixating caps, from_par should always be initialized
with a fixed value.
In case the fixation is from src to sink pad it was setting
the from par (srcpad par) to a fraction range, this patch initializes
it to 1/1, based on the assumption that missing PAR is 1/1.
https://bugzilla.gnome.org/show_bug.cgi?id=641952
2011-03-22 12:44:49 +0100 Luis de Bethencourt <luis@debethencourt.com>
* configure.ac:
configure.ac: redundant use of AC_MSG_RESULT()
cleaned the redundant use of AC_MSG_RESULT() in configure.ac
2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
* autogen.sh:
autogen: wingo signed comment
2011-03-21 19:22:30 +0100 Fraxinas <andreas.frisch@multimedia-labs.de>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Fix syntax in Example: Creating a profile
https://bugzilla.gnome.org/show_bug.cgi?id=645437
2011-03-21 18:33:03 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Add missing schema creation
tiff schema entries were being added to the previous
schema (xap) because a new one wasn't being created
for it.
2011-03-17 21:50:15 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtpbuffer: Off-by-one error when creating RTP header extensions with a two-byte header
2011-03-16 15:38:31 +0200 Mart Raudsepp <mart.raudsepp@collabora.co.uk>
* ext/pango/gsttextoverlay.h:
textoverlay: Clean up alignment docs a bit and remove horiz top alignment enum
2011-02-07 09:13:39 +0200 Mart Raudsepp <leio@gentoo.org>
* tests/check/Makefile.am:
check: Really fix the linking order of libs/tag
Follow-up to commit 5f5c52c, which only fixed the CFLAGS order.
Fix the linker order as well.
2011-03-16 10:19:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: post proper error message if decodebin2/typefind elements are missing
Post better error messages in case typefind/decodebin2 are missing or
could not be loaded for some reason (e.g. because they inadvertently
got blacklisted).
https://bugzilla.gnome.org/show_bug.cgi?id=644892
2011-03-15 19:47:11 +0100 Blaise Gassend <blaise@suitabletech.com>
* ext/alsa/gstalsamixer.c:
alsamixer: Store return values of poll functions in a signed integer
Negative return values are used for errors and storing
them in an unsigned integer will make it impossible to
detect the errors.
Fixes bug #644845.
2011-03-14 19:42:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Increase the seen header packets count when seeing a header packet
This fixes muxing of Speex content and possibly other formats where the
header detection works by counting the packets.
Fixes bug #644745.
2011-03-14 18:35:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: add depth and endianness to DTS caps
https://bugzilla.gnome.org/show_bug.cgi?id=644208
2011-03-14 11:14:04 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: use a class wide mutex to work around pango reentrance issues
Pango is not reentrant. Use a class wide mutex to protect pange use in
gst_text_overlay_render_pangocairo(). This works reliable in contrast to the
hack in my previous commit.
Fixes Bug #412678
2011-03-14 11:12:53 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/pango/gsttextoverlay.c:
Revert "textoverlay: add a hack to init the pango engine"
This reverts commit fee3266056b522cdd34e606b5682553d35eec5a1.
2011-03-14 10:09:35 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin2.c:
plaback: trim trailing whitespace
2011-03-14 10:05:34 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin2.c:
decodebin2: reflow configuring new multiqueue instance
Use a single g_object_set to configure the new multiqueue instance. Also don't
needlessly set "use-buffering" if it is the default.
2011-03-04 14:52:01 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/pango/gsttextoverlay.c:
textoverlay: drop trailing whitespaces
2011-03-04 14:52:28 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/pango/gsttextoverlay.c:
textoverlay: add a hack to init the pango engine
Layout a single char to pre-create all resources.
2011-03-12 17:51:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* tests/check/Makefile.am:
* tests/check/libs/.gitignore:
* tests/check/libs/gstlibscpp.cc:
tests: add libscpp unit test to make sure g++ likes our library headers
2011-03-10 14:22:38 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/elements/encodebin.c:
tests: encodebin: Add reuse test case
Adds a test case to check if encodebin can be reused
https://bugzilla.gnome.org/show_bug.cgi?id=644416
2011-03-10 14:38:47 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Tear down old profiles when setting new ones
In NULL/READY, we should be able to switch profiles on encodebin,
this patch makes it tear down old profiles when new ones are set
if in NULL/READY states
https://bugzilla.gnome.org/show_bug.cgi?id=644416
2010-10-22 14:01:26 +0200 Andoni Morales Alastruey <amorales@flumotion.com>
* gst/tcp/gstmultifdsink.c:
multifdsink: disconnect inactive clients in the select loop too
Clients are usually disconnected in the streaming thread if their inactivity
is bigger than the timeout. If no new buffers are to be rendered in the sink,
these clients will never be disconnected and for that reason it should be
handled in the select() loop too.
2010-11-03 14:37:07 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Don't wait for subtitle streams to preroll
Subtitle streams being parse can cause the pipeline to wait indefinitely
to PREROLL. This makes subtitle streams got to PAUSED even if no data is
available. This should not be a cause for concern as we don't expect to
get much data for subtitle streams other than language tags from the
container.
https://bugzilla.gnome.org/show_bug.cgi?id=632291
2011-03-03 19:14:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: start ringbuffer upon going to PLAYING and already EOS
... otherwise we may end up without running clock in PLAYING.
Fixes #636886.
2011-03-04 14:39:45 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaybin2.c:
playbin2: set several properties in one go
g_object_set is a varargs function. Save 7 g_obvject_calls (and the overhead of
them) by using it accordingly.
2011-03-02 15:38:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: fix compiler warning on 32-bit systems
Mark 64-bit interger constant as such to avoid warnings such as:
gsttypefindfunctions.c:2152: error: integer constant is too large for ‘long’ type
2011-02-28 18:52:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* configure.ac:
configure.ac: export plugin description more platform independent
Fixes #642504.
2011-02-28 18:32:33 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* common:
Automatic update of common submodule
From 1de7f6a to 6aec6b9
2011-02-28 10:10:22 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/check/Makefile.am:
* tests/check/libs/libsabi.c:
* tests/check/libs/struct_i386.h:
tests: add ABI test suite for libs
2011-02-27 09:32:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Only prevent to autoplug the same parser multiple times for the same chain
Parsers are the only element class that are not changing the data and
could lead to an infinite loop. Other element classes like demuxers,
e.g. id3demux, can be used multiple times in a row and sometimes are.
2011-02-26 23:43:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Break the double-factory checking loop immediately if the factory was used already
2011-02-26 23:39:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't use the same element multiple times in the same chain
This is going to lead to an infinite loop of this element and can easily
happen with parsers that accept their own src caps on the sinkpad.
2011-02-26 23:20:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Improve detection of raw caps in expose-all-streams=false mode
Previously we only checked against the raw caps but we should also
check against the return value of autoplug-continue. Additionally fix
a thread-safety issue with accessing the raw caps.
2011-02-25 19:37:07 -0800 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add support for r210
2011-01-03 11:41:56 +0100 Robert Swain <robert.swain@collabora.co.uk>
* gst-libs/gst/video/video.h:
gstvideo: Add GST_VIDEO_BUFFER_PROGRESSIVE flag
Maps to GST_BUFFER_FLAG_MEDIA4. The purpose is to explicitly indicate
whether a telecined buffer is progressive or not without having to make
assumptions based on previous buffers.
2011-02-24 20:59:48 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/encodebin.c:
encodebin: Fix double unref in unit test
2011-02-22 14:54:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/playbin2.c:
checks: add a simple unit test for the source-setup signal
2011-02-22 12:56:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playbin2, uridecodebin: add "source-setup" signal
Add "source-setup" signal for convenience and discoverability. No need
to figure out "notify::source", look up the notify callback signature,
then do an g_object_get() to get the source element..
https://bugzilla.gnome.org/show_bug.cgi?id=626152
2011-02-24 16:22:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Don't handle GstCollectData as GstObject, use the pad instead
2011-02-24 16:02:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/encodebin.c:
encodebin: Fix memory leaks related to request pads
Request pads have to be released by the caller and must be
unreffed after releasing them.
2011-02-24 15:55:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Return a new reference of the pad for the "request-pad" signal
The GObject signal code assumes that the signal handlers return a
new reference or copy. Fixes bug #641927.
2011-02-21 20:34:41 -0800 Leo Singer <leo.singer@ligo.org>
* gst/adder/gstadder.c:
adder: Fill in offset_end field of outgoing buffers
... rather than leave it as GST_BUFFER_OFFSET_NONE
Fix bug #642942.
2011-02-23 14:31:13 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: release all chains when going to NULL
Also fixes #642466.
2011-02-23 14:29:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: undo state change side effect on error way out
... to avoid subsequent cleanup disposing an element not in NULL state.
2011-02-23 10:32:08 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: avoid crashing on the way out when needed chain missing
2011-02-22 15:26:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstvideo.def:
win32: update .def file for new libgstvideo API
2011-02-22 16:41:54 +0200 Stefan Kost <ensonic@users.sf.net>
* tools/gst-discoverer.c:
discoverer: handle desc==NULL
It would otherwise be printed as (null) and mess up indentation (no \n).
2011-02-08 12:42:32 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Chain dispose() up to parent class
2011-02-07 13:04:55 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Keep a ref for the async timeout callback
This makes sure we maintain a ref on the discoverer object while the
async timeout callback is alive to prevent a potential crash if the
object is freed while the callback is pending.
https://bugzilla.gnome.org/show_bug.cgi?id=641706
2011-02-07 13:57:39 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Use g_signal_connect_object instead of g_signal_connect
We want to make sure the discoverer object passed to the various
callbacks doesn't become invalid if a callback is pending and the object
is free'd in the mean time.
https://bugzilla.gnome.org/show_bug.cgi?id=641706
2011-02-10 03:22:42 +1100 Parthasarathi Susarla <partha.susarla@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: detect raw h.263
https://bugzilla.gnome.org/show_bug.cgi?id=623846
2011-02-21 15:58:16 +0200 Teemu Katajisto <teemu.katajisto@digia.com>
* gst-libs/gst/pbutils/encoding-target.c:
pbutils: encoding-target: fix error checking in target file loading
https://bugzilla.gnome.org/show_bug.cgi?id=642949
2011-02-21 17:55:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videoscale.c:
tests: fix videoscale test by ignoring newly-added 64-bit formats
They probably fail because ffmpegcolorspace can't handle those formats.
2011-02-21 18:01:04 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/sdp/Makefile.am:
sdp: Fix copy/paste error in inrospection part of Makefile
2011-02-21 18:00:36 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/tag/Makefile.am:
tag: Fix copy/paste error in inrospection part of Makefile
2011-02-21 18:00:02 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/rtsp/Makefile.am:
rtsp: Fix copy/paste error in inrospection part of Makefile
2011-02-21 12:40:36 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
* gst/audiorate/gstaudiorate.h:
audiorate: add skip-to-first property
API: GstAudioRate::skip-to-first
2011-02-21 12:27:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: fix skip-to-first ts setup
... such as avoiding arithmetic mixing counts and ts, although latter
would typically be 0 so far.
2011-02-21 12:04:09 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* gst/adder/gstadder.c:
Revert "oggmux,adder: Check if collectpads has been freed"
This reverts commit 6d150873e8b4c23d694b0351570de323b1576d76.
Depends on a core commit that was reverted.
2011-02-20 23:49:54 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggmux.c:
* gst/adder/gstadder.c:
oggmux,adder: Check if collectpads has been freed
Core now calls release_pad in finalize, which is usually after
the collectpads has been unreffed.
2011-02-19 18:50:37 -0800 David Schleef <ds@schleef.org>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* gst/videoscale/vs_fill_borders.c:
* gst/videoscale/vs_fill_borders.h:
* gst/videoscale/vs_image.c:
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.c:
* gst/videoscale/vs_scanline.h:
videoscale: Add 16-bit-channel support
2011-02-19 16:41:43 -0800 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add 16-bit-per-channel formats
2011-02-19 12:03:17 -0800 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add ARGB64 and AYUV64
16-bit per channel formats.
2011-02-18 16:26:59 -0800 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add gst_video_format_get_component_depth()
2011-02-18 13:27:23 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
audiotestsrc: each element gets its own instance of GRand, if needed
As a result, pipelines that contain multiple instances of audiotestsrc
with the 'wave' property set to 'white-noise', 'pink-noise', or
'gaussian-noise' will run much faster, since they won't be competing
for access to the global, lock-protected instance of GRand.
Fixes bug #642720.
2011-02-18 17:26:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: If a sink claims to support ANY caps assume that it only supports the usual raw formats
This should be changed again in 0.11, if a sink really claims to support ANY
caps it should support everything or provide correct caps.
2011-02-17 18:11:10 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Add a audioconverter after the audio resampler.
This allows handling non-native-endianness conversion properly.
2011-02-18 14:04:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Use gst_pad_accept_caps() instead of intersecting with the getcaps caps
This might be faster and more accurate in some cases to detect if a
sink supports a format and autoplugging can be stopped.
2011-02-18 12:06:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Add default handler for autoplug-select
uridecodebin proxies this signal and only the first signal handler
will ever be called from decodebin2, which is uridecodebin's proxy
signal handler.
2011-02-18 12:02:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Return NULL from the default autoplug-sort handler
...instead of copying the array. Returning NULL will result
in the original factories array to be used and prevents a useless
array copy in most use cases.
2011-02-18 12:01:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Return NULL from the default autoplug-sort handler
...instead of copying the array. Returning NULL will result
in the original factories array to be used and prevents a useless
array copy in most use cases.
2011-02-18 12:00:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Update autoplug-* signal docs from decodebin2
uridecodebin proxies these signals.
2011-02-18 11:58:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Update documentation of the autoplug-* signals
Add notes about the behaviour if multiple signal handlers are connected.
For most autoplug-* signals only the first signal handler will ever
be invoked.
Also add to the autoplug-sort docs that the signal handler can return NULL
to specify that the order should change and other handlers get the chance
to sort the array.
2011-02-18 11:57:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Keep the original factory list if the sort signal handlers returned NULL
2011-02-16 20:14:25 +0900 tskd2@yahoo.co.jp <tskd2@yahoo.co.jp>
* gst/playback/gsturidecodebin.c:
uridecodebin: expose "autoplug-sort" signal
It is a proxy of the decodebin2's one, and was missing
in the previous code.
See bug #642433.
2011-02-18 10:57:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Use a recursive mutex for the playbin lock
This lock is taken when activating a group, which could result in
calling the autoplug-continue callback, which also needs this lock
to access the sinks.
See bug #642174.
2011-02-18 09:36:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Disconnect signal handlers when removing a failed element
This prevents crashes later if one of the signals is emitted after the
element was removed from decodebin2 already, which can happen in discoverer.
2011-02-15 19:23:48 -0800 David Schleef <ds@schleef.org>
* gst/typefind/gsttypefindfunctions.c:
typefind: Fix mpeg TS detection
2011-02-04 17:36:40 -0800 David Schleef <ds@schleef.org>
* ext/theora/gsttheoraenc.c:
theoraenc: move debug category init earlier
2011-02-03 22:41:23 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.h:
oggparse: better detection of delta unit flag
2011-01-15 18:21:28 -0800 David Schleef <ds@schleef.org>
* ext/theora/gsttheoraenc.c:
theoraenc: Set speed level while running
2011-01-13 15:12:53 -0800 Ralph Giles <giles@thaumus.net>
* ext/theora/gsttheoraenc.c:
Set the theoraenc speed-level property from libtheora's defaults.
The speed-level property, which allows callers to trade of encoding
quality for speed in the libtheora api, has a version-dependent
maximum and default values. Instead of hardcoding the acceptable
range for the theoraenc element's presentation of this setting,
we query the library directly at class initialization time and
set the maximum and default values from that. If the query fails,
we fall back to the previous default setting.
To keep the values reported by gst-inspect (which I'm told use
the spec values from the class) with those available on an\
instantiated element, we remove to setting of enc->speed_level
from the initializer and instead pass G_PARAM_CONSTRUCT to
the property spec flags, asking g_object to set this property
when theoraenc objects are constructed.
NB in theory the maximum speed-level could depend on the actual
video caps. If later versions of libtheoraenc do this, a second
call will need to be made from theora_enc_reset to update the
property, since this function is mostly useful for realtime
adjustment of performance while the pipeline is running.
2011-02-16 11:57:31 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: don't leak parent tags
2011-02-16 11:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: improve logging (and reindent)
Add more logging for the tag merging and use the _OBJECT flavour more.
2011-02-15 17:46:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Optimize autoplug-continue handler a bit
Don't build merge the caps of all sinks but check them one-by-one
until one supports the caps. Also get reffed caps from the sinkpads
instead of a writable copy and add debug output if a sink claims to
support ANY caps.
2011-02-15 17:24:28 +0100 Akihiro Tsukada <tskd2@yahoo.co.jp>
* gst/playback/gstplaybin2.c:
playbin2: Fix handling of non-raw custom sinks
When autoplugging elements in decodebin2, check if
the caps are supported by one of the sink before
continuing autoplugging.
Fixes bug #642174.
2011-02-15 17:01:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't leak elements that fail to go to PAUSED after being autoplugged
Fixes bug #642381.
2011-02-13 14:42:14 +0000 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: Don't reset the video quality setting the bitrate
libtheora has two encoding modes, CBR, where it tries to hit a target
bitrate and VBR where it tries to achieve a target quality.
Internally if the target bitrate is set to anything other then 0 the
encoding-mode is CBR.
This means that the gstreamer element can leave the video_quality
setting alone as long as the user is tweaking the bitrate. Which has the
nice side-effect that if the user explicitely sets the bitrate to 0
(which is actually the default), the quality value doesn't get reset and
one ends up encoding VBR at quality-level 0...
2011-02-09 12:45:23 +0100 Andoni Morales Alastruey <amorales@flumotion.com>
* gst/gdp/gstgdppay.c:
gdppay: ensure buffer's metadata is writable before setting caps
2011-02-14 12:52:59 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f94d739 to 1de7f6a
2011-02-10 23:44:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-plugins-base.doap:
doap: update mailing list location
2011-02-08 23:58:56 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Use nominal bitrate if bitrate tag is unavailable
If the bitrate tag is unavailable, this falls back to the nominal
bitrate tag instead, if that is present.
https://bugzilla.gnome.org/show_bug.cgi?id=641860
2011-02-08 12:31:34 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin2.c:
decodebin2: caps can be NULL
Don't use and unref NULL caps.
2011-02-02 16:49:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: also push cached events downstream when operating in pull mode
Otherwise, having 2 tagdemux in a row followed by an element operating in
pull mode will make the second tagdemux implictly eat the first tagdemux'
tag event(s).
Fixes (part of) #641047.
2011-01-21 18:10:29 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: ensure serialnos are unique
We do that by checking a newly generated one is not already used in
an existing stream, and doing it again if it is.
https://bugzilla.gnome.org/show_bug.cgi?id=640211
2011-02-02 17:30:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: free stream map caps when done
2011-02-02 17:23:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: keep IN_CAPS flag check for header buffers as fallback
In case the ogg mapper doesn't handle all the accepted input formats
(although it really should). Saves us error handling for that case
though. Also log caps properly.
https://bugzilla.gnome.org/show_bug.cgi?id=629196
2011-01-21 16:05:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: use oggstream for less brittleness in recognizing headers
Using the IN_CAPS flag for this is brittle, and will fail if either
vorbisparse or vorbistag (which is itself based on vorbisparse) is
inserted between oggdemux and oggmux. Possibly other elements too
(eg, theoraparse, etc).
Using oggstream ensures we Get It Right More Often Than Not.
https://bugzilla.gnome.org/show_bug.cgi?id=629196
2011-02-02 15:33:36 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: fix copy-and-paste typo in property docs
2011-01-21 10:56:00 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
oggmux: do not skip a pageno at start
Discontinuities are automatically signalled by oggdemux at the start
of a new stream. When oggmux is yet to output actual data pages,
do not signal these discontinuities in the ogg stream.
This patch may miss some actual discontinuities at the very start of
a stream, but avoids the spurious missing pages when encoding happens
normally.
A better fix might involve finding a way to distinguish between actual
data discontinuities and discontinuities merely marking the start of
a new stream.
Fixes an issue with ogg page numbering (would skip a number for no
reason, which then looks like a packet was lost somewhere) when
re-muxing an ogg stream, e.g. when re-tagging in rhythmbox.
https://bugzilla.gnome.org/show_bug.cgi?id=629196
2011-02-01 15:57:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
theoraenc: clean up property descriptions
Remove "This property requires libtheora version >= 1.1" qualifiers
from property descriptions. They aren't needed any longer now that
we require libtheora >= 1.1.
2010-08-19 22:31:07 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
id3tag: map the ID3v2 TENC frame to GST_TAG_ENCODED_BY
https://bugzilla.gnome.org/show_bug.cgi?id=627268
2011-01-29 20:43:08 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/tcp/gsttcpserversink.c:
tcp: use socklen_t where appropriate rather than specific type
In particular, fixes Cygwin build where socklen_t is defined as int
in line with native win32 api definition.
2011-01-29 19:40:23 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: cast argument to isdigit to int
... as that is the specification and fixes compilation on Cygwin:
gstxmptaag.c: In function 'read_one_tag':
gstxmptag.c:1015: error: array subscript has type 'char'
2011-01-31 18:06:18 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
* tests/check/elements/appsink.c:
appsink: add buffer fallback in case the application doesn't handle buffer lists
We shouldn't assume the application handles buffer lists, for
ease-of-use reasons and for backwards compatibility reasons.
2011-01-26 10:32:32 +0800 Cai Yuanqing <Yuanqing.Cai@tieto.com>
* gst-libs/gst/app/gstappsink.c:
appsink: send new-buffer-list signal
Send new-buffer-list signal when emit-signals is TRUE
https://bugzilla.gnome.org/show_bug.cgi?id=640607
2011-01-20 16:25:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: also add https to buffer protocols
HTTPS also needs buffering.
2011-01-30 15:40:53 +0200 Felipe Contreras <felipe.contreras@nokia.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: trivial cleanups
It seems these stuff was neglected from commmit d8942e2.
Signed-off-by: Felipe Contreras <felipe.contreras@nokia.com>
2011-01-27 15:26:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/config.h:
win32: fix DEFAULT_AUDIOSINK, should be direct*sound*sink
https://bugzilla.gnome.org/show_bug.cgi?id=640705
2011-01-27 12:32:35 +0100 Philippe Normand <pnormand@igalia.com>
* gst/typefind/gsttypefindfunctions.c:
typefinding: register H264 typefinder with H264 caps
https://bugzilla.gnome.org/show_bug.cgi?id=640709
2011-01-26 12:16:58 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gststreamsplitter.c:
streamsplitter: release pending events refs
Unref pending events when disposing the streamsplitter.
Also refactor a little to replace a for with a g_list_foreach
2011-01-26 15:42:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
tests: don't run encodebin test if vorbis or theora plugins aren't available
2011-01-26 09:07:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for degas images
2011-01-26 09:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: use image/x-icon as media type for ICON files
That's what we've been using so far (e.g. gdkpixbufdec).
2011-01-18 10:20:29 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/examples/snapshot/snapshot.c:
snapshot: use a keyframe seek
One would usualy get good quality snapshots quickly. The exact seek position
does not really matter.
2011-01-17 23:13:29 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/examples/snapshot/snapshot.c:
snapshot: add a newline to the usage and error output
2011-01-25 18:03:23 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin2.c:
decodebin2: add comment and whitespace trimming
2011-01-12 14:03:12 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for x-annodex
2011-01-25 13:39:25 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: add typefinder for DEGAS images
This fixes at least one DEGAS image from being misdetected as DTS audio.
https://bugzilla.gnome.org/show_bug.cgi?id=625129
2011-01-21 14:56:28 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: cleanup
Remove a pointless string concatentation, and fix an off-by-one in
packetno in a log.
https://bugzilla.gnome.org/show_bug.cgi?id=640189
2011-01-24 11:45:21 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: add detection for windows icon files to get them out of the way
Some of them can otherwise be misdetected for MPEG audio.
https://bugzilla.gnome.org/show_bug.cgi?id=620364
2011-01-17 15:11:15 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* ext/ogg/gstoggdemux.c:
oggdemux: Remove dead code
2011-01-11 15:10:42 +0800 Yang Xichuan <xichuan.yang@tieto.com>
* ext/ogg/gstoggparse.c:
oggparse: Make gst_ogg_parse_submit_buffer() safe
By not passing zero-sized buffers to ogg_sync_buffer()
and checking the return values of libogg functions.
Fixes bug #639136.
2011-01-11 18:18:34 +0100 Lane Brooks <dirjud@gmail.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: Add support for vertical center alignment
Fixes bug #639159.
2011-01-24 15:21:10 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Move static variable to local function
Variable was being written to and could cause crashes
if multiple elements were parsing xmp at the same time.
Moving it to local scope solves the problem.
2011-01-24 18:27:30 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: Add support for video/x-camstudio
2011-01-24 00:00:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development
=== release 0.10.32 ===
2011-01-21 10:50:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.32
2011-01-18 10:45:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.31.4 pre-releases
2011-01-18 10:44:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
docs: update docs
2011-01-18 10:40:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
* tests/check/libs/profile.c:
encoding-target: change keyfile header to 'GStreamer Encoding Target'
which is more in line with other files such as .desktop files.
2011-01-18 01:06:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
pbutils: don't assume LC_MESSAGES is always defined, also check for ENABLE_NLS
Should fix build with mingw32 build bot again.
2011-01-18 00:09:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* win32/common/libgstapp.def:
app: export gst_app_stream_type_get_type()
API: gst_app_stream_type_get_type()
API: GST_TYPE_APP_STREAM_TYPE
https://bugzilla.gnome.org/show_bug.cgi?id=639747
2011-01-17 23:59:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/gstappbuffer.c:
app: make GstAppBuffer get_type() function thread-safe
2011-01-18 01:09:53 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Drop new stream tags once preroll is done
This makes sure we do not touch the stream taglist once the pipeline has
been prerolled. Adding of stream tags happens in the pad event probe
which runs in a different thread from discoverer stream processing, so
modifying the tag list while discoverer might be processing it can
sometimes cause a crash.
https://bugzilla.gnome.org/show_bug.cgi?id=639778
2011-01-17 15:30:08 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Validate timeouts before processing them
This avoids a race where the timeout callback is scheduled to run but we
get sufficient information to finish discovery before actually getting
around to executing the callback. See the documentation of
g_source_is_destroyed() for more details.
https://bugzilla.gnome.org/show_bug.cgi?id=639730
2011-01-18 00:08:32 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Make sure we call _stop() before being freed
This ensures that everything is properly cleaned up before the
GstDiscoverer object is freed. Specifically, it makes sure that we've
removed the async timeout callback before freeing the object to avoid a
potential crash later on.
https://bugzilla.gnome.org/show_bug.cgi?id=639755
2011-01-16 14:55:46 -0800 David Schleef <ds@schleef.org>
* gst/gdp/gstgdppay.c:
gdppay: make newsegment buffer metadata writable
2011-01-16 16:46:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
pbutils: save localised strings properly when writing encoding targets to a file
Use LC_MESSAGES rather than LC_ALL. Save/load description as untranslated string
when using an English language locale. Strip locale information to the language,
so we don't save keys like description[fr_FR.UTF-8]=...
https://bugzilla.gnome.org/show_bug.cgi?id=638860
2011-01-13 13:59:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: set framed=false on DTS caps
2011-01-12 17:51:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-target.c:
docs: add some more Since: markers for new encoding-profile API
2011-01-12 15:51:52 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: require gobject-introspection >= 0.9.12
Earlier versions don't honour the -L/--library-path option,
which we need. See commit 4d0ccdad in gobject-introspection git.
Should "fix" build on lucid/maverick build bots.
2011-01-11 19:19:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.31.3 pre-release
2011-01-11 18:59:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/da.po:
* po/gl.po:
* po/pt_BR.po:
po: update translations
2011-01-11 14:41:53 +0000 Bastien Nocera <hadess@hadess.net>
* tests/examples/seek/jsseek.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
examples: allow building with newer GTK+
GtkFunction is gone, and there's no update policies for
GtkRanges any more (but the default was continuous anyway,
so no need to set it to that mode explicitly).
https://bugzilla.gnome.org/show_bug.cgi?id=639215
2011-01-11 14:59:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/Makefile.am:
gobject-introspection: pass --library-path as well to make it find the right libgstreamer
Makes things work again properly in uninstalled setups (and
presumably in installed setups where GStreamer is installed
into a non-standard prefix). Requires fixes from core git.
https://bugzilla.gnome.org/show_bug.cgi?id=639039
2011-01-11 14:52:51 +0000 Byeong-ryeol Kim <brofkims@gmail.com>
* gst-libs/gst/pbutils/Makefile.am:
gobject-introspection: fix issue when gold linker is used
Need to pass libgstreamer-0.10 explicitly to linker, since we're
calling gst_init(), which in turn is needed because the encoding
target get_type() function calls gst_value_register().
https://bugzilla.gnome.org/show_bug.cgi?id=639039
2011-01-11 15:49:54 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From e572c87 to f94d739
2011-01-10 16:35:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From ccbaa85 to e572c87
2011-01-10 14:53:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 46445ad to ccbaa85
2011-01-10 15:55:26 +0800 Yang Xichuan <xichuan.yang@tieto.com>
* ext/ogg/gstoggdemux.c:
oggdemux: remove outdated comment
https://bugzilla.gnome.org/show_bug.cgi?id=639121
2011-01-08 02:16:19 +0000 Koop Mast <kwm@FreeBSD.org>
* configure.ac:
configure: fix bash-ism
https://bugzilla.gnome.org/show_bug.cgi?id=638961
2011-01-08 02:10:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
gobject-introspection: use same PKG_CONFIG_PATH for g-ir-compiler as for g-ir-scanner
Make sure to use the PKG_CONFIG_PATH set at configure time instead of
just relying on an env-var set one. This makes sure both g-ir-compiler
and g-ir-scanner use the same PKG_CONFIG_PATH for determining include
paths etc.
2011-01-08 01:12:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* pkgconfig/gstreamer-app-uninstalled.pc.in:
* pkgconfig/gstreamer-app.pc.in:
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
* pkgconfig/gstreamer-audio.pc.in:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
* pkgconfig/gstreamer-cdda.pc.in:
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-fft.pc.in:
* pkgconfig/gstreamer-floatcast.pc.in:
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
* pkgconfig/gstreamer-interfaces.pc.in:
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
* pkgconfig/gstreamer-netbuffer.pc.in:
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-pbutils.pc.in:
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
* pkgconfig/gstreamer-riff.pc.in:
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtp.pc.in:
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp.pc.in:
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
* pkgconfig/gstreamer-sdp.pc.in:
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
* pkgconfig/gstreamer-tag.pc.in:
* pkgconfig/gstreamer-video-uninstalled.pc.in:
* pkgconfig/gstreamer-video.pc.in:
pkg-config: add girdir and typelibdir variables to .pc files
We need them when building gir and typelib files for
libraries that depend on these, such as gst-rtsp-server
for example, in an uninstalled setup.
2011-01-07 12:50:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/pbutils-enumtypes.c:
* win32/common/video-enumtypes.c:
0.10.31.2 pre-release
2011-01-07 13:04:11 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/encoding/gstencodebin.c:
* gst/encoding/gstencodebin.h:
encodebin: Add missing-plugin support
https://bugzilla.gnome.org/show_bug.cgi?id=638903
2011-01-07 12:51:11 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Extend documentation
https://bugzilla.gnome.org/show_bug.cgi?id=638901
2011-01-07 00:43:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
tests: never disable g_assert() and cast checks for the unit tests
The unit tests are riddled with g_assert() and friends, sometimes
containing functional code like set_state() calls in them even
(looking at you, pipeline/capsfilter-renegotiation). Make sure we
don't disable assert and cast checks for the unit tests even if
this has been specified for the rest of the code base, e.g. via
--disable-glib-asserts.
2011-01-06 23:17:12 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstpbutils.def:
win32: udpate pbutils .def file for API change
2011-01-06 23:13:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
docs: update docs
2011-01-06 23:13:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/fi.po:
* po/ru.po:
po: update translations
2011-01-06 23:08:34 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: make text property controllable too
Because we can, and because it's the most interesting one
to control really, after xpos/ypos.
2011-01-06 23:01:20 +0000 Lane Brooks <dirjud@gmail.com>
* ext/pango/Makefile.am:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: make some properties controllable
https://bugzilla.gnome.org/show_bug.cgi?id=638859
2011-01-06 20:37:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/.gitignore:
tests: ignore new rtsp test binary
2011-01-05 15:54:15 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: ignore header pages when looking for keyframe
This was causing keyframe_granule to be set to 0 for all streams
when seeking to the beginning of the stream, i.e., at the
beginning of playback. Fixes #619778.
2010-12-29 15:27:44 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: when the last keyframe position is not known, do not use -1
Instead, use either 0 or 1, depending on bitstream version, which give
the correct result for streams which aren't cut off at start.
This allows that function to not return negative granpos.
https://bugzilla.gnome.org/show_bug.cgi?id=638276
2011-01-06 17:57:41 +0000 christian schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Update spec file with discoverer and encodebinchanges
2011-01-05 15:53:09 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
discoverer: Documentation updates
Some cosmetic changes and expands on some bits of the documentation to
make it more newbie-friendly.
2011-01-06 13:08:53 +0100 Robert Swain <robert.swain@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: Fix behaviour for frame rate cap changes
The outgoing buffer timestamp is calculated by scaling an output buffer
count by the src pad frame rate caps. If these caps change, we need to
reset the count and work from a new base timestamp. The new output
buffer timestamp is then the count scaled by the new caps values added
onto the base timestamp.
2011-01-06 08:47:04 +0100 Edward Hervey <bilboed@bilboed.com>
* tools/gst-discoverer.c:
tools: Improve pretty-printing of tags
Avoids escaping strings for nothing and printing out useless buffer contents.
2011-01-06 08:46:42 +0100 Edward Hervey <bilboed@bilboed.com>
* tools/gst-discoverer.c:
tools: don't leak the GMainLoop
2011-01-06 00:28:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
pbutils: config.h include should come before all other includes
2011-01-05 22:02:35 +0100 Edward Hervey <bilboed@bilboed.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-profile.h:
* gst/encoding/gstencodebin.c:
* tests/check/libs/profile.c:
* tests/examples/encoding/encoding.c:
encoding: encoding_profile_get_output_caps => _get_input_caps
Makes more sense name-wise
2011-01-05 20:40:39 +0100 Edward Hervey <bilboed@bilboed.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: Add various new symbols
2011-01-05 01:50:34 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-target.c:
encoding-profile: Minor documentation updates
2011-01-03 19:07:45 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Give a better usage example
2011-01-03 18:52:00 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
* tests/check/libs/profile.c:
* win32/common/libgstpbutils.def:
encoding-target: Fixup loading/saving methods
2011-01-03 18:51:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
encoding-target: more docs cleanups
2011-01-03 16:07:49 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
* tests/check/libs/profile.c:
encoding-target: Change target suffix to .gep
Along with a bunch of other internal cleanups
2011-01-03 13:21:26 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
encoding-target: Add more docs regarding categories
2011-01-03 13:20:19 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
* tests/check/libs/profile.c:
* win32/common/libgstpbutils.def:
encoding-target: Add API for list all categories and targets
API: gst_encoding_list_available_categories
API: gst_encoding_list_all_targets
2010-12-22 18:18:00 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-profile.h:
* tests/check/libs/profile.c:
* win32/common/libgstpbutils.def:
encoding-profile: Add convenience method to find a profile
API: gst_encoding_profile_find
2010-12-22 18:16:33 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* configure.ac:
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
* tests/check/libs/profile.c:
encoding-target: Implement save/load feature
Fixes #637735
2010-12-22 11:41:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
* tests/check/libs/profile.c:
* win32/common/libgstpbutils.def:
encoding-target: Add method to get a profile by name
API: gst_encoding_target_get_profile
2011-01-05 19:30:50 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Convert to new GstElementClass::request_new_pad_full vmethod
2011-01-05 15:31:09 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/pbutils.h:
pbutils: Don't forget to include the encoding headers
2011-01-05 12:02:02 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Fix uninitialized variables
reported by macosx gcc
2010-12-07 14:59:46 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
codec-utils: Minor documentation changes
2011-01-02 15:48:47 -0800 David Schleef <ds@schleef.org>
* gst/typefind/gsttypefindfunctions.c:
typefind: Add stream-format to h264 caps
2011-01-02 17:21:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: default to enable-last-buffer=FALSE for audio sinks
There isn't really any good reason to get the last buffer from an
audio sink, so don't make the sink keep it around unnecessarily.
2010-12-31 12:14:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* gst/playback/Makefile.am:
* gst/playback/gstinputselector.c:
* gst/playback/gstinputselector.h:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
playbin2: use input-selector from core instead of internal copy
2010-12-31 01:24:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
tests: add input-selector-test and output-selector-test
Moved from gst-plugins-bad into -base, becasue it uses videotestsrc
and other elements from -base, so it can't be in core.
2010-11-24 12:22:01 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/icles/output-selector-test.c:
output-selector-test: don't hardcode videosinks and use more colorspace conv.
Use autovideosink instead of hardcoded sinks. Use an additional colorspace
converter between videotestsrc and timeoverlay.
2009-10-27 11:51:05 -0700 Michael Smith <msmith@songbirdnest.com>
* tests/icles/output-selector-test.c:
tests: Remove executable bits from non-executable files.
2009-02-24 16:33:51 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/icles/input-selector-test.c:
tests: move examples directory to tests/examples as in every other GStreamer module
2008-06-19 13:18:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests: Use BOILERPLATE macro and update output-selector test to the latest api changes.
Original commit message from CVS:
* gst/selector/gstoutputselector.c:
* tests/icles/output-selector-test.c:
Use BOILERPLATE macro and update test to the latest api changes.
2008-02-07 13:48:20 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/icles/output-selector-test.c: Add a fixme comment.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
Add a fixme comment.
* gst/selector/gstoutputselector.c:
Fix same leak as in input-selector.
* tests/icles/output-selector-test.c:
Improve the test.
2008-01-29 07:38:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Replace the switch plugin with the selector plugin. Add output-selector as the opposite of input-selector (was switc...
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-switch.xml:
* gst/selector/.cvsignore:
* gst/selector/Makefile.am:
* gst/selector/gstinputselector.c:
* gst/selector/gstinputselector.h:
* gst/selector/gstoutputselector.c:
* gst/selector/gstoutputselector.h:
* gst/selector/gstselector-marshal.list:
* gst/selector/gstselector.c:
* gst/selector/selector.vcproj:
* gst/switch/.cvsignore:
* gst/switch/Makefile.am:
* gst/switch/gstswitch-marshal.list:
* gst/switch/gstswitch.c:
* gst/switch/gstswitch.h:
* gst/switch/switch.vcproj:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/output-selector-test.c:
Replace the switch plugin with the selector plugin. Add output-
selector as the opposite of input-selectoo (was switch). Add a test
for output-selector. Add docs for the elements. The vcproj needs
update. Fixes #500142.
2010-12-30 18:08:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
baseaudiopay: fix timestamps on buffer lists
Fix the outgoing timestamps and RTP timestamps on outgoing buffers when using
buffer lists.
2010-12-29 22:36:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: assume EBML files without doctype are matroska
https://bugzilla.gnome.org/show_bug.cgi?id=638019
2010-12-29 12:53:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/tcp/gstmultifdsink.c:
multifdsink: only keep last valid timestamp
Fixes #634397
2010-10-13 17:09:13 +0200 Andoni Morales Alastruey <amorales@flumotion.com>
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
multifdsink: add first and last buffer's timestamp to the stats
2010-12-29 11:51:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: fix typo in comment
2010-12-28 17:39:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: fix interpretation of Theora granule position
The offset part of the granpos is not a sign of the newer encoding.
Use the version number instead.
This fixes the criticals thrown by theoraparse, and (at last) the
remaining part of #553244.
2010-11-25 17:01:04 +0100 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: protect against ringbuffer disappearing while in a query
Observed a case where the sink went to null-state during the query,
hence the ringbuffer-pointer was NULL, causing a crash.
Moving the ringbuffer-check code until after the query, and hold the
lock during the check and while using the spec-values. It should not matter
to the query wether the ringbuffer is present or not, and it actually
gets a time bit more time to get the ringbuffer set up in this case!
Fixes #635231
2010-12-28 19:39:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: handle pads that are not added yet
Don't try to stream data on pads that are not added yet. This happens while we
discover the different streams.
2010-12-28 11:41:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
basedepay: fix refcounting issue
Make sure that when _make_writable() returns a new buffer, we actually push that
one instead of the old one.
2010-12-25 15:22:42 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: implement tag extraction for Kate streams
This will mainly allow Totem to know the language of those streams,
so the subtitle selection menu gets properly filled out.
https://bugzilla.gnome.org/show_bug.cgi?id=638005
2010-12-26 17:29:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for DVB subtitle caps
2010-12-23 17:18:17 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: set headers on caps
This will allow switching from one stream to another without having to send
the headers for the new stream again.
https://bugzilla.gnome.org/show_bug.cgi?id=637927
2010-12-22 15:29:56 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggstream.c:
oggstream: Fix parsing of theora size
2010-12-22 19:06:56 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Don't use gst_pad_alloc_buffer()
allocate buffers using gst_buffer_new_and_alloc() instead of
gst_pad_alloc_buffer_and_set_caps(), as the first one will
cause the pad to block, and we don't want that since that will
prevent subsequent pads from being fed if a block occurs at
start, when all pads must be fed for playback to start.
This fixes autoplugging of the tiger element and other things.
https://bugzilla.gnome.org/show_bug.cgi?id=637822
2010-12-22 18:12:14 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Also use "Formatter"s for container formats
2010-12-22 18:19:48 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
encoding-target: Fix typo
2010-12-22 10:32:03 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Fix unitialized data warning
Fixes a valgrind warning on jifmux tests on -bad caused by
unitialized bytes.
Fixes #637758
2010-12-22 13:56:12 +0100 Alessandro Decina <alessandro.d@gmail.com>
* gst/encoding/gstencodebin.c:
encodebin: minor fix in error handling.
Don't call gst_bin_remove (bin, <invalid pointer>).
2010-12-21 18:51:29 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.c:
pbutils: More gtk-doc annotations
2010-12-21 10:26:40 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: delay stream-changed messages
https://bugzilla.gnome.org/show_bug.cgi?id=637586
2010-12-21 16:33:50 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-target.c:
* tests/check/libs/profile.c:
encoding-target: Ensure target names and categories are valid
2010-12-21 15:11:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.h:
depay: update some docs
2010-12-21 15:02:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
rtpdepayloade: add support for getting events
Add support for intercepting sink events in the depayloader by adding a new
vmethod.
2010-12-21 13:37:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: keep timestamps when no decoded output
Keep track of the timestamps even when we didn't generate decodable output.
2010-12-21 13:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: avoid using invalid timestamps
2010-12-21 10:41:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: don't pause for live buffering messages
2010-12-20 18:29:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppay: use RTP base time when invalid timestamps
When we have an invalid running-time (because we clipped, for example) use the
RTP base time for timestamping instead of generating wrong RTP timestamps.
2010-12-20 18:28:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
rtppayload: copy applied rate to segment
Use set_segment_full to copy all segment values to the segment structure.
2010-12-21 13:09:34 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* tests/check/elements/encodebin.c:
* tests/check/libs/profile.c:
tests: Update container-less profile checks
2010-12-21 13:08:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Add guard against profiles without format
2010-12-21 13:07:27 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/encoding/gstencodebin.c:
encodebin: Fix usage of non-container profiles
2010-12-17 16:10:53 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/plugins/inspect/plugin-videoscale.xml:
docs: Update for videoscale class changes
2010-12-20 17:46:48 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* common:
Automatic update of common submodule
From 169462a to 46445ad
2010-12-19 13:41:22 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
gstdiscoverer: Don't leak tags
2010-12-19 13:22:23 +0100 Edward Hervey <bilboed@bilboed.com>
* tools/gst-discoverer.c:
gst-discoverer: show global tags by default
2010-12-19 09:53:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/rtsp.c:
rtsp: Fix memory leaks in the gst_rtsp_url_decode_path_components() unit tests
2010-12-18 20:47:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/encoding/Makefile.am:
examples: Fix encodebin example CFLAGS and LDFLAGS
Previously it would only succeed to link if a new enough
libgstpbutils-0.10 was installed in the default library
search path.
2010-12-17 14:16:18 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
ogg: implement packet duration query for kate streams
https://bugzilla.gnome.org/show_bug.cgi?id=637519
2010-12-17 19:06:27 -0600 Rob Clark <rob@ti.com>
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-profile.h:
* gst/encoding/gstencodebin.c:
fix compile errors on macosx
with i686-apple-darwin10-gcc-4.2.1:
encoding-profile.h:134: warning: type qualifiers ignored on function return type
encoding-profile.c:240: warning: type qualifiers ignored on function return type
gstencodebin.c: In function 'next_unused_stream_profile':
gstencodebin.c:454: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
gstencodebin.c:464: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
2010-12-17 00:49:26 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.c:
audioresample: corrected buffer duration calculation to account for nonzero initial timestamp
Since we calculate timestamps by:
timestamp = t0 + (out samples) / (out rate)
and durations by:
duration = ((out samples) + (processed samples)) / (out rate) - timestamp
if t0 is nonzero, this would simplify to
duration = t0 + (processed samples) / (out rate).
This duration is too large by the amount t0. We should have done:
duration = t0 + ((out samples) + (processed samples)) / (out rate) - timestamp
so that
duration = (processed samples) / (out rate).
2010-12-16 20:40:33 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.h:
audioresample: changed num_gap_samples, num_nongap_samples from guint32 to guint64 so that gaps of greater than or equal to 2^32 samples do not cause integer overflow
2010-12-16 20:38:31 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.c:
audioresample: push half a history length, instead of a full history length, at end-of-stream so that output segment and input segment have same duration
2010-12-16 20:34:13 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
audioresample: renamed count_gap, count_nongap to more descriptive num_gap_samples, num_nongap_samples
2010-12-16 20:32:07 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.c:
audioresample: replaced void* with gpointer
2010-12-16 20:30:24 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.c:
audioresample: initial filter transient discarded; unit tests passing
2010-12-16 20:09:58 -0800 Leo Singer <leo.singer@ligo.org>
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/speex_resampler.h:
* gst/audioresample/speex_resampler_wrapper.h:
Revert "Revert "audioresample: Add GAP flag support""
This reverts commit 35c76b3409dde7f2dcc8232388a47a1b99b661a7.
Conflicts:
gst/audioresample/gstaudioresample.c
gst/audioresample/gstaudioresample.h
2010-12-16 10:26:43 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
timeoverlay: add missing break
https://bugzilla.gnome.org/show_bug.cgi?id=637377
2010-12-16 10:11:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Change classification to Filter/Converter/Video/Scaler
2010-12-15 23:47:29 +0200 Stefan Kost <ensonic@users.sf.net>
* win32/common/libgstrtsp.def:
win32: update the def file with the new rtsp api
2010-12-15 17:51:36 +0100 Andy Wingo <wingo@oblong.com>
add gst_rtsp_url_decode_path_components
* gst-libs/gst/rtsp/gstrtspurl.h:
* gst-libs/gst/rtsp/gstrtspurl.c (gst_rtsp_url_decode_path_components):
New public function, returns a strv of uri-decoded path components.
* tests/check/Makefile.am:
* tests/check/libs/rtsp.c: Add tests.
2010-12-15 16:35:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtp.def:
win32: update defs file
2010-12-15 16:30:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtpbuffer: relax arrangement for RTP bufferlists
Don't assume there are exactly 2 buffers but allow cases where the header and
payload are in 1 buffer or where the payload is in more buffers.
2010-12-15 14:55:34 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 20742ae to 169462a
2010-12-15 12:58:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
basedepay: add support for buffer lists in the depayloader
Add support for buffer lists in the depayloader.
2010-09-13 10:08:47 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/encoding/.gitignore:
* tests/examples/encoding/Makefile.am:
* tests/examples/encoding/encoding.c:
* tests/examples/encoding/gstcapslist.c:
* tests/examples/encoding/gstcapslist.h:
examples: encoding example
Along with gstcapslist
2010-08-13 17:36:38 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* gst/encoding/.gitignore:
* gst/encoding/Makefile.am:
* gst/encoding/gstencode-marshal.list:
* gst/encoding/gstencodebin.c:
* gst/encoding/gstencodebin.h:
* gst/encoding/gstsmartencoder.c:
* gst/encoding/gstsmartencoder.h:
* gst/encoding/gststreamcombiner.c:
* gst/encoding/gststreamcombiner.h:
* gst/encoding/gststreamsplitter.c:
* gst/encoding/gststreamsplitter.h:
* tests/check/Makefile.am:
* tests/check/elements/.gitignore:
* tests/check/elements/encodebin.c:
gst: New encoding plugin
https://bugzilla.gnome.org/show_bug.cgi?id=627476
2010-08-13 17:27:52 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/design/Makefile.am:
* docs/design/design-encoding.txt:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/encoding-profile.c:
* gst-libs/gst/pbutils/encoding-profile.h:
* gst-libs/gst/pbutils/encoding-target.c:
* gst-libs/gst/pbutils/encoding-target.h:
* tests/check/Makefile.am:
* tests/check/libs/.gitignore:
* tests/check/libs/profile.c:
* win32/common/libgstpbutils.def:
pbutils: New Profile library
https://bugzilla.gnome.org/show_bug.cgi?id=627476
2010-12-15 12:21:05 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
configure: use the -Bsymbolic-functions linker flag if supported
This feature turns intra library calls into direct function calls and thus makes
them a little faster. The downside is that this causes problems for e.g.
LD_PRELOAD based tools. Thus add a configure option to turn it off.
2010-12-14 00:16:13 -0800 David Schleef <ds@schleef.org>
* gst/typefind/gsttypefindfunctions.c:
typefind: Add check for yuv4mpeg
2010-12-13 18:05:41 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: spell out two more container formats
2010-12-13 16:20:23 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* gst-libs/gst/pbutils/pbutils-private.h:
* tools/gst-discoverer.c:
* win32/common/libgstpbutils.def:
discoverer: query seekability
Besides the duration we can also query the seekability of a stream. Use the new
API in the gst-discoverer tool.
API: gst_discoverer_info_get_seekable
2010-12-13 16:23:04 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 011bcc8 to 20742ae
2010-12-13 13:04:40 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/check/elements/audioresample.c:
tests: audioresample: adjust unit test to relaxed discont checking
2010-12-13 12:34:58 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/Makefile.am:
* docs/design/Makefile.am:
make: move the design doc also on the Makefile.am level (for dist)
2010-12-13 10:05:00 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
audioresample: relax discont checking slightly
2010-12-13 09:56:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
audioresample: provide as much valid output ts and offset as valid input
... by independently tracking time and offset, rather than having no offset
leading to no output ts.
2010-12-13 10:41:24 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/typefind/gsttypefindfunctions.c:
typefinders: name "aac" typefinder "audio/aac"
This is in sync how we call the others.
2010-12-13 09:58:53 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/design-audiosinks.txt:
* docs/design/design-audiosinks.txt:
docs: move design doc to design folder
2010-12-11 19:33:33 +0200 Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
* gst/videotestsrc/generate_sine_table.c:
videotestsrc: Add a missing return statement
2010-12-11 17:18:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Deprecate new-decoded-pad and removed-decoded-pad signals
They're really the same as pad-added and pad-removed from GstElement
and it doesn't make sense to have two signals for the same thing.
2010-12-11 17:14:36 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Emit "remove-decoded-pad" signal when pads are removed from decodebin2
Fixes bug #636198.
2010-12-10 18:57:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
appsink: unset flushing flag when starting
When we start again after being stopped, clear the flushing flag or else
it will always be TRUE.
Fixes #636769
2010-12-09 16:57:35 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add/Fix some media descriptions
Fixes #623413
2010-12-09 08:40:25 +0100 Gavin Stark <g.stark@realdigitalmedia.com>
* sys/xvimage/xvimagesink.c:
xvimagesink: Use gst_caps_can_intersect() instead of gst_caps_intersect()
Fixes a memory leak and bug #636827.
2010-12-08 12:55:24 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: improve iso media typefinding
... by also considering compatible brands rather than only aiming at major brand
(of which there are a seemingly ever expanding great many).
2010-12-08 12:28:32 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/check/libs/pbutils.c:
tests: remove superflous ';' and reindent
2010-12-08 12:09:45 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
docs: fix wrong use of Since: keyword
2010-12-07 20:28:37 +0200 René Stadler <rene.stadler@nokia.com>
* tests/check/gst/typefindfunctions.c:
tests: add AC-3, E-AC-3 typefind tests
2010-12-03 17:33:40 +0200 René Stadler <rene.stadler@nokia.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: ignore AC-3 BSIDs 9, 10 and >16
These are reserved for future extensions which will not be backwards
compatible to E-AC-3.
2010-12-03 16:54:21 +0200 René Stadler <rene.stadler@nokia.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: accept consecutive AC-3 frames of different sizes
This is perfectly valid and occurs in particular when there are
(in)dependent substreams present.
2010-12-03 16:22:32 +0200 René Stadler <rene.stadler@nokia.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: remove useless masking in (E-)AC-3 typefinders
2010-12-03 16:14:15 +0200 René Stadler <rene.stadler@nokia.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: stop scanning after suggesting E-AC-3 caps
2010-12-03 18:08:58 +0200 René Stadler <rene.stadler@nokia.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: fix E-AC-3 frame size parsing
Frame size is given in words; it is already multiplied by two where
needed, so the left shift is superfluous. This extra multiplication
caused the code to inspect the third packet instead of the second,
which would fail for files where the second packet has a size
different from the first.
2010-12-07 17:35:14 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.h:
rtsp: Move around the typedefs to make GIR happy
Otherwise it will generate they symbols as _GstRTSP* (with the leading
underscore).
2010-12-04 14:48:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
* tests/examples/app/appsrc-stream.c:
* tests/examples/app/appsrc-stream2.c:
tests: use GLib 2.22 API unconditionally
2010-12-04 14:45:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/tag/lang.c:
* gst-libs/gst/tag/mklangtables.c:
* gst-libs/gst/video/convertframe.c:
libs: use GLib 2.22 API unconditionally
2010-12-03 17:41:18 +0100 Benjamin Gaignard <benjamin.gaignard@stericsson.com>
* Android.mk:
* android/NOTICE:
* android/alsa.mk:
* android/app.mk:
* android/app_plugin.mk:
* android/audio.mk:
* android/audioconvert.mk:
* android/decodebin.mk:
* android/decodebin2.mk:
* android/gdp.mk:
* android/gst-libs/gst/app/gstapp-marshal.c:
* android/gst-libs/gst/app/gstapp-marshal.h:
* android/gst-libs/gst/audio/audio-enumtypes.c:
* android/gst-libs/gst/audio/audio-enumtypes.h:
* android/gst-libs/gst/interfaces/interfaces-enumtypes.c:
* android/gst-libs/gst/interfaces/interfaces-enumtypes.h:
* android/gst-libs/gst/interfaces/interfaces-marshal.c:
* android/gst-libs/gst/interfaces/interfaces-marshal.h:
* android/gst-libs/gst/pbutils/pbutils-enumtypes.c:
* android/gst-libs/gst/pbutils/pbutils-enumtypes.h:
* android/gst-libs/gst/rtsp/gstrtsp-enumtypes.c:
* android/gst-libs/gst/rtsp/gstrtsp-enumtypes.h:
* android/gst-libs/gst/rtsp/gstrtsp-marshal.c:
* android/gst-libs/gst/rtsp/gstrtsp-marshal.h:
* android/gst-libs/gst/video/video-enumtypes.c:
* android/gst-libs/gst/video/video-enumtypes.h:
* android/gst/playback/gstplay-marshal.c:
* android/gst/playback/gstplay-marshal.h:
* android/gst/tcp/gsttcp-enumtypes.c:
* android/gst/tcp/gsttcp-enumtypes.h:
* android/gst/tcp/gsttcp-marshal.c:
* android/gst/tcp/gsttcp-marshal.h:
* android/interfaces.mk:
* android/netbuffer.mk:
* android/pbutils.mk:
* android/playbin.mk:
* android/queue2.mk:
* android/riff.mk:
* android/rtp.mk:
* android/rtsp.mk:
* android/sdp.mk:
* android/tag.mk:
* android/tcp.mk:
* android/typefindfunctions.mk:
* android/video.mk:
Add build system for Android
2010-12-03 15:46:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstvideo.def:
defs: add new symbol
2010-10-27 13:49:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: additional tag extraction
... supporting theora, flac, speex, celt.
Fixes #629349.
2010-10-27 12:08:25 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggstream: use separate tag extraction vfunction
2010-10-27 11:58:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: refactor vorbis comment tag extraction
2010-10-27 11:16:15 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: plug some oggstream leaks
2010-10-27 10:59:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggstream: streamline tag extraction and prevent some leaks
2010-10-27 10:58:16 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: send stream tags after newsegment and global tags
2010-09-14 23:08:51 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: perform more (vorbis comment header) tag extractions
In particular, move comment header parsing to gstoggstrem.c.
Thanks to Felipe Contreras.
Fixes #629349 (partially).
2010-10-27 10:20:15 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/riff/riff-ids.h:
riff: document omitted field in _gst_riff_strf_auds
(aka WAVEFORMATEX)
2010-10-10 17:15:53 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggstream.c:
oggstream: fix incorrect warning on skeleton headers
2010-11-20 19:02:50 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggparse: Set DELTA_UNIT on buffers
2010-12-03 00:01:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/video.c:
tests: fix video library unit test and skip non-working YUV9/YVU9 parts for now
2010-12-02 23:49:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: add missing break statement for the GST_VIDEO_FORMAT_RGB8_PALETTED case
2010-11-15 22:02:07 +0200 Evan Broder <evan@ebroder.net>
* tools/gst-visualise-m.m:
gst-visualise: trim unused perl dependency
Remove an unused perl module. Fixes #634522.
2010-11-01 23:07:12 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaybin2.c:
playbin2: add some logging for failure case
2010-11-01 23:06:21 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstinputselector.c:
inputselector: log times in human readable form
2010-11-01 22:44:16 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstinputselector.c:
inputselector: more G_PARAM_STATIC_STRINGS use
2010-11-01 22:42:23 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstinputselector.c:
inputselector: move reoccuring logs to LOG and remove a double info
Less debug spew in DEBUG category. No need to log pad again if we use
GST_LOG_OBJECT(pad,...).
2010-12-02 19:11:37 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/rtsp/Makefile.am:
libgstrtsp: Fix typo in .pc to use for GIR
2010-12-02 15:16:25 +0100 Edward Hervey <bilboed@bilboed.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
docs: Add a whole bunch of symbols that were unused to the proper sections
2010-11-10 11:02:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: only parse TTL for IP4 addresses
Only IP4 addresses can have a TTL in the address.
2010-11-10 10:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst-libs/gst/sdp/gstsdpmessage.h:
* win32/common/libgstsdp.def:
sdp: add method to check for multicast addresses
Expose a previously internal method to check for multicast addresses.
See #634093
2010-11-03 11:13:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/pbutils/gstpluginsbaseversion.h.in:
pbutils: Take nano version into account in GST_CHECK_PLUGINS_BASE_VERSION()
If the nano is > 0 the current version should be handled the same as
micro + 1.
2010-11-03 09:51:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add YUV9, YVU9 and IYU1 video formats
API: GST_VIDEO_FORMAT_YUV9: planar 4:1:0 YUV
API: GST_VIDEO_FORMAT_YVU9: planar 4:1:0 YUV (chroma planes swapped)
API: GST_VIDEO_FORMAT_IYU1: packed 4:1:1 YUV (Cr-Y0-Y1-Cb-Y2-Y3)
2010-11-02 11:57:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add 8-bit paletted RGB
API: Add GST_VIDEO_FORMAT_RGB8_PALETTED
API: Add GST_VIDEO_CAPS_RGB8_PALETTED
API: Add gst_video_parse_caps_palette()
2010-10-31 19:17:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gnomevfs/gstgnomevfssrc.c:
gnomevfssrc: Remove dead assignment
2010-10-31 19:14:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/tcp/gsttcp.c:
tcp: Remove dead assignment
2010-10-31 19:11:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: gen_video_chain() always returns a bin, no need to check for that
2010-10-31 19:08:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: If we get EOS for an unknown stream just do nothing
instead of dereferencing NULL pointers. This can happen if the stream
was just removed from the streamsynchronizer in a bad time.
2010-10-31 19:06:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: gen_video_deinterlace_chain() always returns a bin, no need to check that
2010-10-31 19:01:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/v4l/v4l_calls.c:
v4l: If no video tuner is the requested one don't read unitialized data
2010-10-25 14:13:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/ximage/ximagesink.c:
ximagesink: Add docs for the new property
Including Since markers
2010-10-25 14:11:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
xvimagesink: Add docs for the new property
Including Since markers
2010-10-25 14:09:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
xvimagesink: Use PROP_ instead of ARG_ for the property enums
2010-10-25 14:09:20 +0100 Andrea Sebastianutti <andreailseba@virgilio.it>
* sys/xvimage/xvimagesink.c:
xvimagesink: Add read-only properties window-width and window-height
2010-10-25 14:08:43 +0100 Andrea Sebastianutti <andreailseba@virgilio.it>
* sys/ximage/ximagesink.c:
ximagsink: Add read-only properties window-width and window-height
2010-10-17 14:26:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Return correct component width/height for A420
2010-12-02 00:15:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
Bump GLib requirement to >= 2.22
See http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
2010-12-02 00:12:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development
=== release 0.10.31 ===
2010-11-30 19:25:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.31
2010-11-24 17:34:21 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gsturidecodebin.c:
uridecodebin: disconnect signal handlers before disposing
2010-11-22 00:54:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
docs: improve decodebin2 docs a little
Mention that new pads may be created even after no-more-pads.
https://bugzilla.gnome.org/show_bug.cgi?id=634584
2010-11-20 15:45:49 -0800 Evan Nemerson <evan@coeus-group.com>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
introspection: Add information on exported packages to GIRs
https://bugzilla.gnome.org/show_bug.cgi?id=635392
2010-11-18 04:51:56 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
discoverer: Minor documentation fix
docs: Minor discoverer documentation fix
2010-11-18 00:36:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.30.5 pre-release
2010-11-18 00:35:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/bg.po:
* po/ca.po:
* po/es.po:
* po/hu.po:
* po/sk.po:
* po/tr.po:
po: update translations
2010-11-18 00:33:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstdecodebin.c:
decodebin: fix one more pad template ref leak
2010-11-17 10:14:59 +0200 Harri Mähönen <harri.mahonen@gmail.com>
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gsturidecodebin.c:
*decodebin*: don't leak pad templates set on ghostpads
https://bugzilla.gnome.org/show_bug.cgi?id=635067
2010-11-17 01:01:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
playbin2: disable streamsynchronizer magic for this release
Some things aren't quite right yet and cause problems (0-sized buffers
with PREROLL flag set cause crashes in elements that don't expect those;
getting pipeline back to preroll/playing again when audio/video streams
have different lengths and a seek past the end of one of the stream
happens doesn't always work, etc.). Needs further investigation in the
next cycle.
https://bugzilla.gnome.org/show_bug.cgi?id=633700
https://bugzilla.gnome.org/show_bug.cgi?id=634699
2010-11-08 09:27:52 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Fix a gtk-doc gobject-introspection annotation
gst_discoverer_discover_uri() expects the caller to unref the returned
GstDiscovererInfo object. The corresponding gtk-doc annotation was not
updated to reflect this.
2010-11-08 09:26:27 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* tools/gst-discoverer.c:
discoverer: Fix argument type to _container_info_get_streams()
No reason for gst_discoverer_container_info_get_streams() to not take a
GstDiscovererContainerInfo as its argument.
2010-11-05 20:47:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: add --with-gtk option and default to Gtk+ 2.0 while the 3.0 API is still in flux
https://bugzilla.gnome.org/show_bug.cgi?id=634014
2010-11-03 10:35:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix IYU1 support
Fix conversions to IYU1, they allocated infinite amounts of memory before
because no conversion to IYU1 was actually implemented and it was running
into an infinite loop trying to find suitable intermediate formats.
Also fix the stride and sizes used for IYU1.
2010-11-02 12:29:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/rtp.c:
tests: fix invalid free and buffer list leak in rtp library unit test
2010-11-02 12:03:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/tag.c:
tests: fix leak in tag library unit test
2010-11-02 12:01:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: fix leak when parsing undefined EXIF tag into tag list
gst_buffer_set_data() does not set GST_BUFFER_MALLOCDATA, but
the code assumes the buffer takes ownership of the memory
allocated earlier.
2010-11-02 11:57:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: fix GstDateTime leak when converting exif tag to tag list
2010-11-01 17:00:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.30.4 pre-release
2010-11-01 16:59:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/fr.po:
* po/ja.po:
* po/nb.po:
* po/nl.po:
* po/pl.po:
* po/sl.po:
po: update translations
2010-10-30 16:07:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: update docs for discoverer API changes as well
2010-10-30 16:03:18 +0100 Matthias Clasen <mclasen@redhat.com>
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
examples: update some more code for new Gtk+ API, with fallback for older Gtk+ versions
Move code to new Gtk+ 3.x / 2.9x API. We have defines in place
already that make this code work fine on older Gtk+ 2.x.
https://bugzilla.gnome.org/show_bug.cgi?id=632653
2010-10-28 15:13:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
seek: Define the new combobox API to the old functions if using older GTK
https://bugzilla.gnome.org/show_bug.cgi?id=632653
2010-10-30 15:31:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstutils.def:
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstutils.dsp:
win32: remove unused libgstutils stuff
Cruft from before the lib was renamed to pbutils
2010-10-28 18:51:08 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* tools/gst-discoverer.c:
* win32/common/libgstpbutils.def:
discoverer: rename boolean getters for consistency
Rename _get_is_image() to _is_image() and _get_interlaced() to _is_interlaced().
https://bugzilla.gnome.org/show_bug.cgi?id=633311
2010-10-30 12:24:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/pbutils-private.h:
pbutils: remove padding from now-private GstDiscovererInfo structure
2010-10-30 12:03:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* Makefile.am:
* tools/.gitignore:
* tools/Makefile.am:
tools: rename gst-discoverer binary to gst-discoverer-0.10
We're not providing a wrapper like we do for the tools in core,
since wrappers are confusing (e.g. for debugging purposes),
mostly pointless (since the API is likely to change between
major versions), and cause packaging issues when packages for
two different major versions are to be installed in parallel.
https://bugzilla.gnome.org/show_bug.cgi?id=633023
2010-10-30 11:41:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tools/gst-discoverer.c:
tools: update gst-discoverer tool for last-minute API change
https://bugzilla.gnome.org/show_bug.cgi?id=633311
2010-10-29 14:17:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: calculate better timeout value
We want to send the keealive message a little earlier than the timeout value
specifies. Scale this based on the value of the timeout instead of just assuming
5 seconds.
2010-10-29 14:24:54 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: don't let the rtsp connection timeout
Because we should act before the rtsp server does a timeout, we
reduce the timeout-time with 5 seconds, this should be safe to always
keep te rtsp connection alive.
https://bugzilla.gnome.org/show_bug.cgi?id=633455
2010-10-28 15:55:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/elements/videoscale.c:
videoscale: Add unit test for working reverse negotiation
See bug #633147.
2010-10-29 11:48:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix wrong flowreturn handling
Oggdemux will currently try to pad alloc a buffer from the peer when it is
reading the header files. This is a relic from the time where we had an internal
parser and needs to be removed at some point in time.
The problem is that when there is no peer pad yet (which is normal when
collecting headers) we should still continue to parse all the packets of a
page instead of erroring out on NOT_LINKED.
Fixes #632167
2010-10-29 11:47:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
ogg: add some more debug statements
2010-10-26 16:41:28 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaysink.c:
playsink: Fix subpicture overlay when deinterlacing disabled.
Fix a bug when reconfiguring the playsink where the subpicture
stream is broken by attempting to connect it through
streamsynchroniser and second time.
2010-10-28 17:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
discoverer: use const in most of the getters
2010-10-28 03:09:10 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
discoverer: use specific types in getters
Use GstDiscoverer{Audio,Video}Info in getters like
gst_discoverer_{audio,video}_info_get_*(). This avoids the casts in the macros,
help language bindings and is more correct.
2010-10-28 11:56:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
discoverer: Move documentation to the correct section
And don't mention the (not existing) libgstdiscovery.
https://bugzilla.gnome.org/show_bug.cgi?id=633336
2010-10-27 13:16:37 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From 7bbd708 to 011bcc8
2010-10-24 16:09:26 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Get pad caps if we can't get negotiated caps
Better provide something than nothing
https://bugzilla.gnome.org/show_bug.cgi?id=632988
2010-10-24 15:38:30 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
discoverer: Don't ref a NULL caps
https://bugzilla.gnome.org/show_bug.cgi?id=632988
2010-09-24 16:02:42 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't add non prerolled stream to topology
If a final stream didn't preroll, don't add it to the topology since
it doesn't give any information at all.
https://bugzilla.gnome.org/show_bug.cgi?id=632988
2010-10-24 16:17:09 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Description for RealAudio container format
2010-10-24 15:38:42 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for VP6 alpha and ASS subtitle
2010-10-22 17:44:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.30.3 pre-release
2010-10-20 11:01:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
seek: The new combo box text API is available since 2.23.0 and 2.91.1
Only use it conditionally.
2010-10-20 11:01:14 +0200 Matthias Clasen <mclasen@redhat.com>
* tests/examples/seek/jsseek.c:
seek: Don't use deprecated combo box text API
Fixes bug #632653.
2010-10-21 12:24:19 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: workaround internal decodebin2 failing state change
Fixes #632656.
2010-10-21 13:38:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/overlay/gtk-xoverlay.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
tests: don't use deprecated gtk_widget_hide_all()
gtk_widget_hide_all() has been deprecated in gtk+ 2.x and
removed in 2.9x master. Just use gtk_widget_hide() instead.
2010-10-21 13:07:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tools/Makefile.am:
tools: fix linking problems caused by accidentally linking against installed pbutils/gstvideo libs
Fixes build errors in jhbuild:
/foo/build/gst-plugins-base/gst-libs/gst/video/.libs/libgstvideo-0.10.so: undefined reference to `gst_element_factory_list_get_elements'
../gst-libs/gst/pbutils/.libs/libgstpbutils-0.10.so: undefined reference to `gst_element_link_pads_full'
/foo/build/gst-plugins-base/gst-libs/gst/video/.libs/libgstvideo-0.10.so: undefined reference to `gst_element_factory_list_filter'
../gst-libs/gst/pbutils/.libs/libgstpbutils-0.10.so: undefined reference to `gst_pad_link_full'
/foo/build/gst-plugins-base/gst-libs/gst/video/.libs/libgstvideo-0.10.so: undefined reference to `gst_plugin_feature_list_debug'
2010-10-19 00:07:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/pbutils-marshal.list:
* gst-libs/gst/rtsp/gstrtsp-marshal.list:
libs: touch marshal.list files to force rebuild after Makefile.am changes
Force regeneration of marshal.[ch] files after prefix changes in
Makefile.am, to avoid build errors for those of us who don't
habitually make clean first.
2010-10-16 01:08:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/pbutils-enumtypes.c:
* win32/common/pbutils-enumtypes.h:
* win32/common/video-enumtypes.c:
0.10.30.2 pre-release
2010-10-16 01:07:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/LINGUAS:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translations
2010-10-08 17:24:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gsttageditingprivate.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
* tests/check/libs/tag.c:
tag: Adds GST_TAG_CAPTURING_SOURCE
Adds a tag to indicate the source/device used for the capture.
Already maps it in exif and adds tests.
API: GST_TAG_CAPTURING_SOURCE
https://bugzilla.gnome.org/show_bug.cgi?id=631773
2010-10-08 15:51:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gsttageditingprivate.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
* tests/check/libs/tag.c:
tag: Adds GST_TAG_CAPTURING_METERING_MODE
Adds a tag to inform what mode was used by a camera to calculate
the picture capturing exposure
Also adds mapping to exif and tests
API: GST_TAG_CAPTURING_METERING_MODE
https://bugzilla.gnome.org/show_bug.cgi?id=631773
2010-10-08 15:14:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gsttageditingprivate.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
* tests/check/libs/tag.c:
tag: Adds GST_TAG_CAPTURING_SHARPNESS
Adds new tag for tagging sharpness processing used
when capturing an image. Also maps it in the exif
tags.
Tests included.
API: GST_TAG_CAPTURING_SHARPNESS
https://bugzilla.gnome.org/show_bug.cgi?id=631773
2010-10-15 23:54:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspextension.c:
* win32/common/libgstrtsp.def:
rtsp: don't export marshaller function
Make sure the marshaller function isn't exported. As it was
never in a public header file, this should be fine.
2010-10-15 21:22:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/Makefile.am:
pbutils: fix distcheck
Apparently noinst implies dist.
2010-10-15 11:23:02 -0700 David Schleef <ds@schleef.org>
* tests/check/Makefile.am:
tests: Don't dist generated orc code
2010-10-15 11:22:45 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/gstvideoscaleorc-dist.c:
Update generated orc code
2010-10-15 19:18:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/pbutils.h:
* win32/common/libgstpbutils.def:
pbutils: make marshaller private
There's no reason to make the marshaller public API. Don't install
pbutils-marshal.h header file and use prefix that makes sure the
symbol doesn't get exported.
2010-10-15 19:14:49 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/Makefile.am:
pbutils: use fewer variables in Makefile.am to make things clearer
Also fix typo in DISTCLEANFILES.
2010-10-15 17:59:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump Orc requirement to 0.4.11
Has fixes for volume, among other things.
2010-10-15 17:23:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
docs: improve gst_discoverer_new() docs a bit
2010-10-15 16:43:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: private structs need to padding
2010-10-15 11:26:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Fix stupid copy&paste error in last commit
2010-10-13 22:51:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Add support for A420
2010-10-13 20:45:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: API: Add A420 video format
This is planar 4:2:0 YUV plus non-subsampled alpha plane.
2010-10-14 12:31:39 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From 5a668bf to 7bbd708
2010-10-14 16:36:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: fix ADTS caps stream-format detail
Field should be "stream-format", not "stream-type".
2010-07-08 15:22:08 +0200 Andrzej K. Haczewski <ahaczewski@gmail.com>
* gst/typefind/gsttypefindfunctions.c:
typefinding: extend AAC typefinder to detect LOAS streams
Extend AAC typefinder to recognize LOAS stream as specified by
ISO/IEC 14496-3:2009.
https://bugzilla.gnome.org/show_bug.cgi?id=623918
2010-10-13 23:26:35 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gsturidecodebin.c:
*decodebin*: set pad-templates on ghostpads
This makes calling gst_pad_get_pad_template() work.
2010-10-12 21:23:03 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: Update with latest datetime from core
Updates datetime functions to latest APIs in core
2010-10-13 16:12:38 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/theora/gsttheoraparse.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/interfaces/mixertrack.c:
* gst/audioresample/gstaudioresample.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gsturidecodebin.c:
* gst/subparse/gstsubparse.c:
various: add a missing G_PARAM_STATIC_STRINGS flag to object properties
2010-10-13 13:05:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
decodebin2: declare decodebin2 stable, deprecate the old decodebin
https://bugzilla.gnome.org/show_bug.cgi?id=624949
2010-10-13 12:55:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
playbin2: declare stable, deprecate the old playbin
https://bugzilla.gnome.org/show_bug.cgi?id=624949
2010-10-12 16:03:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: only keep last valid granulepos
Only keep the last valid granulepos we see when scanning the last
pages. It is possible that the last page that we inspect has a -1 granulepos, in
which case we want to keep the previous valid time instead.
Fixes #631703
2010-10-10 15:22:52 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: Fix check for last page
2010-10-10 15:22:04 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: change checks from is_skeleton to is_sparse
2010-10-10 15:17:31 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: move is_sparse into stream map
2010-10-11 18:06:18 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/Makefile.am:
tests: vorbis: adds missing lib
Adds missing lib to vorbis check tests makefile
2010-10-11 14:30:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Set GST_ELEMENT_IS_SOURCE flag
uridecodebin behaves like a source, let's mark it as a source
2010-10-10 00:52:13 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/gsttheoradec.c:
theoradec: expose telemetry properties only if libtheora was compiled with --enable-telemetry
Since this is just a debugging feature and libtheora will usually not be
compiled with that option enabled, we should maybe just hide these properties,
since they won't work anyway, and avoid confusing warnings.
Also rename properties to make them less cryptic.
https://bugzilla.gnome.org/show_bug.cgi?id=628488
2010-10-09 23:49:35 +0100 Alexey Fisher <bug-track@fisher-privat.net>
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoradec.h:
theoradec: add properties to enable debugging telemetry overlay
The theora decoder can overlay debugging information on the output
video. This functionality is only available if libtheora has been
compiled with --enable-telemetry. For more details see:
http://people.xiph.org/~xiphmont/demo/theora/demo2.html
Based on original patch by Michael Smith <msmith at xiph org>
https://bugzilla.gnome.org/show_bug.cgi?id=628488
2010-10-10 18:35:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
xvimagesink: Make sure that the caps for upstream negotiation are simple caps
Fixes bug #631774.
2010-10-09 14:17:57 +0100 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
* tests/examples/app/appsrc-stream.c:
* tests/examples/app/appsrc-stream2.c:
examples: g_mapped_file_unref exists already since GLib 2.21.3
2010-10-07 19:32:56 +0200 Guillaume Emont <gemont@igalia.com>
* ext/ogg/gstoggdemux.c:
oggdemux: fix seeking with negative rate with skeleton
Files with a skeleton, or other files with a stream that ends before the end of
the chain would start playing from the end of the chain when trying to seek with
a negative rate at a position between the end of any stream and the end of the
chain.
This is due to the loop in _do_seek() assuming that pages will be encountered
for all streams shortly after the place where we want to seek, as found by
do_binary_search().
In the first iteration of the loop, stream ends are now checked against the
time of the current page.
2010-10-07 18:53:35 +0100 Zaheer Abbas Merali <zaheerabbas@merali.org>
* gst/tcp/gstmultifdsink.c:
multifdsink: gdp protocol is deprecated. People should use gdppay instead.
2010-10-08 12:43:28 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From c4a8adc to 5a668bf
2010-09-28 12:17:41 +0200 Edward Hervey <bilboed@bilboed.com>
* docs/libs/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/descriptions.c:
* gst-libs/gst/pbutils/gstdiscoverer-private.h:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/pbutils-private.h:
pbutils: rename gstdiscoverer-private.h to pbutils-private.h
2010-09-28 12:15:22 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
* gst-libs/gst/pbutils/gstdiscoverer-private.h:
* gst-libs/gst/pbutils/missing-plugins.c:
pbutils: Use copy_and_clean_caps for description methods
This allows the various _get_*_description() methods to be more
forgiving with the provided caps.
2010-10-08 12:51:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 5e3c9bf to c4a8adc
2010-10-08 11:23:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspextension.c:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
rtsp: make public _get_type() functions thread-safe
2010-10-08 10:29:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspurl.c:
rtspurl: minor clean-up
Merge and const-ify two arrays that should be one.
2010-10-08 10:06:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.c:
rtsp: fix enum value name in enums that are public API
https://bugzilla.gnome.org/show_bug.cgi?id=629746
2010-10-08 09:48:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
audio: make public get_type() functions thread-safe
2010-10-08 09:45:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
audio: fix enum value name in enums that are public API
So run-time bindings can introspect the names correctly (we abuse this
field as description field only in elements, not for public API
(where the description belongs into the gtk-doc chunk).
https://bugzilla.gnome.org/show_bug.cgi?id=629746
2010-10-08 12:30:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc-dist.c:
volume: Regenerate generated orc C code again with an orc fix for loading double parameters
2010-10-08 11:50:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
volume: Update generated orc sources
2010-10-08 11:49:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Fix unit test failure for the controlled processing functions
Going over integer arithmetic will lead to minimal rounding errors,
leading to +/-1 changes for volume==1.0. Implement the controlled
processing with floating point arithmetic, which was already done
for the C versions anyway.
2010-10-08 09:10:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Require orc 0.4.10
2010-10-07 23:54:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
audioconvert: Update generated orc files
2010-10-07 23:54:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc.orc:
volume: Update for orc changes
double parameters are declared with .doubleparam now.
2010-10-03 11:21:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
volume: Update generated orc sources
2010-10-03 12:00:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc.orc:
volume: Fix controlled processing via orc
2010-10-03 11:24:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: Actually enable usage of the orc optimized functions
2010-10-03 11:20:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Implement int32 processing with orc
2010-10-01 12:21:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Implement controlled processing for int16/1-2ch and int8/1,2,4ch with orc
2010-10-01 11:13:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Implement controlled processing for f64/1ch and f32/1-2ch in orc
2010-10-01 11:00:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Convert parts of the controlled processing to orc
2010-10-01 10:44:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Implement f64 scaling with orc
This requires orc 0.4.10
2010-10-01 10:38:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/gstaudioconvertorc.orc:
audioconvert: Implement remaining conversion functions from/to doubles to orc
This requires orc 0.4.10
2010-10-07 20:54:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: use g_object_notify_by_pspec() if possible
Use g_object_notify_by_pspec() when building against GLib >= 2.26.
This avoids the pspec lookup which takes the global paramspec pool lock.
2010-10-07 20:37:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: use g_object_notify_by_pspec() if possible
Use g_object_notify_by_pspec() when building against GLib >= 2.26.
This avoids the pspec lookup which takes the global paramspec pool lock.
2010-10-04 10:01:19 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Do not advance segment starts beyond stop times
Advance stop times too when they are getting higher than the
stop time of segments, avoiding assertions.
The stop time has to be advanced too so that running time keep in sync
for gapless mode.
https://bugzilla.gnome.org/show_bug.cgi?id=631312
2010-10-06 16:19:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/rtp.c:
tests: rtp: No need to unref buffer from bufferlist
Buffers obtained from buffer list iterators don't need to
be unreffed.
Test was failing due to this.
2010-10-04 11:22:45 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
vorbisdec: reverse playback; decode pending buffers upon EOS
2010-10-05 19:15:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
videoscale: use math-compat.h here as well
Hopefully the powers that be don't mind the gst/glib include here
too much.
2010-10-05 19:13:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: include new math-compat.h header for rint() on MSVC
Should fix compilation with Visual Studio 2008.
https://bugzilla.gnome.org/show_bug.cgi?id=630802
2010-10-05 17:19:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtp.def:
win32: update def file with new RTP methods
2010-10-05 17:13:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/libs/rtp.c:
check: fix rtp checks
Fix the checks for the extension support in RTP.
2010-10-05 16:36:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: fix position reporting
2010-08-26 12:34:11 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
* gst-libs/gst/rtp/gstrtcpbuffer.h:
rtcpbuffer: Add function to manipulation the data in RTCP feedback packets
Add methods to get/set the length of the Feedback Control Information (FCI) as
well as getting a pointer to the FCI itself.
2010-08-23 16:41:44 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* tests/check/libs/rtp.c:
tests: Test the manipulations of bufferlists containing RFC 5285 header extensions
2010-08-23 14:24:21 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
rtpbuffer: Add function to transform a GstBuffer into a GstBufferList
Add a new function called gst_rtp_buffer_list_from_buffer() that takes
a GstBuffer containing a RTP packets and spits out a GstBufferList
containing two buffers, one with the header and the other with the payload.
2010-08-22 19:44:38 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
rtpbuffer: Add functions to add RFC 5285 header extensions to GstBufferLists
Add functions to add header extensions to buffer lists, these functions only modify
the header part of the buffer lists, so the data is not copied.
2010-08-22 17:22:21 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
rtpbuffer: Add function to read RFC 5285 header extensions from GstBufferLists
2010-08-20 15:30:08 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* tests/check/libs/rtp.c:
tests: Add test for RTP header extension functions
2010-08-20 17:13:06 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
rtpbuffer: Add function to add RTP header extensions with a two bytes header
2010-08-20 12:54:38 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
rtpbuffer: Add function to append RFC 5285 one byte header extensions
2010-08-19 16:26:18 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
rtpbuffer: Add function to parse RFC 5285 header extensions
RFC 5285 describes a generic method to add multiple header extensions to RTP packets.
These functions parse these headers and return them, both for the one-byte header and the
two bytes headers.
2010-10-05 12:05:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
libvisual: only drop frames that are really too old
Also take the frame duration into account so that we don't drop frames that are
partially past the estimated QoS time.
2010-10-05 12:01:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
libvisual: add latency query
Add our own latency to the latency query reply from upstream.
2010-10-05 12:00:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
libvisual: add some defines
Add some defines for width/height/fps and a define for the minimum amount of
samples we need to buffer.
2010-10-04 15:48:51 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* tools/gst-discoverer.c:
gst-discoverer: The 'ready' signal was renamed to 'finished'
2010-10-04 17:27:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/gsttheoraparse.c:
parse: Don't error on discont
We don't need to error out when we detect a discontinuity.
2010-10-04 17:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/gsttheoraparse.c:
theoraparse: set caps on streamheader too
2010-10-04 13:07:14 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
build: Fix include path order for gir generation
This makes sure that the built girs are picked up over installed girs
where this is currently the case.
2010-10-01 14:52:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
codec utils: populate mpeg4 caps "level" field with level, not profile
Call the right function to get the level. Also add some more debug
logging.
2010-10-01 10:47:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
volume: Update generated orc files
2010-10-01 10:42:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
volume: Improve f32 scaling by using only a single array
Passing the same array as dest and src is invalid anyway because
they're maked with the restrict qualifier.
2010-09-30 15:19:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
pbutils: include config.h in codec utils
2010-09-30 00:19:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/codec-utils.c:
docs: add new codec utils API to docs
2010-05-01 01:03:18 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
* gst-libs/gst/pbutils/codec-utils.h:
* win32/common/libgstpbutils.def:
pbutils: Add MPEG-4 Video profile/level extraction
This adds code to translate the profile_and_level indication from the
MPEG-4 video (ISO/IEC 14496-2) headers to a string profile/level. The
mappings are taken from the spec and Wireshark's code, and might need to
be expanded on.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_mpeg4video_get_profile()
API: gst_codec_utils_mpeg4video_get_level()
API: gst_codec_utils_mpeg4video_caps_set_level_and_profile()
2010-04-30 20:50:09 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
* gst-libs/gst/pbutils/codec-utils.h:
pbutils: add H.264 profile/level extraction functions to codec utils
This adds code to parse the first few bytes of H.264 sequence parameter
set in order to extract the profile and level as const strings. This
code was originally in both qtdemux and matroskademux.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_h264_get_level()
API: gst_codec_utils_h264_get_profile()
API: gst_codec_utils_h264_caps_set_level_and_profile()
2010-04-30 15:12:04 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/codec-utils.c:
* gst-libs/gst/pbutils/codec-utils.h:
* gst/typefind/gsttypefindfunctions.c:
* win32/common/libgstpbutils.def:
pbutils: add AAC profile detection to codec utils
This moves AAC profile detection to pbutils, and uses this in
typefindfunctions. This will also be used in qtdemux.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_aac_get_profile()
API: codec_utils_aac_caps_set_level_and_profile()
2010-04-30 13:41:17 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/codec-utils.c:
* gst-libs/gst/pbutils/codec-utils.h:
* gst-libs/gst/pbutils/pbutils.h:
* gst/typefind/Makefile.am:
* gst/typefind/gstaacutil.c:
* gst/typefind/gstaacutil.h:
* gst/typefind/gsttypefindfunctions.c:
* win32/common/libgstpbutils.def:
pbutils: add codec-specific utility functions for AAC
This allows us to add generic codec-specific functionality, like
extracting profile/level data from headers, without having to duplicate
code across demuxers and typefindfunctions.
As a starting point, this moves over AAC level extraction code from
typefindfunctions, so it can be reused in qtdemux, etc.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_aac_get_sample_rate_from_index()
API: gst_codec_utils_aac_get_level()
2010-09-30 13:12:30 +0300 René Stadler <rene.stadler@nokia.com>
* gst-libs/gst/tag/tags.c:
tags: fix unused function warning with debug disabled
2010-09-30 12:59:46 +0300 René Stadler <rene.stadler@nokia.com>
* gst-libs/gst/tag/tags.c:
tags: fix illegal use of internal debug category function
From gstinfo.h:
/* do not use this function, use the GST_DEBUG_CATEGORY_INIT macro */
GstDebugCategory *_gst_debug_category_new (const gchar * name,
And more importantly:
#pragma GCC poison _gst_debug_category_new
So this commit fixes --disable-gst-debug builds.
2010-09-29 18:57:50 +0200 Edward Hervey <bilboed@bilboed.com>
* tools/gst-discoverer.c:
gst-discoverer: Print out topology if available.
If we have some unhandled streams, we can still print out the remaining
topology.
2010-09-29 18:54:28 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't post async-done when not needed
Where it was previously located, we would get async-done for the first
unknown-type, even if other valid streams would appear afterwards.
decode_bin_expose() will take care of posting async-done when the group
is exposed.
But we still want to post it in case the typefinding returned an unknown
type, in which case we will post it after posting an error.
These two changes ensure we do as much as possible before posting async-done.
2010-09-29 16:53:21 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
basertpdepay: ensure metadata is writable
2010-09-29 13:29:20 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add descriptions for more codecs
2010-09-29 12:33:44 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/examples/seek/seek.c:
seek: Fix debug statement argument type
2010-09-28 09:30:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/Makefile.am:
* tools/Makefile.am:
More makefile Fixes
Removing some not needed lines added in the last makefile
fixes commit (previous commit).
Also adds some more makefile files to check tests
2010-06-17 14:32:22 +0300 René Stadler <rene.stadler@nokia.com>
* sys/xvimage/xvimagesink.c:
xvimagesink: allow render rectangle coordinates to be negative
Useful for cropped zooming.
2010-06-17 14:33:44 +0300 René Stadler <rene.stadler@nokia.com>
* gst-libs/gst/interfaces/xoverlay.c:
xoverlay: allow render rectangle coordinates to be negative
This is useful for cropped zooming of the overlay.
2010-09-28 15:15:57 +0300 René Stadler <rene.stadler@nokia.com>
* gst-libs/gst/interfaces/xoverlay.c:
xoverlay: fix endless loop in deprecated method
2010-09-28 08:46:25 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/examples/app/Makefile.am:
* tools/Makefile.am:
Fixing Makefiles
Adds some missing lines to makefiles
2010-09-27 18:14:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/tags.c:
tags: add debug category for tags utility functions
2010-09-27 14:36:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/tags.c:
tags: try ISO-8859-1 as second fallback in case WINDOWS-1252 is not supported
Better safe than sorry. Some embedded systems may use crippled iconv
implementations or not support WINDOWS-1252 for other reasons.
https://bugzilla.gnome.org/show_bug.cgi?id=630471
2010-09-23 23:53:48 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* gst-libs/gst/tag/tags.c:
tags: when converting freeform strings try Windows-1252 as fallback instead of ISO-8859-1
Windows-1252 is a superset of ISO-8859-1, which uses some space
allocated to control characters for additional printable characters.
https://bugzilla.gnome.org/show_bug.cgi?id=630471
2010-09-24 21:30:20 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoraenc.c:
theoraenc: ptalarbvorm speed level goes to 3
2010-09-24 16:31:37 +0200 Vladimir <wl2776 at gmail.com>
* tests/examples/seek/seek.c:
seek: Add #define for seekbar granularity
Fixes #630496
2010-09-24 14:03:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/audio-enumtypes.c:
* win32/common/audio-enumtypes.h:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/gstrtsp-enumtypes.h:
* win32/common/interfaces-enumtypes.c:
* win32/common/interfaces-enumtypes.h:
* win32/common/pbutils-enumtypes.c:
* win32/common/pbutils-enumtypes.h:
* win32/common/video-enumtypes.c:
* win32/common/video-enumtypes.h:
win32: define GST_PACKAGE_RELEASE_DATETIME in win32 config.h as well
Also update enums.
2010-09-24 00:25:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/.gitignore:
.gitignore: ignore new appsrc unit test
2010-09-24 13:09:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: add Since markers
Fixes #630443
2010-07-30 13:54:42 +0200 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* win32/common/libgstaudio.def:
baseaudiosink: Added getter and setter for drift tolerance.
2010-09-24 12:54:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: subtract the render_delay from our latency
The latency reported by the base class includes the render_delay, which we don't
want to include when we start slaving our clocks.
See #630441
2010-09-23 23:57:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: Use G_DEFINE_ABSTRACT_TYPE instead of manual GObject boilerplate code
This also makes the _get_type() function threadsafe.
Fixes bug #630440.
2010-09-23 10:16:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tags: exif: Add mapping for _HORIZONTAL_ERROR
Maps GST_TAG_GEO_LOCATION_HORIZONTAL_ERROR to the
GPSHPositionError tag in exif.
Tests included.
2010-09-22 14:10:18 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
* tests/check/Makefile.am:
* tests/check/elements/appsrc.c:
appsrc: Do not override buffer caps if appsrc caps is null
Make appsrc not set caps on buffers when its own caps is NULL.
This avoids calling make_metadata_writable on all buffers and
prevents losing buffer caps in case we are not replacing it
with something meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=630353
2010-09-21 18:57:42 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraenc.h:
theoraenc: Make the bitrate/quality dynamically modifiable
https://bugzilla.gnome.org/show_bug.cgi?id=630303
2010-09-22 12:35:59 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
* tools/gst-discoverer.c:
discoverer: Fixup DiscovererResult handling
This was a leftover from the changes from a flag to an enum
2010-09-22 12:10:24 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: We don't need the signals from the queues
2010-09-22 01:50:21 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/Makefile.am:
gst-libs: build pbutils after video
Because pbutils now depends on video.
2010-09-21 18:33:36 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From aa0d1d0 to 5e3c9bf
2010-09-20 21:04:48 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
discoverer: fix docs
While the doc parser allows for certain variation, it is a good idea to not
use random characters here and there, but try to stick to the little markup
syntax there is.
2010-09-20 16:45:32 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Fix debug statement.
Fixes build on macosx
2010-09-20 16:28:52 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/volume/gstvolumeorc-dist.c:
volume: orc fixup for loading float arguments
This is only used with DISABLE_ORC.
2010-09-20 11:24:10 +0200 Edward Hervey <bilboed@bilboed.com>
* tools/.gitignore:
* tools/Makefile.am:
* tools/gst-discoverer.c:
tools: Standalone tool for discovering media file properties
Fixes #625944
2010-09-20 11:23:36 +0200 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstpbutils.def:
win32: Update with symbols from GstDiscoverer
Fixes #625944
2010-09-20 11:23:17 +0200 Edward Hervey <bilboed@bilboed.com>
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
docs: Documentation for new pbutils GstDiscoverer
Fixes #625944
2010-09-20 11:22:32 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/Makefile.am:
* gst-libs/gst/pbutils/.gitignore:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstdiscoverer-private.h:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* gst-libs/gst/pbutils/pbutils-marshal.list:
* gst-libs/gst/pbutils/pbutils.h:
pbutils: New Discoverer utility
Fixes #625944
2010-09-20 11:13:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add mp3 to the apetag extensions
2010-09-18 13:15:08 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix regression in ball pattern
Was painting using two different methods.
2010-09-17 11:46:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: Fix ACCEPTABLE_CHAR() macro to work with signed and unsigned chars
2010-09-17 11:44:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/sdp/gstsdpmessage.c:
Revert "sdp: Remove useless check in macro"
This reverts commit e6a041b69fd21c42651d98cf8a3064e43cecc51c.
It's not a useless check, the signedness of "char" and "gchar" is
defined by the ABI.
2010-09-17 10:43:04 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: Remove useless check in macro
A signed char is always < 128. Fixes a warning on macosx build.
2010-09-16 18:03:23 -0700 David Schleef <ds@schleef.org>
* gst/adder/gstadderorc-dist.c:
* gst/adder/gstadderorc-dist.h:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.h:
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
orc: update generated files to fix MSVC compile issues
2010-09-16 17:56:31 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/gstvideoscaleorc.orc:
videoscale: Don't use broken orc feature
2010-09-16 19:30:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/interfaces/xoverlay.c:
xoverlay: G_GUINTPTR_FORMAT is since 2.22
Don't rely on too new symbols, we only depend on 2.20.
2010-09-16 15:01:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/playrec/.gitignore:
* tests/examples/playrec/Makefile.am:
* tests/examples/playrec/playrec.c:
examples: add synchronized playback and capture example
Add an example that demonstrates synchronized playback and capture.
2010-09-16 17:15:32 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.h:
videotestsrc: Fix indentation
2010-09-16 17:14:20 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: add bar pattern
Simple bar with foreground color on the background color
2010-09-16 15:07:15 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/videotestsrc.c:
tests: use gst-check API in videotestsrc
use gst_check_drop_buffers in videotestsrc to
clear the global buffers list.
2010-09-16 14:55:55 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/videotestsrc.c:
tests: Fix unit test of videotestsrc
Use UYVY for unit tests, it's exactly the same as Y422.
(which is currently disabled in videotestsrc)
2010-09-15 15:13:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update for new strings
2010-09-15 15:12:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.h:
docs: add Since: comment to docs for new GST_VIDEO_FORMAT_UYVP
2010-09-14 11:20:42 -0400 Tristan Matthews <le.businessman@gmail.com>
* ext/gnomevfs/gstgnomevfssrc.c:
gnomevfsrc: set GST_PARAM_MUTABLE_READY flag on the "handle" property
Fixes #629672
2010-09-15 15:19:04 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: fix segfault on negative horizontal-speed
2010-09-15 14:15:13 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Actually dispose the unused ghostpads
2010-09-15 11:28:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/speex_resampler.h:
* gst/audioresample/speex_resampler_wrapper.h:
Revert "audioresample: Add GAP flag support"
This reverts commit 129af0d8e6a74e8edef3e77c3626616b674b7cc1.
This shouldn't be committed at all, it isn't ready and apparently
was in the wrong branch locally.
2010-09-15 11:26:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/convertframe.c:
* gst-libs/gst/video/video.h:
* tests/check/libs/video.c:
video: Add a destroy notify parameter to gst_video_convert_frame_async()
Binding generators apparently need this as they can't really know
that the callback is guaranteed to be called exactly once and that
the user_data can be freed at the end of it.
2010-09-14 12:00:39 +0200 Leo Singer <lsinger@caltech.edu>
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/speex_resampler.h:
* gst/audioresample/speex_resampler_wrapper.h:
audioresample: Add GAP flag support
Fixes bug #586570.
2010-09-05 15:17:47 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
* sys/v4l/gstv4lxoverlay.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.cpp:
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
* tests/icles/stress-xoverlay.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
* win32/common/libgstinterfaces.def:
xoverlay: Add guintptr versions of functions
And deprecate the gulong versions. This is to support platforms
where sizeof(unsigned long) < sizeof(void *). Fixes #627565.
API: Add gst_x_overlay_set_window_handle()
API: Deprecate: gst_x_overlay_set_xwindow_id()
API: Add gst_x_overlay_got_window_handle()
API: Deprecate: gst_x_overlay_got_xwindow_id()
API: Add GstXOverlay::set_window_handle()
API: Deprecate: GstXOverlay::set_xwindow_id()
2010-09-14 12:31:58 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add UYVP
2010-09-12 20:36:19 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: Various improvements
Replace moving-color-bars pattern with smpte100, and change
moving-speed to horizontal-speed. Default is now 0. Add
a rotation stage to pattern building.
Allocate a temporary scanline for building images. Remove
unused code. Disable several patterns that we're unable to
test and probably never used. Add other variants of bayer
sampling. Convert some patterns to use videotestsrc_blend_line.
2010-09-10 18:10:40 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: clean up blink pattern
2010-09-10 15:57:54 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Clean up the RGB code
2010-09-10 14:40:44 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: Convert to intermediate AYUV/ARGB
Scanlines are generated into AYUV/ARGB, then converted to the
various formats.
2010-09-10 12:48:30 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: rearrange code to work on scanlines
2010-09-10 12:03:07 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix recent breakage of smpte75 pattern
2010-09-01 15:18:31 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: add moving color bars pattern
This pattern is moving the color bars with a given
speed. Negative speed is inverting the moving direction.
https://bugzilla.gnome.org/show_bug.cgi?id=628500
2010-06-14 15:42:09 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videoscale/vs_image.c:
* gst/videoscale/vs_scanline.c:
videoscale: refactor using more Orc code
Convert downsampling to Orc. Convert horizontal linear scaling
to Orc. Combine horizontal and vertical scaling into one pass.
2010-09-12 19:34:28 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add UYVP, 10-bit 4:2:2
2010-09-14 08:41:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
video: Add gst_video_convert_frame_async() to the docs
2010-09-14 08:40:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstvideo.def:
win32: Add gst_video_convert_frame() and gst_video_convert_frame_async() to the .def files
2010-09-14 08:40:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/video.c:
video: Add unit test for gst_video_convert_frame_async()
2010-09-14 08:39:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/convertframe.c:
* gst-libs/gst/video/video.h:
video: Add async variant of the convert frame function
API: gst_video_convert_frame_async()
2010-09-12 16:53:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/video.c:
video: Add a simple unit test for the new convert_frame API
Unfortunately this can't test the encoding because there's no
image encoder in base.
2010-09-12 16:51:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/convertframe.c:
video: Strip framerate from the target caps
There will always be only a single output buffer and if the
target caps have a different framerate than the input there
will be a negotiation error during conversion.
2010-09-12 16:36:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/convertframe.c:
video: Refactor convert_frame a bit and fix some minor memory leaks in error cases
2010-09-09 14:11:52 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/Makefile.am:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
* gst/playback/gstscreenshot.c:
* gst/playback/gstscreenshot.h:
playback: Switch to using gst_video_convert_frame
https://bugzilla.gnome.org/show_bug.cgi?id=629157
2010-09-09 13:44:54 +0200 Edward Hervey <bilboed@bilboed.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/convertframe.c:
* gst-libs/gst/video/video.h:
video: Add new method for converting a video frame
https://bugzilla.gnome.org/show_bug.cgi?id=629157
2010-09-13 10:02:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: prevent another race with shutdown state change
2010-09-11 14:55:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstsdp.def:
win32: Add new SDP symbols to the .def files
2010-09-10 18:42:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: remove leftover g_print
2010-09-10 17:55:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst-libs/gst/sdp/gstsdpmessage.h:
sdp: add methods to convert between uri and message
Add methods to convert between uri and sdpmessages, loosly based on
http://tools.ietf.org/html/draft-fujikawa-sdp-url-01
API: GstSDPMessage::gst_sdp_message_parse_uri
API: GstSDPMessage::gst_sdp_message_as_uri
2010-09-10 10:40:52 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/videotestsrc.c:
tests: videotestsrc change the pattern property for the tests
2010-09-10 08:42:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/adder/gstadderorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/volume/gstvolumeorc-dist.c:
orc: Fix generated source files
2010-09-09 20:45:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
tests: fix passing of URIs containing '*' and '?' to the seek example
Only do wildcard expansion (why?!) on things that look like local
file paths. Fixes passing of URIs containing '*' and '?' (see #629212).
2010-09-09 21:51:18 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/check/Makefile.am:
* tests/check/generic/states.c:
tests: allow running state tests for all elements
Now one can use GST_NO_STATE_IGNORE_ELEMENTS=1 make generic/states.check
to try elements that would normaly be skipped.
2010-09-09 11:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/adder/gstadder.c:
adder: Do debug category initialization in plugin_init again
2010-09-09 10:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/adder/gstadderorc-dist.c:
* gst/adder/gstadderorc-dist.h:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.h:
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
orc: Update generated source files everywhere
2010-09-09 10:57:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/adder/gstadder.c:
* gst/adder/gstadderorc.orc:
* gst/audioconvert/gstaudioconvertorc.orc:
* gst/audioconvert/plugin.c:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrcorc.orc:
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
Revert "Revert "Use init functions for Orc code""
This reverts commit 93aa13639d74449dc68296427e5dbcfe8aca5f51.
Everything should work now after regenerating the disted source files.
2010-09-07 19:04:23 +0200 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstaudio.def:
win32: Add new symbol to libgstaudio
2010-09-07 18:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudio: avoid taking extra ref on sink/src
Don't take an extra ref on the sink and source because that creates a reference
cycle. Instead, use the invalidate method of the clock when the sink and source
are freed. This way, we don't call into the time function anymore after the
objects are disposed.
2010-09-07 18:06:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
audioclock: add a function to invalidate the clock
Add a function to invalidate the time function of a clock. Useful for when the
function becomes invalid.
2010-09-07 16:26:56 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/Makefile.am:
check: Fix linking order of libs/tag
2010-09-07 16:26:30 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/gst-plugins-base.supp:
check: Make fontconfig leak suppression more generic
2010-09-07 08:46:15 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Adds mappings for new image ppi tags
Adds mappings for GST_TAG_IMAGE_HORIZONTAL/VERTICAL_PPI into
our exif lib
Tests included.
Fixes #626570
2010-09-07 08:22:27 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
tags: Add GST_TAG_IMAGE_HORIZONTAL/VERTICAL_PPI tags
Adds new tags for representing the intended PPI of images/videos
API: GST_TAG_IMAGE_HORIZONTAL_PPI
API: GST_TAG_IMAGE_VERTICAL_PPI
Fixes #626570
2010-09-07 11:41:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From c2e10bf to aa0d1d0
2010-09-06 18:17:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
rtp: improve basertpdepayload's error message when no input caps were set
This is pretty much an FAQ, so try to make the error message a bit
more helpful. Also, don't tell people to file a bug in bugzilla
about this (which is what happens if the default error message for
CORE_NEGOTIATION is used).
2010-09-06 13:14:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
rtppayload: notify of first timestamp/seqnum
Notify of the first timestamp/seqnum pushed out by the payloader.
Fixes #612264
2010-09-06 11:53:35 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/videotestsrc/.gitignore:
videotestsrc: .gitignore new generate_sine_table
2010-09-06 11:44:17 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstinputselector.c:
* gst/playback/gstinputselector.h:
* gst/playback/gstplaybin2.c:
playback: ref the selector pad class inside input-selector
Minimizes the delta to original element in -bad and allows us to keep the
type static.
2010-09-05 20:57:48 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/Makefile.am:
* gst/videotestsrc/generate_sine_table.c:
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Use static sine table
2010-09-05 20:35:13 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: Add foreground/background-color properties
Replace solid-color property with foreground-color and add
background-color. Pull some common code out of each of the
pattern generating functions. Fix many of the patterns to
use foreground-color/background-color instead of white/black.
Generated images are indentical to previously if foreground-color
and background-color are left as default.
API: GstVideoTestSrc::foreground-color
API: GstVideoTestSrc::background-color
2010-09-05 18:58:03 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From d3d9acf to c2e10bf
2010-09-05 17:04:31 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: deprecate colorspec property
Fixes: #616392.
2010-09-05 12:57:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc.orc:
audioconvert: Simplify float->s32 conversion
orc 0.4.7 is doing saturated conversion from floats to integers
and it's not necessary to do this manually anymore.
2010-09-05 12:14:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From ca1c867 to d3d9acf
2010-09-05 12:12:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
audioconvert: Update disted orc files
2010-09-05 12:09:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: Enable float processing with orc again
2010-09-05 12:08:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Require orc 0.4.8.1 for the volume test
2010-08-26 19:16:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/gstaudioconvertorc.orc:
audioconvert: Use the ORC double support
2010-09-04 09:06:08 +0200 Leo Singer <lsinger@caltech.edu>
* gst-libs/gst/tag/gstexiftag.c:
exiftag: Fix compiler warnings with old gcc versions
Old gcc complains about possibly uninitialized variables which
are always initialized before usage in reality. Fixes bug #628747.
2010-08-06 11:53:38 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstfactorylists.c:
* gst/playback/gstfactorylists.h:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playback: Switch to gstfactorylist from core
https://bugzilla.gnome.org/show_bug.cgi?id=626181
2010-09-02 12:57:42 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: fix typo in property description
2010-09-01 17:52:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Add support for AYUV
2010-09-01 11:37:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: Fill segment until the end on EOS
2010-09-01 11:33:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Fill the segment on EOS or at least produce enough frames to use the complete buffer duration
Fixes bug #628400.
2010-09-01 11:22:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Don't ignore flow returns when filling the previous segment during NEWSEGMENT handling
2010-09-01 11:11:47 +0200 Philippe Normand <pnormand@igalia.com>
* tests/examples/seek/seek.c:
seek: allow seeking behind the currently downloaded position.
2010-09-01 10:06:09 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: use GST_BOILERPALTE macro
2010-08-31 10:09:51 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaysink.c:
playback: Set queues silent property to TRUE
We don't use the queue signals within playsink.
2010-08-30 14:59:22 -0500 Rob Clark <rob@ti.com>
* ext/pango/gsttextoverlay.c:
textoverlay: fix Cb/Cr inversion for colored text overlays
In case of odd values for xpos or ypos, the division by two in CbCr
plane would result in an off-by-one error, which in the case of NV12,
NV21, or UYVY would cause inversion of blue and red colors. (And
would be not so easily noticed for I420 as it would just cause the
chroma to be offset slightly from the luma.)
This patch also fixes a silly typo from the earlier patch which
added NV12 support that broke UYVY support.
2010-08-30 15:50:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Don't reset the pad when pushing resulted in NOT_LINKED
The pad might be linked later and after resetting it it will
only work after resetting all of oggdemux.
2010-08-27 20:45:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Only enable progressive downloading if the upstream duration in bytes is known
Otherwise we might try to enable it for live streams, where this would
cause playback to fail completely.
Fixes bug #628028.
2010-08-27 17:23:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
oggdemux: Don't use GST_FLOW_IS_FATAL()
And while we're at it, handle WRONG_STATE as error too
in oggdemux and WRONG_STATE and NOT_LINKED in oggaviparse.
2010-08-27 11:49:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
* gst/adder/gstadderorc.orc:
* gst/audioconvert/gstaudioconvertorc.orc:
* gst/audioconvert/plugin.c:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrcorc.orc:
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
Revert "Use init functions for Orc code"
This reverts commit b2051090b43f82b23bb01826f09053479bbd7874.
Fixes the build again until someone pushes the regenerated .c/.h
files too.
2010-08-22 23:01:19 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: clean up code
Merge various color structures into one.
2010-08-22 22:16:45 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: Add ball pattern
2010-08-19 15:43:09 -0700 David Schleef <ds@schleef.org>
* gst/adder/gstadder.c:
* gst/adder/gstadderorc.orc:
* gst/audioconvert/gstaudioconvertorc.orc:
* gst/audioconvert/plugin.c:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrcorc.orc:
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc.orc:
Use init functions for Orc code
2010-08-26 15:17:20 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/volume/gstvolume.c:
volume: make the orc usage for float conditional again
See bug #628009. The tests still fail in the orc code (which we just don't call
now).
2010-08-25 12:19:05 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst-libs/gst/riff/riff-media.c:
riff: add support for 2vuy
It is the apple alternative for Microsofts UYVY.
(http://ntta.szm.com/Tutors/FourCC.htm)
Only use the UYVY for the caps to enable support in other
gstreamer elements.
https://bugzilla.gnome.org/show_bug.cgi?id=627924
2010-08-25 19:01:57 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
* gst/volume/gstvolumeorc.orc:
volume: enable ORC for float in volume
2010-08-25 11:19:31 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* configure.ac:
* gst-libs/gst/tag/gstexiftag.c:
configure: Add check for log2
Adds check for log2 and only use it in exif library if it is
available.
2010-08-25 15:32:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
tag: Link to $(LIBM) for pow(), log2() and friends
2010-08-25 08:41:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Move some tags to their correct IFDs
Put some tags in their correct IFDs
2010-08-20 16:39:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Always write FlashPixVersion tag
FlashPixVersion is mandatory and constant. Write it always.
2010-08-20 15:59:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Adds flash tags mapping
Adds a mapping for GST_TAG_CAPTURING_FLASH_FIRED/_MODE to
the exif Flash tag.
Tests included.
2010-08-19 15:47:18 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gsttageditingprivate.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
* win32/common/libgsttag.def:
tag: exif: More photography mappings
Adds mappings from:
GST_TAG_CAPTURING_EXPOSURE_PROGRAM -> ExposureProgram
GST_TAG_CAPTURING_EXPOSURE_MODE -> ExposureMode
GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE -> SceneCaptureType
GST_TAG_CAPTURING_GAIN_ADJUSTMENT -> GainControl
GST_TAG_CAPTURING_WHITE_BALANCE -> WhiteBalance
GST_TAG_CAPTURING_CONTRAST -> Constrast
GST_TAG_CAPTURING_SATURATION -> Saturation
Also renames gst_tag_image_orientation_from_exif_value and
gst_tag_image_orientation_to_exif_value to remove the 'gst'
prefix and not including in the win32 defs.
Tests included.
2010-08-19 09:39:39 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Add macros for the exif ids
Use macros for exif ids to avoid having those numbers spread
all over the code.
2010-08-17 15:56:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Adds photography tags mappings
Adds the following mappings for the exif helper:
* GST_TAG_CAPTURING_DIGITAL_ZOOM_RATIO -> DigitalZoomRatio
* GST_TAG_CAPTURING_FOCAL_LENGTH -> FocalLength
* GST_TAG_CAPTURING_SHUTTER_SPEED -> ExposureTime, ShutterSpeedValue
* GST_TAG_CAPTURING_FOCAL_RATIO -> FNumber, ApertureValue
* GST_TAG_CAPTURING_ISO_SPEED -> ISOSpeed, PhotographicSensitivity
Tests included.
2010-08-17 15:05:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Adds mapping for GST_TAG_APPLICATION_DATA
Adds mapping for GST_TAG_APPLICATION_DATA to the exif
'maker-note' tag.
2010-08-20 14:54:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
tag: Adds GST_TAG_CAPTURE_FLASH_FIRED/_MODE
Adds a new tag for informing if flash was used while
capturing an image and the flash mode selected by the
user during this capture
API: GST_TAG_CAPTURING_FLASH_FIRED
API: GST_TAG_CAPTURING_FLASH_MODE
https://bugzilla.gnome.org/show_bug.cgi?id=626651
2010-08-17 07:21:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
tag: More photography related tags
API: GST_TAG_CAPTURING_EXPOSURE_PROGRAM
API: GST_TAG_CAPTURING_EXPOSURE_MODE
API: GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE
API: GST_TAG_CAPTURING_GAIN_ADJUSTMENT
API: GST_TAG_CAPTURING_WHITE_BALANCE
API: GST_TAG_CAPTURING_CONTRAST
API: GST_TAG_CAPTURING_SATURATION
Fixes #626651
2010-08-17 06:47:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
tag: Adds some basic photography tags
Adds the following basic photography tags.
API: GST_TAG_CAPTURING_SHUTTER_SPEED
API: GST_TAG_CAPTURING_FOCAL_RATIO
API: GST_TAG_CAPTURING_FOCAL_LENGTH
API: GST_TAG_CAPTURING_DIGITAL_ZOOM_RATIO
API: GST_TAG_CAPTURING_ISO_SPEED
Fixes #626651
2010-08-24 15:06:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
audioconvert: Require ORC 0.4.7 for the loadl/storel opcodes
And update disted files to allow compilation with no or too old ORC.
2010-08-24 11:39:09 +0200 Alessandro Decina <alessandro.decina@collabora.co.uk>
* gst/adder/gstadder.c:
adder: Make sure FLUSH_STOP is always sent after a flushing seek.
Send FLUSH_STOP right after forwarding the seek event upstream if necessary.
This makes sure that adder->srcpad is not left flushing if seeking fails or if
upstream is blocked.
The same fix was already applied to videomixer in 49b2a946.
2010-08-24 11:11:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/gstaudioconvertorc.orc:
audioconvert: Use ORC for the float<->int32 conversion
This should speed up standard Vorbis encoding and decoding pipelines a bit.
Thanks to David Schleef for the assistance to get the ORC code right
and explaining everything.
2010-08-24 10:12:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Support NV21 too and minor cleanups
2010-08-24 10:03:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Fix component width for NV12/NV21
Both have width/2 as component width for the chroma planes.
2010-08-24 09:51:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix NV21 rendering
Using the same as for NV12 will result in wrong colors and crashes.
2010-08-23 18:51:18 -0400 Chris Shoemaker <chris.shoemaker@cox.net>
* ext/pango/gstclockoverlay.c:
* ext/pango/gstclockoverlay.h:
clockoverlay: only rerender text if time string has changed
The textoverlay element will rerender the text string whenever
overlay sets the 'need_render' flag to TRUE. Previously, we
lazily set the flag to TRUE every time the time string was requested.
Now, we save a copy of the previously given string, and only set
'need_render' to TRUE if the string has changed.
In my tests with a 30fps video stream, and a time string including
a seconds field, this change reduced the CPU usage of the clockoverlay
element from 60% to 5%.
Fixes bug #627780.
2010-08-23 13:59:38 -0500 Rob Clark <rob@ti.com>
* ext/pango/gsttextoverlay.c:
textoverlay: add NV12 support
Fixes bug #627768.
2010-08-20 12:03:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Mark duplicated frames with the GAP flag
We currently don't use the GAP flag for video and the docs say
that this is for buffers, that have been created to fill a gap
and contains neutral data. For video this is the previous frame.
This information can be used by encoders to encode the duplicated
frames more efficiently. See bug #627459.
2010-08-19 18:51:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Add back support for 8 bit paletted RGB
This was removed by 3a00a97fd2b4015e93cdcabaa75da406aa599570
while making the pad template caps more compact.
Fixes bug #626629.
2010-08-18 16:45:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
alsasrc/sink: add property to get the card name
fixes #627203
2010-08-18 16:42:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsa.h:
alsa: add method to retrieve the card name
Reuse an existing method to retrieve the card name.
2010-08-18 12:34:07 +0200 American Dynamics <GStreamer-Bugs at tycosp.com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
basertpdepay: don't clear the discont flag too early
Set the discont flag when we receive a DISCONT buffer and only clear the discont
state when we pushed out a DISCONT buffer.
Fixes #626869
2010-08-14 19:08:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
docs: fix typo in appsink docs so function gets cross-referenced properly
2010-08-14 19:02:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
* configure.ac:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
build: use new AG_GST_PKG_CONFIG_PATH m4 macro from common
Sets up a GST_PKG_CONFIG_PATH variable for use in Makefile.am
(avoids trailing ':' in PKG_CONFIG_PATH used).
2010-08-14 18:36:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
introspection: set PKG_CONFIG_PATH so that our in-tree libs come first when calling scanner
When calling gobject-introspection scanner, make sure our own
freshly-built libs within the source tree (well, build dir) come
first in the PKG_CONFIG_PATH. May or may not help to make sure
that it doesn't pick up older external plugins-base libs (or
.gir files) from outside the source tree / build directory as
dependencies of the introspected lib instead of using the
stuff we just built in a sibling directory.
https://bugzilla.gnome.org/show_bug.cgi?id=623698
2010-08-06 17:16:27 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playback: Delay usage of GstFactoryList
By delaying it to when it's actually needed, we speed things up a bit
since some elements might have been added/removed in between.
https://bugzilla.gnome.org/show_bug.cgi?id=626718
2010-06-17 09:10:11 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playbin2: uridecodebin: add property to configure ring buffer size
2010-08-13 17:23:46 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 3e8db1d to ec60217
2010-08-13 13:59:08 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/gst-plugins-base-plugins-sections.txt:
plugin-docs: the <TITLE> tag should come right after <FILE>.
Fixes missing plugin entries. If the object name, e.g. GstGIOSrc came before the
title, we ended up with differnt section_id in the generated docbook.
2010-08-12 18:14:38 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/app/gstappsrc.c:
appsrc: fix the classification.
Change "Src" into "Source" (we use that elsewhere). I did not keept "Src" as it
is quite unlikely that someone plugs appsrc by searching the registry by classification.
2010-08-12 15:26:08 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
alsa: remove 'dir' out variable
Alsa seems to expect that we initialize it. Remove the variable and pass NULL
as we actually don't use it. In alsasink also #ifdef one section that is
grabing diagnostics to be disabled, when logging is disabled (the code was
using the out parameter as well).
Fixes #626125
2010-08-12 11:46:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: remove chroma-site and color-matrix fields from RGB caps
2010-08-11 12:49:40 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: prevent deadlock with _chain when deactivating pad
Fixes #626581.
2010-08-12 12:50:27 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/Makefile.am:
playback: bad bad editor, readd missing line to fix the build
2010-08-12 12:08:35 +0300 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/playback/.gitignore:
* tests/examples/playback/Makefile.am:
* tests/examples/playback/decodetest.c:
* tests/examples/playback/test.c:
* tests/examples/playback/test2.c:
* tests/examples/playback/test3.c:
* tests/examples/playback/test4.c:
* tests/examples/playback/test5.c:
* tests/examples/playback/test6.c:
* tests/examples/playback/test7.c:
* tests/icles/Makefile.am:
* tests/icles/playback/.gitignore:
* tests/icles/playback/Makefile.am:
* tests/icles/playback/decodetest.c:
* tests/icles/playback/test.c:
* tests/icles/playback/test2.c:
* tests/icles/playback/test3.c:
* tests/icles/playback/test4.c:
* tests/icles/playback/test5.c:
* tests/icles/playback/test6.c:
* tests/icles/playback/test7.c:
tests/playback: due to popular demand mv them from examples to icles
The tests are toys and not reference demos.
2010-08-12 10:02:56 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: send preroll buffer when delaying preroll eos
That is, if eos is received which will not be forwarded, and the stream
has not yet seen any data, then send a buffer to preroll downstream
(which might otherwise be accomplished by the eos event).
2010-08-12 10:01:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: remove some heuristic in chain configuration code
.. since queues are now inserted unconditionally.
2010-08-11 10:27:35 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
playbin2/playsink: update subtitle handling for streamsynchronizer
Streamsynchronizer excepts to see stream-changed msg for all streams, but to
arrange for this, video and subtitle streams need to be decoupled by means
of queues (due to pad blocks that may occur).
Fixes #626463.
2010-08-10 13:06:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: always have a queue in chain head to aid streamsynchronizer
Specifically, as the latter may have one thread pushing EOS to several streams,
that needs to be decoupled into various thread to prevent preroll hanging
problems.
2010-08-10 11:28:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: drop lock when pushing eos downstream
... to prevent deadlock (e.g. upon seek) when downstream waits in preroll.
2010-08-10 11:19:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: clear stream eos state on FLUSH and new stream
2010-08-10 11:19:22 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: set READY sinks to NULL before freeing chain upon failure
2010-08-12 10:49:59 +0300 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* gst/playback/.gitignore:
* gst/playback/Makefile.am:
* gst/playback/decodetest.c:
* gst/playback/test.c:
* gst/playback/test2.c:
* gst/playback/test3.c:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
* tests/examples/Makefile.am:
* tests/examples/playback/.gitignore:
* tests/examples/playback/Makefile.am:
* tests/examples/playback/decodetest.c:
* tests/examples/playback/test.c:
* tests/examples/playback/test2.c:
* tests/examples/playback/test3.c:
* tests/examples/playback/test4.c:
* tests/examples/playback/test5.c:
* tests/examples/playback/test6.c:
* tests/examples/playback/test7.c:
playback: move tests from plugin-dir to tests/examples/playback
2010-08-11 18:08:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
xvimagesink: Suggest caps with different width/height if bufferalloc is called with impossible width/height
2010-08-11 17:16:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/videoscale.c:
videoscale: Add some debug output to the videoscale negotiation test
2010-08-11 17:03:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Only set the PAR if the caps already had a PAR
Otherwise we're producing different caps and basetransform thinks that it
can't passthrough buffer allocations, etc.
In 0.11 all video caps really should have the PAR set...
2010-08-11 17:00:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
xvimagesink: It's not a bad thing if the preferred video format needs less bytes per frame
2010-08-11 08:47:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tags: exif: Fix bug in inner ifd parsing
Do not use the result of inner ifd's parsing to increment
the current tag index. The reasons are:
1) The function returns a boolean.
2) The inner ifd's tags are in a separate table, so they shouldn't
interfere with its parent ifd table parsing.
2010-08-11 08:03:44 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Put ExifVersion in the correct IFD
ExifVersion is from the 'exif' ifd, not the 0th ifd.
2010-08-10 19:50:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Refactor functions declaration
Use some macros to declare serialization/deserialization
functions prototypes.
2010-08-10 19:30:11 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Add another DateTime mapping
datetimes can also be represented by the 0x132 tag. Map it, too.
2010-08-10 11:29:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Fix bug on image-orientation parsing
Do not skip one extra tag when parsing image-orientation tags.
2010-08-10 10:57:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From bd2054b to 3e8db1d
2010-08-10 11:52:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
exiftag: Compare with G_MAXUINT16 instead of -1
Fixes a compiler warning on the OS X buildbot.
2010-08-09 18:04:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
xmp: exif: Adds GST_TAG_APPLICATION_NAME mappings
adds xmp and exif helper library mappings for GST_TAG_APPLICATION_NAME
tag.
2010-08-04 13:01:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Write ExifVersion tag
Write ExifVersion tag unconditionally when creating exif
buffers. Might help other applications parsing of this data.
2010-08-04 13:02:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/tag.c:
tests: tag: Test to try to serialize multiple exif tags
Adds a new test for exif data that tries serializing data
from multiple ifd tables and check if it works.
2010-08-09 17:25:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tags: exif: Fix inner tags offset rewriting
Fixes a bug that made exif helper lib fail to rewrite inner ifd tags
offsets when there were more than 1 inner ifd.
2010-07-22 17:29:57 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: configurable text color and position
Rather than only left, right, top, etc, allow for horizontal and vertical
positioning on a scale from 0 to 1.
Also cater for configuring rendered text color.
Fixes #624920.
API: GstTextOverlay:xpos
API: GstTextOverlay:ypos
API: GstTextOverlay:color
2010-07-21 14:20:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: add solid-color pattern
... which generalizes the current listing of white, black, etc.
In particular, also allow specifying alpha channel, and modify
some structures and pattern filling to cater for alpha value as well.
Fixes #624919.
API: GstVideoTestSrc:solid-color
2010-08-08 17:42:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: static forward declarations are forbidden by the C standard
...and actually cause compiler errors on VC++. Change it to an extern
forward declaration and non-static definition.
2010-08-05 13:56:29 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 2004d03 to bd2054b
2010-08-04 19:24:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Check if the compiler supports ISO C89 or C99 and which parameters are required
This first checks what is required for ISO C99 support and sets the relevant
compiler parameters and if no C99 compiler is found, it checks for a
C89 compiler. This enables us to check for and use C89/C99 functions
that gcc hides from us without the correct compiler parameters.
2010-08-04 15:18:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosink/baseaudiosrc: Post CLOCK-LOST/CLOCK-PROVIDE when going to/from READY
Otherwise the clocks are redistributed every time the pipeline
goes to PAUSED, which is quite expensive.
2010-08-03 15:03:27 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Make xmp lib aware for the different tag types
Makes the xmp helper lib aware that the tags can be simple,
sequences or bags (there is still struct and alt, but those
aren't handled yet). Adding this info makes serialization
and deserialization more consistent.
2010-08-02 09:56:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
xmp: Add a new layer of indirection (GstXmpSchema)
Instead of storing all tags in a single hashtable, store them
grouped by schema in a GstXmpSchema, and add those to the toplevel
hashtable.
2010-08-03 14:37:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Make bag tags deserialization work correctly
If we find a bag of tags of type string in the xmp packet, we
should concat them, this is not the ideal approach, but at
least works for now as we don't know what type of tag it
is (simple, structure, seq, alt or bag)
2010-08-04 21:44:22 +1000 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/seek/seek.c:
examples/seek: Don't unpause on clock-lost unless playing
If the pipeline is paused by the user, don't pause/unpause
on clock-lost.
2010-07-02 12:10:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: improve debugging
2010-07-02 12:09:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.h:
ringbuffer: whitespace fixes
2010-06-28 10:53:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: use more efficient way of getting caps
When inspecting the caps of a pad, try to get the pad _CAPS first before calling
the getcaps function.
2010-08-02 11:06:00 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/pipelines/oggmux.c:
oggmux: Fix test build when theora and vorbis aren't available
Ifdef properly to avoid build failures
2010-08-01 06:50:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: Detect avc1 ftyp as video/quicktime
Detects avc1 ftyp as video/quicktime (iso variant)
2010-07-27 11:25:12 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: removing useless locking
Everything in the xmp helper lib is initiallized once and on a thread
safe way, and after that there are only reads going on, no more
writing. Based on that, drop the locking.
2010-06-20 23:53:38 +1000 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/seek/jsseek.c:
jsseek: Set joystick io encoding to 'NULL'
Fix problems with newer glib reporting bad encodings on the binary
data emerging from the joystick device fd.
2010-07-26 20:25:55 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: fix printf format compiler warnings
Make OSX build bot happy.
2010-07-26 18:23:33 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: initialize datetime variable in xmp tag parsing code
Fixes (correct) compiler warning on the OSX build bot.
2010-07-26 17:48:14 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: require core from git
For GstDateTime stuff used in libgsttag.
2010-07-26 17:04:02 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
* configure.ac:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstpluginsbaseversion.c:
* gst-libs/gst/pbutils/gstpluginsbaseversion.h.in:
* gst-libs/gst/pbutils/pbutils.h:
* tests/check/libs/pbutils.c:
* win32/common/libgstpbutils.def:
pbutils: add compile time and runtime version checks for gst-plugins-base
So people can check what version of the gst-plugins-base libs they're
building against or linked against.
API: GST_PLUGINS_BASE_VERSION_MAJOR
API: GST_PLUGINS_BASE_VERSION_MINOR
API: GST_PLUGINS_BASE_VERSION_MICRO
API: GST_PLUGINS_BASE_VERSION_NANO
API: GST_CHECK_PLUGINS_BASE_VERSION
API: gst_plugins_base_version()
API: gst_plugins_base_version_string()
2010-06-30 16:36:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Map GST_TAG_DATE_TIME
Adds mapping to the exif helper library for GST_TAG_DATE_TIME.
Tests included.
https://bugzilla.gnome.org/show_bug.cgi?id=594504
2010-06-23 12:02:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Maps GST_TAG_DATE_TIME
Adds mapping for GST_TAG_DATE_TIME.
Tests included.
https://bugzilla.gnome.org/show_bug.cgi?id=594504
2010-07-26 16:05:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Fixate the pixel-aspect-ratio if necessary
2010-07-24 18:17:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Delay EOS events until all streams are EOS
This fixes a race condition in playbin2's gapless mode, where the
EOS of other streams might arrive in the sinks before the last stream
ends and the switch to the new track happens. The EOS sinks won't
accept any new data then and playback stops.
To prevent this, delay all EOS events until all streams are EOS
and advance the sinks of the EOS streams by filler newsegment
events if necessary.
Fixes bug #625118.
2010-06-01 23:43:45 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: export 3gp profile in caps
This reads the 3gp profile from the major/compatible brands and puts
this as a 'profile' field in caps. This can be used by demuxers to
decide whether they can handle this stream or not. Also needed for
DLNA.
https://bugzilla.gnome.org/show_bug.cgi?id=620291
2010-07-24 11:48:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
examples: Use cairo instead of to-be-deprecated GDK API
Fixes bug #625001.
2010-07-24 09:22:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
* configure.ac:
configure: set release date/time
Use the new AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO macro.
2010-07-20 12:08:52 +0530 Parthasarathi Susarla <partha.susarla@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: detect enhanced AC-3
https://bugzilla.gnome.org/show_bug.cgi?id=623846
2010-07-22 09:13:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
* gst/playback/gststreamsynchronizer.h:
streamsynchronizer: Fix another deadlock when going PAUSED->READY while streams are waiting for the GCond
2010-07-20 21:05:45 +0200 Edward Hervey <bilboed@bilboed.com>
playsink: Switch to faster pad linking methods
Logic for choice of GST_PAD_LINK_CHECK_* is as follows:
* Where return of pad_link wasn't checked before : NOTHING
* Where linking is between known compatible elements : NOTHING
* All other cases : TEMPLATE_CAPS
Slashes down playsink reconfigure by up to 50% cpu time.
2010-07-19 15:58:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
* gst/playback/gstscreenshot.c:
playsink: Set add-borders=true on the videoscale instances
This makes sure that we always keep the display aspect ratio and
add black borders if necessary, which is usually something you want
for viewing a video.
2010-07-19 15:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Rename borders property to add-borders
2010-07-19 09:39:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
videoscale: update disted orc files for latest changes
2010-07-17 20:24:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/Makefile.am:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videoscale/vs_fill_borders.c:
* gst/videoscale/vs_fill_borders.h:
* gst/videoscale/vs_image.h:
videoscale: Add support for adding black borders to keep the DAR if necessary
Fixes bug #617506.
2010-07-18 15:08:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_scanline.c:
videoscale: Fix linear scaling of UYVY scanlines
Fixes bug #624656.
2010-07-17 19:57:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Fix caps fixating if the height is fixed but the width isn't
2010-07-16 20:41:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
videoscale: Remove interlaced scaling again
This behaviour was not preferred and caused visible image quality
degradations. The real solution would be, to apply a real
deinterlacing filter before scaling the frames.
Fixes bug #615471.
2010-07-16 19:06:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Add helper method for filling the VSImage struct
2010-07-18 11:43:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/files/Makefile.am:
tests: don't forget to dist test file for typefinding unit test
2010-07-18 11:38:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/gst/typefindfunctions.c:
* tests/files/623663.mts:
tests: add unit test for mpeg-ts typefinding bug
See #623663.
2010-07-18 11:24:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: make mpeg-ts typefinder scan more data
We only look for packets with payload, but it appears there may be packets without,
which makes it harder to find the N packets with payload in a row that we need in
order to typefind this successfully, so scan some more data than necessary in the
optimistic scenario. Alternatively we could change IS_MPEGTS_HEADER().
Fixes #623663.
2010-07-16 18:51:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
* gst/playback/gststreamsynchronizer.c:
playsink/streamsynchronizer: Remove and deactivate pads after calling the change_state function of the parent class
Fixes some deadlocks.
2010-07-16 18:25:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Drop DISCONT flag on first buffer for new streams
Also reset stream state when going back to READY and on flush-stop.
2010-07-11 14:44:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstplaysink.c:
* gst/playback/gststreamsynchronizer.c:
* gst/playback/gststreamsynchronizer.h:
* gst/playback/test7.c:
playsink: Fix gapless playback in many non-simple scenarios
Before gapless playback failed when switching between audio-only,
video-only and audio-video files, when choosing different clocks
and when the different streams had different durations.
This is now handled by a helper element, which keeps track of the
running times of all streams and synchronizes them.
Fixes bug #602437.
2010-07-11 14:43:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Remove QOS event adjustments for gapless playback mode
2010-07-09 17:15:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: Post clock-provide and clock-lost messages when going from/to PLAYING
2010-07-09 17:15:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: Post clock-provide and clock-lost messages when going from/to PLAYING
2010-07-08 16:11:12 +0200 Philip Jägenstedt <philipj@opera.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: only associate .webm with WebM
.weba (audio) and .webv (video) were speculation on my part before
the public launch. As of yet no decision has been made on the
file extension for audio-only WebM, and I'm pretty sure there will
never be one for video-only.
Fixes bug #623837.
2010-07-08 09:54:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: Use new gst_audio_clock_new_full()
2010-07-08 09:54:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: Use new gst_audio_clock_new_full()
2010-07-08 08:32:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
* win32/common/libgstaudio.def:
audioclock: API: Add gst_audio_clock_new_full() with a GDestroyNotify for the user_data
Elements usually use their own instance as instance data but the
clock can have a longer lifetime than their elements and the clock
doesn't own a reference of the element.
Fixes bug #623807.
2010-07-04 20:29:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/theora/Makefile.am:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraenc.h:
theoraenc: Implement two pass encoding
Fixes bug #621349.
2010-07-04 20:14:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* ext/theora/gsttheoraenc.c:
configure: Require libtheora >= 1.1
It's more than a year old at the time of the next -base release,
has many encoder and decoder improvements and gets us rid of a lot
of #ifdefs
2010-07-04 20:08:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
theora: Use PROP_ instead of ARG_ for property enum values
2010-05-04 12:09:57 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: use proper error message code for failing state change
2010-07-16 11:24:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development
=== release 0.10.30 ===
2010-07-15 01:20:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.30
2010-07-15 00:32:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/cs.po:
* po/lv.po:
po: update translations
2010-07-14 12:59:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Disconnect and destroy uridecodebins when going from READY to NULL
Fixes spurious errors that happen after an error and playing a working
stream afterwards or signals that are emitted for non-active groups.
Fixes bug #624266.
2010-07-08 14:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/design/Makefile.am:
docs: dist more of the gst-plugin-base design docs
2010-07-07 00:35:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.29.4 pre-release
2010-07-07 00:24:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/LINGUAS:
* po/es.po:
* po/fr.po:
* po/it.po:
* po/nl.po:
* po/pt_BR.po:
* po/sl.po:
* po/sv.po:
po: update translations
2010-07-06 09:47:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
Revert "playbin2: Don't put "raw" subtitle types in the raw caps for decodebin2"
This reverts commit 9d7538247ff1bf9841b53eeb71ddc47f2c662415.
If the DVD subpicture caps are not part of the raw caps, uridecodebin
doesn't qualify resindvdbin as raw source and plugs decodebins, which
causes broken DVD playback because of bugs elsewhere.
This change was originally added to only expose supported, raw subtitles,
e.g. if the subtitle sink did not support DVD subpictures but a converter
to some supported format exists. It's not very important right now because
we have nothing (that is autoplugged) to convert from plaintext/pango-markup
or DVD subpictures to something else.
Fixes bug #623583.
2010-07-04 17:27:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Use correct Y offset for YVYU -> RGB conversions
Fixes bug #623530.
2010-07-04 17:26:03 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Use correct Y offset for the YUY2 -> RGB conversions
Fixes bug #623530.
2010-07-04 14:55:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/plugins/inspect/plugin-ogg.xml:
docs: update ogg introspection info after riff fourcc addition
2010-07-02 20:09:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix conversion of packed 4:2:2 YUV to 8 bit grayscale
The last pixel wasn't written before for odd widths.
Fixes bug #623418.
2010-07-02 14:56:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Fix conversion of packed 4:2:2 YUV to RGB
The last pixel wasn't written before.
Fixes bug #623384.
2010-07-02 13:59:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix invalid memory accesses with odd widths/heights during subsampling
Fixes bug #623375.
2010-07-01 21:21:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: If setup of the source element fails in READY->PAUSED deactive the current group
Otherwise the uridecodebin will be still a child of playbin2 and
its signals will still be connected. In future state changes this
will then emit unrelated signals that will confuse playbin2 or,
even worse, cause crashes and assertions.
Fixes bug #623318.
2010-06-30 21:20:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: add FLV4 fourcc and map it to video/x-vp6-flash
Fixes #623176.
2010-06-30 15:13:10 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
netbuffer: declare with G_DEFINE_TYPE for type safety
Fixes #623233.
2010-06-24 16:55:57 +0200 Fredrik Söderquist <fs@opera.com>
* ext/ogg/gstoggdemux.c:
oggdemux: Handle errors from _get_next_page in _do_seek.
If the source element failed here, oggdemux would crash.
Fixes #623218.
2010-06-30 11:00:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: keep things sorted alphabetically
On special request. Because it's important, apparently.
2010-06-29 18:48:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.29.3 pre-release
2010-06-29 18:46:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: fix --disable-external
2010-06-28 15:43:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* autogen.sh:
* configure.ac:
Bump automake requirement to 1.10
For maintainability reasons and $(builddir).
Fixes #622944.
2010-06-27 10:43:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
examples: Remove some #if GTK_CHECK_VERSION(2,12,0)
We depend on GTK+ >= 2.14 already.
2010-06-26 21:28:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/Makefile.am:
videotestsrc: Explicitely link with $(LIBM)
2010-06-26 21:27:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/Makefile.am:
videoscale: Explicitely link with $(LIBM)
2010-06-26 18:19:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/video-enumtypes.c:
0.10.29.2 pre-release
2010-06-26 18:19:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/adder/gstadderorc-dist.c:
* gst/adder/gstadderorc-dist.h:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.h:
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
gst: update orc files
2010-06-26 18:19:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translations
2010-06-26 17:55:12 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Properly clean DecodeChain after errors.
If an error happens, the PAUSED state will never be reached. If an
application re-uses decodebin2 (like totem) where one would normally
set to READY between each file, the cleanup that normally happens in
the PAUSED=>READY codepath will never be called, resulting in the
following file to re-use the previous demuxer/decoder/...
https://bugzilla.gnome.org/show_bug.cgi?id=622807
2010-06-26 12:39:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/design/design-orc-integration.txt:
docs: fix a few typos
2010-06-26 12:03:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videoscale.c:
checks: simplify GstBus usage in videoscale unit test
There's no need to run a main loop, add a bus watch and deal with
helper structs here just to wait for an EOS message.
2010-06-26 11:38:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videoscale.c:
checks: speed up videoscale unit test a little
Use new gst_element_link_pads_full() function to link elements,
and disable all checks when linking (don't try this at home).
Down to 18s from 3m20s. Scary.
2010-06-25 17:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: check for ringbuffer state first
Check for the state of the ringbuffer before doing the checks of the other
buffer properties, when we're not started, we don't care about those values.
2010-06-24 13:30:59 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Use a more concise pad template
Speeds up caps nego 2 fold
https://bugzilla.gnome.org/show_bug.cgi?id=622696
2010-06-24 15:31:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/icles/audio-trickplay.c:
tests: make audio-trickplay test compile when the gst debugging system is disabled
Fixes unused variable warning in that case.
2010-06-24 15:13:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/gst/typefindfunctions.c:
tests: add test that runs all typefinders over random data
2010-06-06 12:31:35 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: Mark ISO 14496-14 files as video/quicktime
These are currently being marked as audio/x-m4a which is incorrect.
https://bugzilla.gnome.org/show_bug.cgi?id=620720
2010-06-24 13:42:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
.gitignore: add temporary orc test directory
2010-06-24 13:30:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
tests: add plugin loading whitelist to test environment
Only want to load core/-base plugins here.
2010-06-24 15:09:04 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 73ff93a to a519571
2010-06-24 08:41:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gsttageditingprivate.c:
tag: Fix printf format string
Use %s for strings, not %d.
2010-06-24 12:06:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_scanline.c:
videoscale: Fix resampling of ARGB scanlines
Previously we would read behind the end of the source lines.
2010-06-16 14:08:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Adds GST_TAG_IMAGE_ORIENTATION mapping
Adds GST_TAG_IMAGE_ORIENTATION mapping to xmp helper lib.
Tests included.
2010-06-16 11:19:37 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/gsttageditingprivate.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* tests/check/libs/tag.c:
* win32/common/libgsttag.def:
tag: exif: Adds mapping for GST_TAG_IMAGE_ORIENTATION
Adds GST_TAG_IMAGE_ORIENTATION to the exif helper lib mapped tags.
Tests included.
2010-06-23 12:10:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: clear ts-offset pointer
We need to clear the pointer to our ts-offset element when we destroy the video
chain elements to make sure nobody derefs it to invalid memory afterwards.
2010-06-23 10:16:07 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaysink.c:
playsink: Reset ts_offset field when freeing chain
Otherwise we would end up with a bogus ->audiochain->ts_offset field
which would cause segfaults/assertions when trying to modify the
'ts-offset' property in update_av_offset().
Was easy to trigger when using a list of audio+video files mixed with
video-only files in totem.
2010-06-18 16:37:14 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/check/elements/adder.c:
* tests/check/elements/appsink.c:
* tests/check/elements/audiotestsrc.c:
* tests/check/elements/gdpdepay.c:
* tests/check/elements/gdppay.c:
* tests/check/elements/multifdsink.c:
* tests/check/elements/videotestsrc.c:
* tests/check/elements/vorbisdec.c:
tests: use our own macros for the tests main function
2010-06-18 14:17:30 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
tag: Use gst_tag_list_peek_string_index in vorbistag
Use _peek_string_index instead of _get_string_index to avoid
a string copy
2010-06-14 12:27:02 +0200 Philippe Normand <pnormand@igalia.com>
* sys/ximage/ximagesink.c:
* sys/ximage/ximagesink.h:
ximagesink: Ask pad peer to accept new caps once only
In buffer_alloc, if the buffer caps are new, call
gst_pad_peer_accept_caps once only, it's useless to call it in the
cases where we know it will always fail.
Fixes bug #621190
2010-06-17 17:07:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Add YUY2/YVYU to all RGB formats conversions
2010-06-17 16:57:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix Y42B to YUY2/YVYU/UYVY conversion for odd widths
2010-06-17 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix YUY2/YVYU/UYVY to Y42B conversion for odd widths
2010-06-17 16:06:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
docs: update introspected plugin docs for gstdoc-scangobj and other changes
Update common for latest gstdoc-scangobj and inspect xml files for
escaping and pad template order changes. Update other gtk-doc files
for API additions and object hierarchy changes.
2010-06-16 19:15:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: improve autoplugging
Use the pad caps when they are available to continue the autoplugging. If the
pad caps are set, they are fixed and then we can directly continue autoplugging.
2010-06-15 16:49:17 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From 9339ccc to 35617c2
2010-06-15 16:53:49 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 5adb1ca to 9339ccc
2010-06-15 16:34:54 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 57c89b7 to 5adb1ca
2010-06-15 15:32:34 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From c804988 to 57c89b7
2010-06-15 13:09:37 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/elements/audioresample.c:
Revert "audioresample: set pads as negotiable"
This reverts commit 5f74f3a82eb54f9a9517f99dffbe45ce4d474870.
2010-06-15 13:09:29 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/elements/audioconvert.c:
Revert "audioconvert: set pads negotiable"
This reverts commit bbd7dee8f604bd0373a82e6e5cc3eec8313806ac.
2010-06-14 15:19:32 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/vs_scanline.c:
videoscale: Fix black horizontal line in image
2010-06-14 15:05:16 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Init char variable for gps coordinates
Initialize char variable for gps coordinates deserialization to 0
to identify when it couldn't be parsed/found and error out.
Fixes #621509
2010-06-14 18:10:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/audioconvert.c:
audioconvert: set pads negotiable
2010-06-14 17:48:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/audioresample.c:
audioresample: set pads as negotiable
2010-06-14 16:25:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Revert accidental downgrade of common revision.
2010-06-14 16:07:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/videoscale.c:
videoscale: And only expect a single buffer in the unit test
2010-06-14 16:02:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/videoscale.c:
videoscale: Only convert one buffer instead of five
Should make the unit test a lot faster.
2010-06-14 14:13:32 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Fix unitialized variables
yay macosx compilers :(
2010-06-14 14:13:16 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/video/video.c:
video: Fix unitialized variable.
yay macosx compilers :(
2010-06-14 13:27:01 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Use Quarks for structure name/field checking
2010-06-14 13:26:02 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Speed up _remove_format_info
Instead of copying full caps, use the fact that the provided caps only have
one structure and only copy around structures.
2010-06-14 13:24:06 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Transfer structures instead of copying them
Avoids many expensive structure copies
2010-06-14 13:20:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Use GLIB_EXTRA_CFLAGS
2010-06-14 13:02:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 7a0fdf5 to c804988
2010-06-14 11:31:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 6da3bab to 7a0fdf5
2010-06-14 11:20:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/jsseek.c:
jsseek: Don't use deprecated GLib API
Fixes once again bug #605100.
2010-06-14 11:16:45 +0200 Prahal <prahal at yahoo.com>
* gst/playback/gstdecodebin2.c:
decodebin2: use accumulator for autoplug-sort
Use an accumulator for the autoplug-sort signal so that we can stop the emission
when a signal handler produced a valid result. This avoids the object handler
to overwrite the results from user signals.
Fixes #621161
2010-06-14 11:11:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: activate_chain must not be called with a NULL chain
It will crash later and shouldn't really happen anyway unless
something is really wrong.
2010-06-14 11:08:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gnomevfs/gstgnomevfssrc.c:
gnomevfssrc: Fix possible NULL pointer dereference
It's always an error if gst_buffer_try_new_and_alloc() returns NULL
2010-06-14 11:03:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Return FALSE from the seek handler if no seek callback was set
2010-06-14 09:53:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiostreamsrc.c:
giostreamsrc: Fix copy&paste error in the docs
2010-06-14 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: fix debug message printf format some more
Just cast the pointer diff, so it works everywhere without
warnings. Can't use %tu, because that modifier is C99. Warning
was: "format '%li' expects type 'long int', but argument 8 has
type 'int'".
2010-06-13 22:17:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/videoscale.c:
videoscale: Add ffmpegcolorspace after videotestsrc for the unit test
2010-06-13 20:57:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: ...and add Y16 case for the linear scaling
2010-06-13 20:38:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Add Y16 case for 4-tap scaling
2010-06-13 18:27:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/Makefile.am:
tests: Fix linking of the tags test
2010-06-13 08:20:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.h:
video: Re-add (but deprecated) GST_VIDEO_{RED,GREEN,BLUE}_MASK_1[56]
2010-06-12 21:04:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Use correct variables for debug output
2010-06-12 16:51:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix Y16 from/to GRAY8 conversion
2010-06-12 16:31:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Don't crash when doing gray YUV to GRAY conversion
2010-06-12 16:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
videoscale: Update disted orc files
2010-06-12 16:16:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Allow video/webm for progressive downloading
2010-06-12 13:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Add support for more gray formats
2010-06-01 16:45:34 +0000 Martin Bisson <martin.bisson@gmail.com>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video.{c,h}: Fix an endianness bug fix.
This commit makes sure the endianness is ok for RGB/BGR 15/16 formats.
2010-06-01 14:42:54 +0000 Martin Bisson <martin.bisson@gmail.com>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video.{c,h}: Add support for RGB and BGR with 15 and 16 bits.
2010-06-12 13:35:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
videoscale: Use libgstvideo for caps parsing, etc
2010-06-12 13:04:43 +0200 Philippe Normand <phil@base-art.net>
* ext/ogg/gstoggstream.c:
oggdemux: Fix format string compiler warning on OS X
2010-06-12 13:00:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Use GST_VIDEO_CAPS_GRAY{8,16}
2010-06-12 12:57:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videoscale/vs_scanline.c:
videoscale: Implement linear merging of Y16 scanlines with orc
2010-06-12 08:26:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 733fca9 to 6da3bab
2010-06-11 22:16:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* ext/cdparanoia/Makefile.am:
cdparanoia: check for cdparanoia with pkg-config first
cdparanoia now has a .pc file in post-0.10.2 SVN, so use
that to check for cdparanoia before we try all the other
checks. Besides being generally nicer, this may help with
correctly detecting cdparanoia on OSX some day (see #609918).
2010-06-11 12:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: look for dts frames at non-zero offsets too
Scan a bit into the data when checking for dts frames instead
of expecting the frame sync to be right at the start of the
data. This is needed for some dts-disguised-as-pcm-in-wav files.
See #413942.
2010-06-10 18:12:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: add typefinder for dts audio
2010-06-11 15:23:14 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/tag/gstexiftag.c:
gstexiftag: Fix unitialized variables
I hate thee macosx
2010-06-11 08:47:27 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/tag/gstexiftag.c:
gstexiftag: Fix debug statements
2010-06-11 08:47:17 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/tag/gstexiftag.c:
exiftag: Fix unitialized variable
2010-06-10 20:45:42 +0300 Stefan Kost <ensonic@users.sf.net>
* win32/common/libgsttag.def:
win32: update def file
2010-06-10 20:36:32 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/tag.h:
docs: fix gtk-doc warnings
Variable names in function prototypes in the headers should match the doc-
comment.
2010-06-10 08:47:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: If the text-sink claims to support ANY caps assume it only support raw plaintext subtitles
Fixes bug #621071.
2010-06-10 08:46:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/icles/playbin-text.c:
icles: Only accept plain subtitles in the playbin-text icles test
2010-06-09 22:34:24 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add support for VP6F (On2 VP6 Flash variant)
2010-06-09 12:35:40 -0700 David Schleef <ds@schleef.org>
* configure.ac:
Use the Orc m4 macro
2010-06-09 12:40:00 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From fad145b to 733fca9
2010-06-09 12:33:51 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From 47683c1 to fad145b
2010-06-09 15:58:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
tag: exif: Refactor byte-order handling
Only check for valid byte-order values when creating the
exif readers and writers
2010-05-10 14:01:46 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
* tests/check/libs/tag.c:
tag: exif: Adds new geo-location tag mappings
Adds mappings for:
GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION
GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION
GST_TAG_GEO_LOCATION_MOVEMENT_SPEED
GST_TAG_GEO_LOCATION_ELEVATION
Does some refactoring in the code to reduce number of parameters
passed to functions
Tests included.
2010-04-04 22:25:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/tag.c:
tests: tag: Adds unit tests for exif helper lib
Adds some simple unit tests for exif helper lib functions
Fixes #614872
2010-04-03 23:02:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstexiftag.c:
* gst-libs/gst/tag/tag.h:
* win32/common/libgsttag.def:
tag: Adds basic exif tags support
Adds exif helper lib functions to parse exif buffers from/to
taglists. Exif is tipically used in jpeg images, but it can
also be embedded into TIFF, AVI and WAV formats.
Adds a couple function to handle exif in tiff header structures, that is how
exif is embedded in jpeg and (obviously) in tiff.
API: gst_tag_list_to_exif_buffer
API: gst_tag_list_to_exif_buffer_with_tiff_header
API: gst_tag_list_from_exif_buffer
API: gst_tag_list_from_exif_buffer_with_tiff_header
Fixes #614872
2010-06-09 17:02:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Handle SEEKING query in push mode too
2010-06-09 16:38:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Update the total time from the Skeleton 4 indexes
Fixes bug #620939, see bug #607945.
2010-06-09 16:33:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: Implement latest version of the Skeleton 4.0 spec
Fixes bug #620939.
2010-06-09 16:59:10 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/volume/gstvolume.c:
volume: make the orc codes available for testing.
Add a USE_ORC define for now and switch 'this' to 'self'. Having orc enabled
passes the test suite and various manual gst-launch pipelines.
2010-06-08 13:34:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
.gitignore: add orc-related temp files
2010-06-08 13:26:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* gst/audioresample/Makefile.am:
* gst/audioresample/gstaudioresample.c:
Fix build if orc is not installed
Orc is not a hard requirement. Things should still compile and
work without orc, but slow fallback code may be used in this
case. Fix up configure to not error out if orc is not installed
and wrap use of orc profiling in audioresample in #ifdefs.
Fixes #620136 some more.
2010-06-08 13:11:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: Implement correct parsing of Skeleton 4.0 index packets
2010-06-08 12:01:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: Add parsing of Skeleton 4.0 indexes
2010-06-08 11:40:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: Parse segment length and content offset from fishead
And print them for debugging purposes. Not sure if we can do anything useful
with this information.
2010-06-08 11:31:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: Parse Skeleton stream major/minor version
2010-06-08 11:26:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: Use binary search for searching in the index
2010-06-08 11:02:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/video.c:
video: Fix unit test, the Y800 checks were not used before and were not working
2010-06-08 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Return TRUE in gst_video_format_is_gray() for Y800 and Y16
2010-06-08 00:33:31 -0700 David Schleef <ds@schleef.org>
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
audioconvert, videotestsrc: Update generated Orc code
Fixes compile errors with initialization of unions.
2010-06-08 00:32:36 -0700 David Schleef <ds@schleef.org>
* REQUIREMENTS:
requirements: change liboil to Orc
2010-06-06 23:50:05 -0700 David Schleef <ds@schleef.org>
* gst/audioresample/Makefile.am:
* gst/audioresample/gstaudioresample.c:
audioresample: convert from liboil to orc
2010-06-06 23:48:35 -0700 David Schleef <ds@schleef.org>
* tests/check/Makefile.am:
tests: Add orc tests
2010-06-06 23:48:15 -0700 David Schleef <ds@schleef.org>
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c:
* gst/volume/gstvolumeorc-dist.c:
* gst/volume/gstvolumeorc-dist.h:
* gst/volume/gstvolumeorc.orc:
volume: convert from liboil to orc
2010-06-06 23:47:53 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/Makefile.am:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.h:
* gst/videotestsrc/gstvideotestsrcorc.orc:
* gst/videotestsrc/videotestsrc.c:
videotestsrc: convert from liboil to orc
2010-06-06 23:47:16 -0700 David Schleef <ds@schleef.org>
* gst/videoscale/Makefile.am:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.h:
* gst/videoscale/gstvideoscaleorc.orc:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.c:
* gst/videoscale/vs_scanline.h:
videoscale: convert from liboil to orc
2010-06-06 23:46:41 -0700 David Schleef <ds@schleef.org>
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.h:
* gst/audioconvert/gstaudioconvertorc.orc:
audioconvert: convert from liboil to orc
2010-06-06 23:45:58 -0700 David Schleef <ds@schleef.org>
* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
* gst/adder/gstadderorc-dist.c:
* gst/adder/gstadderorc-dist.h:
* gst/adder/gstadderorc.orc:
adder: convert from liboil to orc
2010-06-06 23:45:10 -0700 David Schleef <ds@schleef.org>
* docs/design/Makefile.am:
* docs/design/design-orc-integration.txt:
docs: Add notes about Orc integration
2010-06-06 23:34:39 -0700 David Schleef <ds@schleef.org>
* configure.ac:
configure: convert liboil check to orc
2010-06-08 07:34:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Start a new page for every CMML buffer
2010-06-07 14:38:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2: add av-offset property
Add av-offset property to control the audio and video sync offset. This can be
used to to manually correct badly synced streams.
See #620529
2010-06-07 08:31:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
ffmpegcolorspace: Map "Y8 " and "GREY" to "Y800" and add it to the template caps
2010-06-07 08:17:13 +0200 Martin Bisson <martin.bisson@gmail.com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Add support for Y800 and Y16
Fixes bug #620441.
2010-06-07 08:16:01 +0200 Martin Bisson <martin.bisson@gmail.com>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add support for Y800 and Y16
Fixes bug #620441.
2010-06-06 16:46:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: fix log function printf format issue
2010-06-05 18:14:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: stop jpeg typefinding once we found a SOF marker
2010-06-05 18:05:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/gst/typefindfunctions.c:
tests: fix memory leak in unit test
2010-05-19 15:40:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: improve jpeg typefinder
Make jpeg typefinder check more than just the first two bytes
plus Exif or JFIF marker. This allows us to report MAXIMUM
probability in cases where there's no Exif or JFIF marker,
making typefinding stop early. Also extract width and height,
because we can.
2010-06-05 17:22:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/gst/typefindfunctions.c:
* tests/files/Makefile.am:
* tests/files/partialframe.mjpeg:
tests: add small unit test for AC3 vs. JPEG typefinding issue
2010-06-05 16:58:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: fix AC-3 typefinding so that it actually checks for a second frame
Fix typo that made the AC-3 typefinder not actually check for a
second frame, but rather compare the sync point found to itself,
which resulted in the AC-3 typefinder reporting an overly optimistic
MAXIMUM or VERY_LIKELY probability when it found a possible frame
sync.
2010-06-05 12:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstscreenshot.c:
playbin2: improve screenshot code
Use appsrc and appsink in the screenshot code to make things nicer.
2010-06-05 11:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: fix documentation string
2010-06-05 11:05:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: add convert-frame action signal
Add a convert-frame action signal.
Fixes #620279
2010-06-05 11:02:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstscreenshot.c:
* gst/playback/gstscreenshot.h:
playbin2: move marshaller to screenshot
Move the marshaller for the convert_frame signal to the screenshot file in
preparation for moving it to playsink.
See #620279
2010-06-05 10:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2: move convert_frame to playsink
Move the convert_frame function to playsink and make it part of the API. This is
in preparation to add the convert_frame signal to playsink.
See #620279
2010-06-05 10:31:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: add property to get the last frame
Add a property to get the last video frame.
See #620279
2010-06-04 19:30:14 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Handle raw streams we don't want.
If a file contains raw streams (not requiring a decoder) that we do
not want (expose-all-streams == FALSE), we would previously consider
those of unknown-type (missing a decoder) ... whereas in fact it was just
because they don't need decoders.
This only applies if expose-all-streams is FALSE.
2010-06-03 13:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
Revert "baseaudiosink: Allocate and free the clock in NULL->READY and reverse"
This reverts commit cea2644ed86097aadedc9e8731e78a22ffc6246b.
Many audio sink assume that they can create a clock in
the instance init function and it will be there forever
and not be cleared by the state change functions.
2010-06-02 12:19:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: Allocate and free the clock in NULL->READY and reverse
2010-06-01 23:49:07 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From 17f89e5 to 47683c1
2010-06-01 22:54:33 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From fd7ca04 to 17f89e5
2010-06-01 13:00:22 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* tests/examples/overlay/Makefile.am:
examples: get the right Qt moc binary to use via pkg-config
Should make us do the right thing in cases where both Qt3 and Qt4
are installed.
Fixes #620211.
2010-05-31 19:28:45 +1000 Jonathan Matthew <jonathan@d14n.org>
* ext/gio/gstgiobasesink.c:
gio: map GIO NO_SPACE error to NO_SPACE_LEFT
Fixes bug #620140.
2010-05-28 08:27:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* gst-libs/gst/interfaces/streamvolume.c:
configure: Remove (now) useless check for cbrt
2009-12-02 22:16:22 -0800 David Schleef <ds@schleef.org>
* gst-libs/gst/interfaces/streamvolume.c:
interfaces: Use pow() instead of cbrt() for MSVC
2010-05-26 11:54:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 357b0db to fd7ca04
2010-05-26 08:51:09 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/audiorate/gstaudiorate.c:
audiorate: Fix buffer offset_end when within tolerance.
This fixes issues if we then have downstream elements that operate
on offset/offset_end.
And add the expected timestamp in the debug logs
2010-05-24 11:27:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/fft/kiss_fft_f32.c:
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.c:
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.c:
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.c:
* gst-libs/gst/fft/kiss_fft_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c:
* gst-libs/gst/fft/kiss_fftr_f64.c:
* gst-libs/gst/fft/kiss_fftr_s16.c:
* gst-libs/gst/fft/kiss_fftr_s32.c:
fft: Merge kissfft 1.2.8
This reduces memory footprint for the FFT and adds
OpenMP support (but we don't use it).
2010-05-22 10:05:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: Fixate interlaced, chroma-site and color-matrix fields if necessary
2010-05-22 10:02:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* gst/videorate/gstvideorate.c:
videorate: Use new string fixation function from core
2010-05-22 09:48:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Fixate color-matrix and chroma-site fields if necessary
2010-05-22 09:39:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Fixate the interlaced field if necessary
Fixes bug #619310.
2010-05-22 08:55:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add IVF typefinder
2010-05-21 18:16:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: pass object to logging functions, use GST_DEBUG_FUNCPTR
2010-05-20 15:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add basic descriptions for new WebM and VP8 types
2010-05-20 14:21:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Fix sizes again, this time for real
2010-05-20 13:58:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
oggdemux: Fix size checks
2010-05-20 10:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
oggdemux: Drop all other Ogg VP8 header packets and make VP8 mapping check a bit more strict
2010-05-20 08:52:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
ogg: Some more minor adjustments for the VP8 Ogg mapping
2010-05-19 21:35:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
ogg: Update to the latest VP8 mapping
2010-05-10 05:53:22 +0200 Philip Jägenstedt <philipj@opera.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: Detect WebM as video/webm
Refactor matroska_type_find into ebml_check_header and a new
matroska_type_find and webm_type_find.
2010-05-14 13:31:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: Fix granulepos->key granule calculation for Dirac
2010-05-14 11:02:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: Add support for mapping specific granulepos to key granule mapping
2010-05-05 13:59:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
ogg: Implement Ogg VP8 mapping
2010-04-27 15:24:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: Add support for On2 VP8
2010-05-19 16:17:19 +0200 Alessandro Decina <alessandro.decina@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: fix a typo introduced by 9d753824.
video/x-raw-float => audio/x-raw-float. Fixes #619090.
2010-05-18 08:45:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Don't put "raw" subtitle types in the raw caps for decodebin2
We handle them from the autoplug-continue signal, where the caps supported
by the subtitle sink or overlay are known already.
2010-05-15 21:15:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Use = instead of == in shell scripts for equality checks
2010-05-14 18:23:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 4d67bd6 to 357b0db
2010-05-14 17:24:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Always take the object lock when accessing the caps
Fixes bug #618625.
2010-05-14 17:17:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Don't fail if subtitles are used but only audio is available and no visualizations
Instead simply disable displaying of the subtitles for now, as was
intended by that part of code...
Fixes bug #610866.
2010-05-14 17:13:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Fix deadlock caused from an additional lock instead of unlock
Also improve debug output for the playsink lock.
2010-05-13 12:16:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Use passthrough mode if width and height are not changed
It doesn't matter if the PAR changes or not, processing of every pixel
is only necessary when the width or height changes.
2010-05-13 12:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: relax width and height constraints
Increase the acceptable video sizes from [16,4096] to [1, MAX].
See #618392
2010-05-13 08:05:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Use simple fraction multiplication functions instead of going through GValues
2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/videoscale.c:
videoscale: Add a unit test for checking if the negotiation works as expected
2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Try harder to keep the DAR if possible
Fixes bug #371108.
2010-05-10 15:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Log PAR and DAR of input and output caps when setting caps
2010-05-10 14:52:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Set input width/height if the output caps don't have any width or height
2010-05-10 13:01:44 +0200 Andoni Morales <ylatuya@gmail.com>
* gst/videoscale/gstvideoscale.c:
videoscale: Try to keep DAR when scaling
Fixes bug #371108.
2010-05-10 19:09:28 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
basertpaudiopayload: Add extra frame for non-complete frame lengths
Some payloaders like rtpg729pay can add a shorter frame at the end of a
RTP packet. We need to count it like a full frame for timestamps.
https://bugzilla.gnome.org/show_bug.cgi?id=618324
2010-05-10 18:53:29 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
basertpaudiopayload: Set duration on buffers
Set the duration of the buffers from their size
2010-05-11 16:12:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: Fixate PAR to 1/1 if possible
2010-05-11 10:07:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Check for GTK+ 3.0 and if it's not available for GTK+ 2.0
2010-05-10 12:44:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/ximage/ximagesink.c:
ximagesink: Check if the X context is allocated before using it
It should be allocated at these places already or the state changes
would have failed... but better add an additional check here.
2010-05-10 12:28:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/ximage/ximagesink.c:
ximagesink: Post an error message on the bus if no supported pixmap formats can be found
Might fix bug #615851.
2010-05-07 19:49:57 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace : whooops
2010-05-07 19:21:13 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: more minor cleanups
2010-05-07 17:16:28 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: speedup caps transformation
* don't re-create our possible caps every single time, just use the
template caps.
* don't intersect the caps against the template, basetransform has already
done that for us.
62% speedup of _transform_caps() (instruction calls, measured with callgrind)
2010-05-07 12:19:25 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gsturidecodebin.c:
uridecodebin: add the 'expose-all-streams' property from decodebin2
API: expose-all-streams
https://bugzilla.gnome.org/show_bug.cgi?id=617868
2010-05-06 18:50:51 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Add a property to not expose/decode all streams
API : expose-all-streams
If disabled:
* only the streams that CAN be decoded and match the final caps will have a
decoder plugged in and be exposed.
* the streams that COULD HAVE BEEN decoded but do not match the finals caps
will not have a decoder plugged in and will not be exposed.
If no decoder is available to decode a certain stream, then the missing element
message will still be emitted regardless of the value of the property.
https://bugzilla.gnome.org/show_bug.cgi?id=617868
2010-05-06 17:47:12 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: rename are_raw_caps to are_final_caps, correct comment
https://bugzilla.gnome.org/show_bug.cgi?id=617868
2010-05-07 17:16:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/sdp/gstsdpmessage.h:
sdpmessage: add new TIAS bandwidth modifier
Add TIAS modifier as specified in RFC 3890.
Do some whitespace fixes.
2010-05-07 00:10:22 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audioconvert/audioconvert.c:
audioconvert: disambigue comment due to popular demand
Write "target depth" instead of "our depth" or previous ambigous "out depth".
2010-05-06 15:40:34 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: disconnect signals in some more cleanup cases
2010-05-06 13:10:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: don't seek when no current chain
Avoid a crash when we try to seek when there is no current chain.
2010-05-06 12:21:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: ignore the skeleton start time
Ignore the skeleton start time as it is usually wrong for live streams
and we have the needed logic to calculate it anyway.
2010-05-06 12:06:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: wait for headers before exposing chains
Wait until we have all the stream headers before we start exposing the streams
of a chain.
2010-05-06 10:56:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: use index to estimate bitrate
When we have an index, use it to much more accurately estimate the total stream
bitrate.
2010-05-06 11:34:53 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtsp/gstrtsptransport.h:
docs: be more firendly to gtk-doc limitted parsing capabilities
2010-05-06 09:42:02 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspmessage.c:
* gst-libs/gst/rtsp/gstrtsprange.c:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
docs: fix wrong doc markup
2010-05-06 09:17:33 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/videoscale/gstvideoscale.c:
videoscale: use can_intersect to avoid a caps copy
2010-05-06 09:14:25 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/videorate/gstvideorate.c:
videorate: trucate own caps, instead of copying and using the first only
We got the caps from an intersect, it is our own, hence we can truncate it.
Besides gst-indent has chooses to line-up all caps in one line again :/.
2010-05-06 09:12:32 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin.c:
decodebin: use can_intersect to avoid a caps copy
2010-05-06 09:11:17 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/libvisual/visual.c:
libvisual: trucate own caps, instead of copying and using the first only
We got the caps from an intersect, it is our own, hence we can truncate it.
2010-05-06 08:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/gstvorbisdeclib.c:
* ext/vorbis/gstvorbisdeclib.h:
vorbis: have a copy_sample func as a func pointer
Make some more variants for copy_sample funcs and use them via function pointer.
2010-05-06 08:16:45 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audioconvert/audioconvert.c:
audioconvert: fix typo in comment
2010-05-06 08:15:16 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: use gst_caps_can_intersect() more
In place where we just need to know whether caps intersect, we can use this
quicker function.
2010-04-15 13:09:45 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/position-formats.c:
examples: add a test for difference position formats
The test runs position and duration queries on the pipeline in all formats.
2010-04-15 13:08:39 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/icles/audio-trickplay.c:
example: update status (adder is fixed now)
2010-04-15 13:08:01 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/icles/playbin-text.c:
example: make app static
2010-05-05 13:25:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: printf format fixes
2010-05-04 15:32:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: weekday and month names in RTSP date string should be in C locale
Create date string using C locale weekday and month names.
Fixes #617636.
2010-05-04 17:54:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: add all qtdemux types to downloadable types
Add all the media types that qtdemux can handle to the list of downloadable
types.
2010-05-04 17:38:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: handle corrupt indexes
Make sure we handle and receover from corrupt indexes.
2010-05-04 15:47:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix EOS check
2010-05-04 13:51:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: avoild division by 0
2010-05-04 13:50:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: cleanup unused defines
2010-05-04 13:36:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: use the index in push mode when we can
When seeking in push mode, try to use the index first before we use the bitrate
estimation.
2010-05-04 13:05:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: use skeleton duration when possible
2010-05-04 13:02:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggstream: parse duration from 3.3 skeleton
2010-03-02 11:16:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: more index parsing work
2010-03-01 13:50:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: clean up fishead/fisbone parsing
Remove some redundant code for parsing fishead streams. Actually use the data we
parsed (mostly start_time).
2010-05-04 11:19:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: implement seek in push mode
Refactor start time collection code.
When we receive a flush_stop, resync to the new start time and push out a new
segment event.
2010-05-03 16:52:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: make event handling better
Explicitly handle FLUSH events and resync on FLUSH_STOP.
Make send_event return a boolean.
Use more performant send_event function to forward events.
2010-04-30 18:37:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: implement seeking in pushmode
Convert seek requests to bytes using the bitrate and forward them upstream. Does
not quite work because the flushing and resyncing is not implemented yet.
2010-04-30 18:03:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: refactor for seeking in pushmode
refactor the code a little to prepare for seeking in push mode
2010-05-03 12:46:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Adds new geo-location mappings
Adds GST_TAG_GEO_LOCATION_MOVEMENT_SPEED,
GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION and
GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION to xmp
mappings.
Tests included.
2010-04-26 22:08:41 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Adds xmp mappings for device tags
Adds xmp mappings for GST_TAG_DEVICE_MANUFACTURER and
GST_TAG_DEVICE_MODEL. Also adds tests for it.
2010-04-30 19:56:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspbase64.c:
* gst-libs/gst/rtsp/gstrtspbase64.h:
rtsp: deprecate remaining base64 function now that we depend on GLib 2.20
API: deprecate gst_rtsp_base64_decode_ip(), use g_base64_decode_inplace() instead
2010-04-30 19:37:33 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
basertpdepayload: ensure writable metadata
2010-04-30 17:41:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: use bitrate to estimate length in pushmode
Parse the bitrate from the various streams.
Use the bitrate and the upstream length in bytes to estimate the total stream
duration in push mode.
2010-04-30 14:07:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* ext/gio/gstgiobasesrc.c:
Bump GLib requirement to 2.20
See http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
2010-04-30 13:36:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: forward duration query duration during group switch if no cached duration
... such as during first group setup.
Fixes #616396.
2010-04-02 16:37:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: only EOS when all streams are EOS
2010-04-02 16:36:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix debug message
2010-04-30 08:45:43 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/icles/audio-trickplay.c:
test: fix copy and paste error of variable name
2010-04-18 20:46:37 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: only accept seek-types none and set
Previously we were also acting on cur and end, but treating them like none.
2010-04-14 23:31:20 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
adder: rework timestamping
Adder was using always incrementing timestamps. Seeking was done by setting the
position in the newsegment event. This was failing when doing segmented seeks
with rate<0.0, as offset (and thus timestamp) would go below 0.
Now we take both cur and end from the seek event. We construct newsegment events
depending including cur and end from the seek event. We set position to the
start of the segment. Timestamp is set to start or end of segment depending on
rate. Offset is recalculated.
2010-04-26 17:30:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Add support for deinterlacing
This is disabled by default and can be enabled with the
deinterlace flag.
Fixes bug #547603.
2010-04-26 11:12:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplay-enum.c:
* gst/playback/gstplay-enum.h:
playbin2: Add flag for enabling/disabling automatic deinterlacing
2010-04-26 11:11:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplay-enum.c:
playbin: Use g_once_init_{enter,leave} instead of GOnce for enum/flag registration
2010-04-23 17:01:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/Makefile.am:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
ffmpegcolorspace: Use GST_BOILERPLATE and use GstVideoFilter as base class
This gives automatic QoS handling.
2010-04-23 16:24:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Correctly reconfigure the video chain when switching from a subtitle to a non-subtitle file
Fixes bug #616422.
2010-04-23 16:08:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: If a text sink is provided, let subtitle parsing be done by decodebin2 if required
This way subtitle sinks only get buffers in the format that they
understand, i.e. raw parsed text in most cases.
Fixes bug #614942.
2010-04-23 15:30:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Set subtitle encoding on the decodebins again
2010-04-23 15:22:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: API: Add GST_VIDEO_FORMAT_v308 for packed 4:4:4 YUV
2010-04-23 15:14:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
videoscale: Some random cleanup
2010-04-23 15:06:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Add support for Y444, Y42B and Y41B
2010-04-23 14:42:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Reorder template caps by the amount of information contained in the color formats
2010-04-22 15:46:17 -0400 Joshua M. Doe <joshua.doe@us.army.mil>
* gst/videorate/gstvideorate.c:
videorate: add support for video/x-raw-gray
2010-04-29 15:05:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-playbin.xml:
docs: remove references to and introspection data of plugins that no longer exist
Some plugins (decodebin2, playbin) have been renamed or merged
into different plugins (uridecodebin, playback).
2010-04-29 15:02:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development.
=== release 0.10.29 ===
2010-04-28 02:16:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.29
2010-04-28 01:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2010-04-25 23:14:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.28.3 pre-release
2010-04-20 17:20:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-plugins-base.doap:
doap: update repository info from cvs->git and maintainers
2010-04-23 14:39:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From fc85867 to 4d67bd6
2010-04-22 20:58:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Fix Y41B->Y444 conversion
...which is the intermediate conversion for conversion to all
other formats.
Fixes bug #616545.
2010-04-16 20:03:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: Don't leak the input buffer in error cases
Fixes bug #615572.
2010-03-29 12:53:11 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/ogg/gstoggmux.c:
docs: fix typo in link name
2010-04-15 12:59:53 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: gracefully handle ximagesink>xwindow == NULL
Expose could be called before we have set the xwindow. Handle this gracefully
like we do in image_put.
Fixes #615789
2010-04-15 11:44:49 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
ximagesink: refactor _update_geometry()
Refactor like in xvimagesink. Remove the extra parameter and adjust the assert check.
2010-04-15 07:18:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* configure.ac:
configure: Drop -Wcast-align
Commit message copied from core's commit from Benjamin Otte:
246f5dba96a5b50bb74621af67b30942cca72af5
Apparently gcc warns that GstMiniObject is not castable to
GstEvent/Message/Buffer due to them containing 64bit variables, even
though ARM hackers claim that those only need 4byte alignment. And as
long as gcc behaves that way, this warning is not very useful.
So we'll remove the warning until this problem is fixed.
Fixes #615698
2010-04-14 14:13:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* gst-libs/gst/tag/lang-tables.dat:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/video-enumtypes.c:
0.10.28.2 pre-release
2010-04-14 13:50:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translations
2010-04-13 16:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: init geometry when setting new xid
Don't rely on expose event to query geomentry after new xid is set.
Fixes #615647.
2010-04-14 13:43:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/audioconvert/Makefile.am:
* tests/examples/app/Makefile.am:
* tests/examples/dynamic/Makefile.am:
* tests/examples/gio/Makefile.am:
* tests/examples/volume/Makefile.am:
* tests/old/examples/switch/Makefile.am:
build: use LDADD instead of LDFLAGS to specify libs to link to when building executables
Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
This should make sure arguments are passed to the linker in the right
order, and makes LDFLAGS usable again.
Based on initial patch by Brian Cameron <brian.cameron@oracle.com>
Fixes #615697.
2010-04-12 14:02:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: add channels and rate to ADTS caps if we can
2010-04-12 13:33:18 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst/typefind/Makefile.am:
* gst/typefind/gstaacutil.c:
* gst/typefind/gstaacutil.h:
* gst/typefind/gsttypefindfunctions.c:
typefinding: add AAC level to ADTS caps
This adds code to calculate the level for a given AAC stream and export
it in the stream caps. For AAC LC streams, the level is calculated
according to the definition under the AAC Profile. For other streams,
the definition under the Main Profile is used.
HE-AAC support is still to be done, and is dependent on detecting the
presence of SBR and PS in the stream.
Level is added as a field of type string because that's the way it's
done in H.264 caps as well. There are only a few possible levels, so
not using a numerical type is not too painful in this case, and
consistency is nice.
Fixes #613589.
2010-03-10 13:32:53 +0000 Arun Raghavan <arun.raghavan@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: add AAC profile to ADTS caps
This looks at the AAC profile for ADTS streams and adds the profile as a
string in the corresponding caps.
Profile is the actual profile, base-profile denotes the minimum codec
requirements to decode this stream. In this case they're always the
same, but they may differ e.g. in case of certain HE-AAC streams that
can be partially decoded by LC decoders (with loss of quality of course)
if no suitable HE-AAC decoder is available.
Fixes #612312.
2010-04-11 22:58:15 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: add support for negative playback rates
Decrement sample counter when playing backwards. Set proper segment when playing
backwards (0..cur instead or cur..-1). Add more logging and fix a format string.
2010-03-26 19:00:47 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiopayload: use ptime-multiple
Based on patch by Olivier Crête <olivier.crete@collabora.co.uk>
Fixes #613248
2010-04-09 16:06:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
audiopayload: add property to control packet duration
Add a property to specify that the amount of data in a packet should be a
multiple of ptime-multiple.
See #613248
2010-04-09 11:20:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 218568f to fc85867
2010-04-08 17:49:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/Makefile.am:
* gst/playback/Makefile.am:
* gst/playback/gstplayback.h:
playback, ogg: dist new gstplayback.h and gstogg.h
2010-04-09 08:23:33 +0200 Thomas Green <thomasgr33n@googlemail.com>
* gst/playback/gstplaybin.c:
playbin: Only unref the volume element on dispose and when a new audio sink is set
Unreffing it whenever the sinks are removed will make the volume
element unavailable after a playbin reuse because it is only
recreated if the audio sink has changed.
Fixes bug #614288.
2010-04-08 07:39:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Be sure that metadata is writable before setting caps
Call gst_buffer_make_metadata_writable before attempting
to set caps on the buffer.
2010-04-08 12:21:50 +0200 Edward Hervey <bilboed@bilboed.com>
* ext/gio/gstgio.c:
* ext/gnomevfs/gstgnomevfs.c:
ext: Invert rank of gio and gnomevfs elements
2010-04-08 01:26:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
alsa: don't pass non-constant strings as printf format strings
Fixes 'format not a string literal and no format arguments' compiler
warning when compiling with -DGST_DISABLE_PRINTF_EXTENSION.
2010-04-07 20:21:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.h:
docs: add gtk-doc chunks with Since: tags for new GST_VIDEO_CAPS_GRAY* API
2010-04-07 19:07:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* autogen.sh:
* configure.ac:
build: bump autoconf requirement to 2.60 for gobject-introspection.m4
Require autoconf 2.60 (which was released in June 2006).
Fixes #600718.
2010-04-07 17:25:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
video: Fix parsing of 8-bit grayscale caps
2010-04-07 17:21:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.h:
video: API: Add GST_VIDEO_CAPS_GRAY{8,16}
2010-04-07 17:08:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
video: API: Add gst_video_format_is_gray() to the docs
2010-04-07 17:07:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstvideo.def:
video: Add new symbol to the exported symbols list
2010-04-07 17:06:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add support for 8-bit and 16-bit grayscale formats
2010-04-06 10:55:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtspconnection: Handle closed POST socket in tunneling
Catch more socket errors.
Rework how sockets are managed in the GSource, wake up the maincontext instead
of adding/removing the sockets from the source.
Add callback for when the tunnel connection is lost. Some clients (Quicktime
Player) close the POST connection in tunneled mode and reopen the socket when
needed.
See #612915
2010-04-04 21:24:44 -0700 David Schleef <ds@schleef.org>
* configure.ac:
configure: fix cdparanoia check
Linking with libcdda_paranoia.so requires also linking with
libcdda_interface.so.
2010-04-04 18:00:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/libs/tag.c:
tests: tag: Refactor a bit
Refactor xmp tags unit tests and remove an useless assertion.
This will make easier to add unit tests to serialize/deserialize
taglists.
2010-04-04 21:18:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
alsa: Ignore errors when unpreparing or closing the device
Errors could happen here when the device was removed already
or when something is broken anyway. If errors happen here and
they're propagated, the element can't shutdown cleanly.
Fixes bug #614545.
2010-04-04 20:55:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/alsa/gstalsamixer.c:
alsamixer: Detect errors from device polling, stop the task and post an error message
Partially fixes bug #614545.
2010-04-04 12:13:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* tests/examples/seek/Makefile.am:
examples: build silly joystick seek example only on linux
jsseek depends on linux headers and should therefore only be built
on linux.
Fixes #614764.
2010-04-03 22:49:11 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: swap timestamps in forward and reverse mode.
In reverse mode we want use the next next timestamp (and not the other way
around). Fixes the tests again. Also readd a log line that was dropped with
previous commit.
2010-04-03 14:03:45 +0100 Vincent Untz <vuntz@gnome.org>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
libs: point gobject-introspection scanner to .la files
Point g-ir-scanner to the .la file of our library, which hopefully
makes it find the right dependencies in all cases (ie. our locally
built libgstreamer and not the system-installed one). This is also
how it's done in Gtk+ and how it's documented in the wiki, see
http://live.gnome.org/GObjectIntrospection/AutotoolsIntegration
Fixes #603710.
2010-04-02 21:01:25 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
audiotestsrc: implement reverse playback
Support playback at negative rates. When having a GstController assigned, the
element will produce time dependend output.
2010-04-02 20:56:19 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/icles/audio-trickplay.c:
tests: extend audio-trickplay test app
Tell status in top comment. Use debug logging instead of print to be able to
see timing issue in debug log viewer. Add more commandline flags. Test reverse
playback.
2010-04-02 18:56:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: Only use embed_xid if HAVE_X is defined
Fixes bug #614622.
2010-04-01 19:13:22 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/pipelines/basetime.c:
tests/basetime: Don't run test with osxaudiosrc
libcheck runs the actual tests in a forked process and that makes the guys
in Cupertino really sad.
2010-04-01 18:51:17 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/pipelines/capsfilter-renegotiation.c:
tests: Unref the bus once we're done with it
2010-04-01 16:49:37 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
common: Update for new suppressions
2010-04-01 13:55:15 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaysink.c:
gstplaysink: Remove unused variable.
The value of klass is never used
2010-04-01 13:53:37 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Removing dead assignment.
The value of group is overwritten a few lines below before being used.
2010-04-01 13:51:13 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: Remove unused variable
2010-04-01 13:48:42 +0200 Edward Hervey <bilboed@bilboed.com>
* ext/gnomevfs/gstgnomevfssink.c:
gstgnomevfssink: Return the proper GstFlowReturn.
We were always returning GST_FLOW_OK previously even if we encountered errors.
2010-03-30 23:44:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
gst-libs: more gobject-introspection fixes
Use right .pc file variable for compiler includes this time:
g-ir-compiler wants the girdirs not the typelibdirs as includes.
2010-03-30 20:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/jsseek.c:
examples: fix printf format warning in jsseek example
Yes, I know about G_GSIZE_FORMAT.
2010-03-30 19:56:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
gst-libs: fix up gobject-introspection some more
Use new girdir and typlibdir from core .pc files, so we can figure
out the right includes to pass to the gobject-introspection tools,
whether core is installed in the same prefix as gobject-introspection
or in a different prefix or uninstalled. This also keeps us from adding
bogus paths to the includes that only work if core is uninstalled.
Also add some missing includes/pkgs where needed.
2010-03-30 19:29:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/Makefile.am:
Our RIFF library depends on both the audio and tag libraries
Update rules in Makefile.am accordingly.
2010-03-30 15:10:42 +0200 Robert Swain <robert.swain@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Fix aduio_raw_sink typo
2009-11-28 21:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/seek/.gitignore:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/jsseek.c:
examples: Add a silly joystick based shuttle example
2010-03-29 20:07:52 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoraenc.c:
theoraenc: 0-length packets are delta units
2010-03-29 10:47:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/Makefile.am:
gst-libs: build independent sub-directories in parallel if make -jN is used
Build those libraries that don't depend on any other gst-plugins-base
libraries in parallel if make -jN is used.
2010-03-29 00:22:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
* ext/Makefile.am:
* gst/Makefile.am:
* sys/Makefile.am:
* tests/examples/Makefile.am:
build: build plugin and example directories in parallel if make -jN is used
We know our plugins and examples are independent of each other, so may
just as well build them in parallel. Makes the output a bit messy, but
that shouldn't be a problem and can easily be avoided with make -j1.
2010-03-28 21:50:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/Makefile.am:
gst-libs: specify dependencies in Makefile.am to make them explicit
2010-03-24 09:59:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
* sys/xvimage/xvimagesink.c:
* tests/icles/test-xoverlay.c:
xoverlay: change new set_render_rectangle() vfunc to take four arguments so we don't depend on libgstvideo
Don't make libgstinterfaces (and thus libgstaudio etc.) indirectly depend
on libgstvideo by using the GstVideoRectangle helper structure in the API,
which causes undesirable dependencies, esp. with the gobject-introspection
(people will point and laugh at us if they find out that libgstaudio
depends on libgstvideo). Instead, pass the x, y, width and height parameters
directly to the function.
Re-fixes #610249.
2010-03-25 18:45:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: we can handle avi in download mode too
Add avi to the whitelisted types that can be used for download buffering.
2010-03-26 15:57:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: Provide packet duration function for old FLAC mapping too
Fixes bug #613809.
2010-03-18 22:12:40 +0000 Damien Lespiau <damien.lespiau@intel.com>
* autogen.sh:
autogen.sh: Don't call configure with --enable-plugin-docs
configure gives a nice warning:
configure: WARNING: unrecognized options: --enable-plugin-docs
and indeed, I could not find anything in the configure.ac or the m4
macros that would allow enabling that option. Remove it then.
2010-03-24 23:04:43 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Do not remove tag from list twice
There was a but when parsing the tags that removed two tags
from the list when only one was parsed
2010-03-24 14:43:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Add some comments
Just adds some comments explaining some stuff about the
(de)serialization functions. Add myself to the copyright list too.
2010-03-24 10:18:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Adds _USER_RATING mapping for xmp
Adds a new mapping for _USER_RATING on xmp helper lib
and also adds tests for it
2010-03-23 09:32:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Add Elevation tag mapping
Adds a mapping to the _ELEVATION tag, this is a different
mapping as it has to be mapped into exif:GPSAltitude and
exif:GPSAltitudeRef at the same time. So we needed to refactor
a little more to be able to deserialize it properly.
Now, when parsing a xmp buffer into a taglist all tags are
added to a list before being parsed so that when one of the
altitude tags are found the deserialization function can search
for its complementary tag to do the correct parsing
Fixes #613690
2010-03-23 09:48:19 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Fix off by one
Avoid ignoring single char tags, like exif:GPSAltitudeRef
Fixes #613690
2010-03-22 15:18:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
tag: xmp: Adds mappings for LATITUDE and LONGITUDE
Adds the mappings for those tags and tests
for tags serialization.
Fixes #613690
2010-03-22 22:03:09 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Refactor buffer parsing
When parsing the xmp buffer into the gst taglist store the
found tags into a list to be parsed only after finding all
tags on the buffer. This allows the parser function to search
this list for complimentary tags that should be parsed together
Fixes #613690
2010-03-20 11:17:38 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: Refactor mappings storage
This commit is only refactoring, no fetaures added.
Do not store tags in flexible arrays as it doesn't allow us
to use nested flexible arrays. This is going to be needed in the
following commits to map gst tags that are stored into
2 separate tags in xmp (Not that they are alternatives, but
they are complementary).
For example, GST_TAG_ELEVATION is represented in the exif
schema with 2 fields: the absolute altitude and an integer
to indicate if it is above or below sea level.
The previous mappings storage wouldn't allow us to
express it.
Also store a serialization and a deserialization function
for each xmp tag as some of them require some non-trivial
convertion to its string form.
Fixes #613690
2010-03-24 18:51:42 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From 55cd514 to c1d07dd
2010-03-24 18:55:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: parse more info from the buffering query
Parse more info from the buffering query and log this as debug info.
2010-03-24 12:10:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.c:
rtsptransport: ignore unparsable ranges
Ignore unparsable port ranges instead of erroring out.
Fixes #613591
2010-03-23 18:36:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstrtsp.def:
win32: Add new gst_rtsp_lower_trans_get_type() symbol to the symbol lists
2010-03-23 11:01:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: add some more fourcc for MPEG-4 video
2010-03-22 09:15:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: require core git
2010-03-22 08:38:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-fft.pc.in:
pkgconfig: Add @LIBM@ to the FFT pkg-config files
2010-03-22 08:35:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* pkgconfig/gstreamer-app-uninstalled.pc.in:
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
* pkgconfig/gstreamer-floatcast.pc.in:
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
* pkgconfig/gstreamer-video-uninstalled.pc.in:
pkgconfig: Fix include and library paths for the uninstalled pc files
2010-03-20 13:42:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/gio/gstgiobasesrc.c:
gio: add cast to avoid compiler warning with old GLib versions
g_file_input_stream_query_info() had char * instead of const char *
as attribute argument before 2.20.
Fixes #613387, spotted by tetsuyayasuda@gmail.com
2010-03-20 12:55:36 +0000 Torsten Schönfeld <kaffeetisch@gmx.de>
* gst-libs/gst/interfaces/xoverlay.c:
docs: add Since: tags to gst_x_overlay_handle_event() docs
Fixes #613403.
2010-03-19 22:33:58 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
Constify some strings in the API
Needed by plugins-good
2010-03-19 16:41:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Only set color-matrix and chroma-site for relevant formats
The color-matrix only makes sense for colorful formats, i.e. not Y800
and the chroma-site only for non-4:4:4(:4) formats.
2010-03-19 15:37:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoradec.h:
theoradec: add QoS messages to the decoder
Post QoS messages when we drop a frame because of QoS.
2010-03-19 15:00:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtsptransport.h:
rtsp: add GType for transport flags
Make a method to register the transport flags as a GType.
2010-03-19 01:00:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/cdparanoia/Makefile.am:
* ext/gio/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/tcp/Makefile.am:
* gst/videotestsrc/Makefile.am:
* sys/v4l/Makefile.am:
* tests/examples/app/Makefile.am:
* tests/examples/overlay/Makefile.am:
* tests/icles/Makefile.am:
build: Makefile.am fixes
Mostly just add missing $(GST_BASE_CFLAGS), but also fix up order
of flags (see docs/random/moving-plugins).
2010-03-19 00:46:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/pipelines/.gitignore:
.gitignore: ignore new unit test binary
2010-03-17 23:57:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure.ac: -Wmissing-prototypes and -Wnested-externs are not valid for C++
Fixes building Qt-based overlay examples in combination with -Werror.
2010-03-17 16:32:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure.ac: wrap overly long warning flag lines
2010-03-17 19:24:27 -0300 Reuben Dowle <reube.dowle@navico.com>
* sys/ximage/ximagesink.c:
ximagesink: Fix caps leak
Unref caps when peer doesn't accept caps
Fixes #613198
2010-03-17 08:13:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/pipelines/capsfilter-renegotiation.c:
tests: capsfilter-renegotiation: Adds a new unit test
Adds a new test for checking that capsfilter 'caps' property
changes cause caps renegotiation on the pipeline.
2010-03-17 16:46:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_scanline.c:
videoscale: Use correct boundary checks for YUY2/UYVY
Fixes bug #613093.
2010-03-17 16:39:13 +0100 Peter Kjellerstedt <peter.kjellerstedt@axis.com>
* gst-libs/gst/rtsp/gstrtspdefs.c:
rtsp: Further clean up of gst_rtsp_strresult()
Since we no longer use an array of error messages, there is no reason
to clamp the error code, which allows us to simplify the code some more
and also to actually report the correct error code for unknown errors.
2010-03-17 15:41:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: Remove useless cast
It's not necessary anymore after latest core change to GstValueArray.
2010-03-17 12:08:30 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
Add more warning flags
The warnings are:
-Wcast-align
-Winit-self
-Wmissing-include-dirs
-Waddress
-Waggregate-return
-Wno-multichar
-Wnested-externs
No code needed to be fixed.
2010-03-17 11:14:29 +0100 Benjamin Otte <otte@redhat.com>
* gst/audioconvert/gstfastrandom.h:
Fix for -Wold-style-definition
I didn't add the flag to configure because libvisual ships headers that
trigger this warning.
2010-03-17 10:53:21 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
* ext/pango/gstclockoverlay.h:
* gst/subparse/mpl2parse.c:
Add -Wformat-nonliteral -Wformat-security
And fix the resulting compile failures.
I'm sorry about the patch necessary to gstclockoverlay.h but after
talking to Tim we decided we can live with it.
2010-03-17 10:51:57 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/rtsp/gstrtspdefs.c:
rtsp: Refactor gst_rtsp_strresult
2 goals in the refactoring:
- Put the error messages closer to their enum values, so that it's easy
to see which error belongs to which value.
- Make gcc not complain with -Wformat-nonliteral
2010-03-17 10:47:07 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/tag/gstxmptag.c:
xmp: Refactor code
I initially looked here because I wanted compiles to not fail with
-Wformat-nonliteral but ended up refactoring the code to make it look
nicer.
As I lack a large collection of XMP tagged files, I only did rough
testing of the code. The testsuite passes though.
2010-03-16 20:05:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* m4/Makefile.am:
* m4/a52.m4:
* m4/aalib.m4:
* m4/as-arts.m4:
* m4/as-ffmpeg.m4:
* m4/as-liblame.m4:
* m4/as-slurp-ffmpeg.m4:
* m4/esd.m4:
* m4/gconf-2.m4:
* m4/glib.m4:
* m4/gst-artsc.m4:
* m4/gst-matroska.m4:
* m4/gst-sdl.m4:
* m4/gst-shout2.m4:
* m4/gst-sid.m4:
* m4/gtk.m4:
* m4/libfame.m4:
* m4/libmikmod.m4:
m4: remove some unused .m4 files
2010-03-16 18:31:15 +0100 Benjamin Otte <otte@redhat.com>
* ext/alsa/gstalsaplugin.c:
* ext/ogg/gstoggdemux.c:
More ENABLE_NLS fixes
2010-03-16 18:06:16 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/gettext.h:
Fix for ENABLE_NLS being undefined for -Wundef
2010-03-15 22:49:53 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
* ext/libvisual/visual.c:
* ext/theora/gsttheoraenc.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
* gst-libs/gst/tag/tags.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* gst/typefind/gsttypefindfunctions.c:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lelement.c:
* sys/xvimage/xvimagesink.c:
* tests/check/elements/audioconvert.c:
* tests/check/elements/gdpdepay.c:
* tests/check/elements/playbin.c:
* tests/check/elements/playbin2.c:
* tests/check/elements/videorate.c:
* tests/check/libs/pbutils.c:
* tests/check/libs/video.c:
* tests/check/pipelines/simple-launch-lines.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
* tests/icles/stress-playbin.c:
Add -Wwrite-strings to configure
Fixes for the code included
2010-03-16 15:45:23 +0100 Benjamin Otte <otte@redhat.com>
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdeclib.h:
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* ext/vorbis/gstvorbistag.c:
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gststreamselector.c:
* gst/playback/gsturidecodebin.c:
* gst/subparse/gstssaparse.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/elements/audioconvert.c:
* tests/check/elements/playbin.c:
* tests/check/elements/playbin2.c:
* tests/check/elements/textoverlay.c:
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/pbutils.c:
* tests/old/testsuite/alsa/formats.c:
* tests/old/testsuite/alsa/sinesrc.c:
gst_element_class_set_details => gst_element_class_set_details_simple
Also change my email from the old university one to the current one.
2010-03-15 22:17:56 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
Add -Wundef flag
2010-03-16 16:15:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: allow for more ipv6 addresses
Use hints in getaddrinfo() so that we can also resolve ipv6 addresses.
2010-03-11 14:52:09 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: arrange for a running ringbuffer/clock for _wait_eos
Fixes #612223.
2010-03-16 01:08:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videorate.c:
tests: fix videorate test
Fix up videorate test for latest videotestsrc changes: just check for
the important bits in the negotiated caps, not for exact equality with
our filter caps. Also don't leak the videorate element in the test.
2010-03-15 12:54:32 -0500 Rob Clark <rob@ti.com>
* gst-libs/gst/riff/riff-media.c:
riff: add mapping for On2 VP7 fourccs
Fixes #612968.
2010-03-15 12:54:01 -0500 Rob Clark <rob@ti.com>
* gst-libs/gst/riff/riff-media.c:
riff: add mapping for On2 VP62 fourcc
See #612968.
2010-03-15 23:46:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/multichannel.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtsp/gstrtsptransport.h:
docs: more helper libraries docs fixes
Quieten gtk-doc a bit more.
2010-03-15 23:47:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspextension.c:
docs: add GstRTSPExtension to docs
Add minimal docs for GstRTSPExtension so people know it exists.
2010-03-15 18:45:13 +0000 David Hoyt <dhoyt@llnl.gov>
* gst/typefind/gsttypefindfunctions.c:
typefind: use g_ascii_strncasecmp() instead of strncasecmp()
g_ascii_strncasecmp() is more portable and likely more robust as
well (with random binary data as input).
Fixes #612845.
2010-03-15 13:39:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
docs: fix typo in gst_tag_list_from_xmp_buffer() docs chunk
2010-03-15 13:32:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/navigation.c:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
docs: fix up interfaces library docs to make gtk-doc happy
2010-03-15 13:24:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: add new libgstvideo API to documentation
2010-03-15 13:19:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstinterfaces.def:
* win32/common/libgstvideo.def:
win32: add recently added API to .def files
Also add API markers to make life easier for the release manager:
API: gst_x_overlay_set_render_rectangle()
API: gst_video_parse_caps_color_matrix()
API: gst_video_parse_caps_chroma_site()
2010-03-15 13:14:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: use C comments instead of C++-style comments
2010-03-15 13:10:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: use g_value_set_static_string() for string constants
2010-03-15 14:26:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Avoid g_object_set() on NULL if a text sink is used
Fixes bug #611702.
2010-03-15 14:10:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Correctly escape brackets in DKS regex
Fixes bug #612783.
2010-03-15 11:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: make timeout usec more accurate
Adjust the returned usec from the elapsed time so it represents the remaining
timeout.
2010-03-15 11:41:35 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/check/elements/videorate.c:
tests: update videorate test for videotestsrc changes
Add color-matrix to the caps we are comparing. Add logging og the caps in the
test.
2010-03-15 01:35:15 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: add chroma-zone-plate pattern
pattern=chroma-zone-plate is pattern similar to zone-plate,
but in the chroma channels instead of luma.
2010-03-15 01:34:09 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoradec.c:
theoradec: add chroma-site to caps
2010-03-15 01:33:36 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: add chroma-site to caps
2010-03-15 01:31:20 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: add gst_video_parse_caps_chroma_site()
2010-03-14 19:10:16 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoradec.c:
theoradec: add color-matrix to caps
2010-03-14 16:17:46 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add color-matrix to template caps
2010-03-14 22:14:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/seek/seek.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
tests: make Gtk+ test programs compile with -DGSEAL_ENABLE
Fixes #612552, at least for now.
2010-03-14 22:13:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* Makefile.am:
build: add cruft alert for common/shave* leftovers to top-level Makefile.am
2010-03-14 13:11:53 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: Don't drop zero-sized packets
Zero-sized packets have relevence to Theora.
2010-03-12 15:47:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: Revert rounding behaviour changes when using controlled volume properties
Now the controlled and non-controlled code paths are all having
exactly the same rounding behaviour and the unit tests pass again.
2010-03-12 15:44:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: Only allocate a mute value array if a control source exists for the mute property
2010-03-12 13:55:55 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From e272f71 to 55cd514
2010-03-10 10:50:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst-libs/gst/tag/gstxmptag.c:
tags: Add new mapping to XMP helpers
Adds geotagging mappings to XMP helpers
Fixes #609539
2010-03-11 20:16:44 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/interfaces/Makefile.am:
Don't have 2 include dirs
Seems to have been accidentally introduced in
7269bc26d0a4bf44bd77a039fb54777625ef5f39.
2010-03-11 16:35:10 +0100 Edward Hervey <bilboed@bilboed.com>
* tests/icles/audio-trickplay.c:
tests: Fix another unitialized variable
2010-03-11 16:09:26 +0100 Edward Hervey <bilboed@bilboed.com>
* tests/icles/audio-trickplay.c:
tests: Fix unitialized variable.
2010-03-11 15:38:18 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
* ext/ogg/gstoggdemux.c:
* ext/theora/gsttheoraparse.c:
* ext/vorbis/gstvorbistag.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.h:
* gst/audioconvert/gstchannelmix.h:
* gst/playback/gstplaysink.c:
Add -Wredundant-decls to warning flags
... and fix all the warnings that flag throws.
2010-03-11 13:32:14 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c:
* ext/ogg/gstogg.h:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstogmparse.c:
* ext/ogg/vorbis_parse.c:
* ext/ogg/vorbis_parse.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/riff/riff.c:
* gst-libs/gst/rtsp/gstrtspbase64.c:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/tag/lang.c:
* gst/ffmpegcolorspace/Makefile.am:
* gst/ffmpegcolorspace/gstffmpeg.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplayback.c:
* gst/playback/gstplayback.h:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/videorate/gstvideorate.h:
* tests/check/elements/appsink.c:
* tests/check/elements/audiorate.c:
* tests/check/elements/audioresample.c:
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/mixer.c:
* tests/check/libs/navigation.c:
* tests/examples/gio/giosrc-mounting.c:
Add -Wmissing-declarations -Wmissing-prototypes to warning flags
Includes all the fixes necessary to make stuff compile again.
2010-03-11 12:49:02 +0100 Benjamin Otte <otte@redhat.com>
* ext/gio/gstgiobasesink.c:
gio: Remove unused function
2010-03-11 11:14:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/vorbis/gstvorbisparse.c:
vorbisparse: make sure header buffer metadata is writable before modifying it
Fixes unit test failures with core git.
2010-03-11 12:18:00 +0100 Benjamin Otte <otte@redhat.com>
* tests/check/elements/multifdsink.c:
check: Ref buffers after setting caps on them
Reffing makes metadata unwritable, so we need to set the caps before.
2010-03-11 12:04:32 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
Add WARNING_CXXFLAGS where ERROR_CXXFLAGS are
This matches the previous commit doing the same for CFLAGS in response
to the common/ module changes.
2010-03-11 12:04:37 +0100 Edward Hervey <bilboed@bilboed.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2010-03-11 10:38:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/icles/test-xoverlay.c:
tests: don't use Gtk+ 2.18 API for no good reason
The rest of the code directly uses widget->allocation as well, so no point
in using the new API in other places.
2010-03-11 11:20:48 +0100 Benjamin Otte <otte@redhat.com>
* common:
Automatic update of common submodule
From df8a7c8 to e272f71
2010-03-11 10:55:21 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/interfaces/xoverlay.c:
xvoverlay: correct version number in docs
2010-02-26 13:56:21 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/audio-trickplay.c:
tests: add a test for trickplay in audio synthesis graphs
Right now this mostly demonstatest what not works. That is seeking with
start-type = NONE to only update the rate and playing backwards. Also
it shows that non-flushing seeks tend to lockup adder. Separate unit tests
for the issues follow.
2010-02-08 17:20:35 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstxmptag.c:
* gst-libs/gst/tag/tag.h:
* tests/check/libs/tag.c:
* win32/common/libgsttag.def:
tags: add basic xmp metadata support
XMP metadata can be embedded in many media container formats. Implement own
parser and formatter that can be used to convert between an xpacket and a
GstTagList. Add unit tests.
2010-02-19 14:38:36 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/test-xoverlay.c:
example: add an example for xoverlay::set_render_rectangle()
This add a new example which animates a target recangle for the video.
2010-02-19 14:46:43 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
xvimagesink: implement set_render_rectangle
Previously we hardcoded the target rectangle passes to Xv(Shm)PutImage. Extend
the implementation to use a full rectangle and don't assume 0,0 for top,left.
2010-02-17 15:00:13 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
xoverlay: add new vmethod ::set_render_rectangle()
Add set_render_rectangle() vmethod to the interface to better support windowless
toolkits (e.g. qt graphicsview or video on canvas in general). Right now we
always fill the widget to 100%. With the patch we can use a rectangular target
region. Fixes #610249.
API: GstXOverlay::set_render_rectangle()
2010-02-16 12:06:08 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: take new size from event thread and do not poll for every frame
We can update the geometry in ConfigureNotify (unless we disable event-
handling). If event handling is disabled, one should use _expose() to trigger a
redraw and update the geometry.
2010-03-10 21:51:59 +0100 Benjamin Otte <otte@redhat.com>
* common:
Automatic update of common submodule
From 9720a7d to df8a7c8
2010-03-10 21:01:20 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
Update for recent changes to common submodule
This just replaces every "$ERROR_CFLAGS" usage with a usage of
"$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as
previously.
Actually using that separation will happen later.
2010-03-10 20:43:46 +0100 Benjamin Otte <otte@redhat.com>
* common:
Automatic update of common submodule
From 0b6e072 to 9720a7d
2010-03-10 16:09:45 +0100 Benjamin Otte <otte@redhat.com>
* common:
Automatic update of common submodule
From 7cc5eb4 to 0b6e072
2010-03-10 14:36:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: do not cache FLUSH_START/_STOP events
... and similarly so for serialized events.
2010-03-10 14:34:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: provide correct error message if configured audio/video sink fails
2010-03-10 10:22:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisdec.h:
vorbisdec: remove unused field
2010-02-02 11:34:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/check/pipelines/vorbisdec.c:
tests: enable strict discontinuity checking on vorbisdec pipeline
Closes #423086.
2010-03-10 01:09:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 7aa65b5 to 7cc5eb4
2010-03-10 01:07:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/video.c:
docs: fix Returns: for gst_video_parse_caps_color_matrix()
2010-03-10 00:46:34 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update for changed string
2010-03-10 00:42:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videorate.c:
tests: fix typo in videorate unit test pipeline description
Two consecutive ! ! leave a 'Link without source' error in the debug log.
2010-03-10 00:41:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/videorate.c:
tests: don't use deprecated functions in videorate unit test
2010-03-10 00:29:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/libgstvideo.def:
win32: add new API to libgstvideo.def
2010-03-09 15:39:55 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggmux.c:
oggmux: Don't flush after every frame for theora
2010-03-09 21:26:58 +0000 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 44ecce7 to 7aa65b5
2010-03-09 13:05:23 -0800 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add color-matrix handling to caps
2010-01-30 22:55:01 -0800 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: Add color-matrix to caps
2010-02-26 16:25:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* pkgconfig/Makefile.am:
* tests/examples/overlay/Makefile.am:
* tools/Makefile.am:
build: Make some more rules silent if requested
2010-02-26 15:40:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
configure: Use automake 1.11 silent rules instead of shave if available
This makes sure that we use something that is still maintained and
also brings back libtool 1.5 support.
2010-02-23 19:12:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Don't fail if there are subtitles and audio but no video
Change playbin2 to not error out if there are subtitles and audio
but no video. If visualizations are enabled the subtitles are rendered on top
of the visualization stream, otherwise the subtitles are not linked at all and
only the audio is played (and a warning message is posted).
If there are only subtitles but neither audio nor video an error message is
still posted.
Fixes bug #610866.
2010-02-17 19:18:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
volume: If a controller is used, use sample accurate property values
Fixes bug #609801.
2010-03-09 19:17:04 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/video/video.c:
gstvideo: Fix typos in comments
2010-03-09 17:32:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development
=== release 0.10.28 ===
2010-03-08 23:20:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.28
2010-03-08 23:19:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2010-03-08 21:57:03 +0100 Benjamin Otte <otte@redhat.com>
* ext/theora/gsttheoraenc.c:
theora: Fix SIGFPE when using 0/1 framerate
libtheora crashes with a 0 framerate, so let's forbid it.
https://bugzilla.redhat.com/show_bug.cgi?id=571289
2010-03-08 14:50:25 +0000 David Schleef <ds@schleef.org>
* ext/ogg/dirac_parse.c:
oggdemux: fix dirac header parsing
Fixes #611900.
2010-03-08 14:46:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/overlay/Makefile.am:
examples: make sure to dist qtgv-xoverlay.h header file
This time for real.
Fixes #610832.
2010-03-08 12:11:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
basedepay: clarify some documentation
2010-03-08 11:25:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
alsasrc: return right number of bytes that we wrote
2010-03-08 11:20:51 +0100 Dake Gu <gudake@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: fix handling of x-server-ip-address
Fix handling of x-server-ip-address.
2010-03-02 11:25:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/design/draft-keyframe-force.txt:
docs: update keyframe force event
Add field to send all headers.
=== release 0.10.27 ===
2010-03-06 00:09:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.27
2010-03-06 00:08:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2010-03-05 15:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: first check for QtGui >= 4.6, only then for >= 4.0
If we first check for >= 4.0 the second check for >= 4.6 will just
short-cut since we are using the same prefix for the variables for
both checks, and they've already been set previously. So the examples
requiring >= 4.6 were built even in the >= 4.0 case.
2010-03-03 20:18:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.26.4 pre-release
2010-03-03 20:17:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/ja.po:
po: update translations
2010-03-03 20:15:44 +0000 Josep Torra Valles <n770galaxy@gmail.com>
* gst/playback/gstplaysink.c:
playsink: avoid g_object_set() on NULL pointers
There may not be an overlay element if a text-sink is set.
Fixes #611702.
2010-03-01 12:17:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: mark skeleton streams correctly
Mark skeleton streams because we need to ignore them for calculating the
duration of the stream.
Fixes #611227
2010-02-24 01:10:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* po/nl.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.26.3 pre-release
2010-02-23 16:57:53 +0100 Götz Waschk <waschk@mandriva.org>
* tests/examples/overlay/Makefile.am:
examples: Dist header file for the Qt graphics view example
Fixes bug #610832.
2010-02-23 11:41:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: use the chain begin_time instead of our counter
We update the passed begintime argument to narrow our search region in the
binary search. This means that it does not always contain the chain begin time
after a couple of bisects. Use the real chain->begin_time to bring the
granuletime to the time in the chain instead.
Fixes #610005
2010-02-19 18:24:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* tests/check/elements/videorate.c:
videorate: tests: New unit tests for upstream caps nego
Adds unit tests that check videorate's upstream caps
negotiation works properly (put passthrough caps
first)
Fixes #608025
2010-01-27 15:07:47 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Improve upstream negotiation
Put peer pad caps preferred framerates first, indicating
they are videorate's first choices, removing an unnecessary
conversion.
Fixes #608025
2010-02-21 19:52:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
playbin2, playsink, subtitleoverlay: Set subtitle encoding properly
For this add subtitle encoding properties to playsink and subtitleoverlay
and update the values in the containing elements.
Also update the font description in textoverlay or the used renderer
element if it is changed during playback.
Fixes bug #610310.
2010-02-22 13:01:19 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.cpp:
examples: also add sink detection and set title to qt examples
Also set a title in the qt examples like it is now done in the gtk example.
Fix the newly added find_video_sink in the gtk example and add similar function
to the qt examples.
2010-02-19 14:40:43 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/examples/overlay/.gitignore:
gitignore: ignore files in new example directroy
2010-02-17 14:59:33 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/Makefile.am:
make: fix copy and paste error in git rules (audio<->video)
2010-02-19 17:44:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Ghost the video sinkpad if a text sinkpad is available
Only don't ghost it if no visualizations are need and if
no text is needed and no textchain was created yet.
Fixes bug #610379.
2010-02-19 00:22:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.26.2 pre-release
2010-02-19 00:20:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translation files
2010-02-19 00:17:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/overlay/.gitignore:
Ignore new overlay examples
2010-02-18 23:47:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/overlay/gtk-xoverlay.c:
examples: don't hard-code xvimagesink for Gtk+ GstXOverlay example
Try to find a working videosink, don't hardcode xvimagesink. Also
add some borders to window and give it a title so that it's clear
that this is really a Gtk+ window and not a window created by the
videosink.
2010-02-18 11:42:55 -0800 David Schleef <ds@schleef.org>
* gst/tcp/gsttcp.c:
tcp(client/server)src: Fix handling of closed sockets
The peer closing the socket should cause an EOS, instead of
silently doing nothing. This changes the behavior to be
more like fdsrc. Fixes: #610386
2010-02-18 12:42:53 +0000 Patrick Radizi <patrick.radizi@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: make sure not to dereference NULL username or password
Fixes #610268.
2010-02-17 21:22:54 -0800 David Schleef <ds@schleef.org>
* ext/theora/gsttheoradec.c:
theoradec: Fix chroma copying for 4:2:2
Fix mixup of height/width, causing only half the chroma lines to
be copied when outputting buffers. Fixes: #610329.
2010-02-16 15:43:26 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* gst-libs/gst/interfaces/xoverlay.c:
* tests/examples/Makefile.am:
* tests/examples/overlay/Makefile.am:
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.h:
examples: add video overlay examples for gtk, qt and qt graphics view
Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
adds all boilerplate to configure for using c++. The qt based examples are
optional like their gtk counterparts.
2010-02-16 17:20:01 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/compiling.sgml:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: cleanup library docs
Correct name of included files. Remove files that are not used anymore. Add many
new api entries to their sections.
2010-02-15 11:11:04 +0200 Stefan Kost <ensonic@users.sf.net>
* tests/icles/test-colorkey.c:
test-colorkey: remove the XInitThreads()
We don't do this is any other example, this should be done for us in gdk it if
would be needed.
2010-02-16 10:09:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: use same message string for missing elements as in playbin
Use the same translated message string for missing core elements as
playbin uses, which is a bit nicer and also indicates that there is
something wrong with the user's GStreamer installation (which arguably
is the case if elements like typefind or queue2 are missing).
2010-02-08 13:54:57 +0200 Kaj-Michael Lang <milang@tal.org>
* gst/typefind/gsttypefindfunctions.c:
typefind: Handle stm module format
Fixes #609314.
2010-02-15 12:10:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstivorbisdec.c:
ivorbisdec: set rank to SECONDARY
2010-02-15 12:09:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* configure.ac:
* ext/Makefile.am:
* ext/vorbis/Makefile.am:
* ext/vorbis/gstivorbisdec.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/gstvorbisdeclib.c:
* ext/vorbis/gstvorbisdeclib.h:
vorbisdec: also support ivorbis tremor decoder
... which only needs a bit of refactoring and extracting to support
the minor difference in (i)vorbis interface.
Fixes #609063.
2010-02-03 14:37:43 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
vorbisdec: reduce some hard-coding
... such as assuming float all over, and base src caps on template caps.
2010-02-15 10:23:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/playbin.c:
playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2
2010-02-15 09:04:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggparse.c:
oggparse: Fix another format string compiler warning
2010-02-15 08:56:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Fix format string compiler warnings
2010-02-15 08:48:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Post a missing element message and an error message if no uridecodebin can be found
2010-02-15 08:46:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Post missing element messages if a core plugin is missing
And post a warning in cases where we can still continue to work
or an error when the missing element is fatal.
2010-02-15 08:28:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/playbin2.c:
playbin2: Enable all unit tests
They're all working and valgrind clean now.
2010-02-15 08:26:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: First post a missing-plugin message, then emit the unkown-type signal
This makes sure that there *always* is a missing plugin message in the bus
before any errors or warning messages.
2010-02-15 08:20:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Missing decoder errors should be STREAM CODEC_NOT_FOUND
and not CORE MISSING_PLUGIN.
2010-02-15 08:18:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Free the subtitle URI
2010-02-15 08:06:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Post missing plugin messages if a required element can't be created
Especially if no suitable URI source can be found.
2010-02-15 06:50:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/.gitignore:
tests: Add decodebin2 test to .gitignore
2010-02-15 01:18:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Set ghostpad targets to NULL when freeing a decode chain
Otherwise the ghostpad will still be linked to the peer and there
will still be a reference kept, leading to nothing being unlinked
and destroyed until decodebin2 is finalized.
This fixes reuse of decodebin2 if a raw stream is connected to
its sinkpad.
2010-02-15 01:17:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/elements/decodebin2.c:
decodebin2: Add simple unit test, mainly a copy of the decodebin unit test
The only difference between the two unit tests right now is,
that the decodebin2 test resets the element to READY before trying
to reuse it instead of NULL. decodebin2 guarantees to be reusable
without going back to NULL.
2010-02-15 00:11:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
See #609252.
2010-02-14 23:16:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 96dc793 to 44ecce7
2010-02-14 23:10:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/Makefile.am:
playbin2: Enable playbin2 unit test
It now contains a single working unit test and can be enabled.
The other more useful unit tests still need fixing.
2010-02-14 22:16:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/playbin.c:
playbin: Fix indention in the unit test
2010-02-13 01:08:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
volume: Replace this variables by self
2010-02-12 19:43:13 +0100 Josep Torra Valles <n770galaxy@gmail.com>
* gst/playback/gstplaysink.c:
playsink: Reset the sink's state to NULL before unreffing it unless it's the same instance again
This makes sure that we don't destroy the last reference before the
element gets back to NULL state. Fixes assertion failures if a playbin2
instance is reused but different sinks are automatically chosen because
of different caps.
2010-02-12 18:00:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: fix Since tag
2010-02-12 14:19:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/riff/riff-read.c:
riff: treat JUNQ chunks like JUNK chunks
2010-02-12 14:29:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Update basesrc segment duration and post duration messages from the streaming thread
2010-02-11 14:10:02 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/tags.c:
tags: improve docs about determining the encoding
2010-02-11 14:09:05 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/tag/gstvorbistag.c:
comment: fix wrong header comment
2010-02-01 13:50:14 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/riff/riff-ids.h:
riff: add a variant of the JUNK tag that several adobe products produce
JUNQ has same semantics as JUNK.
2010-02-01 19:01:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: add min-percent property
Emit need-data when the amount of data in the internal queue drops below
min-percent.
Fixes #608309
2010-02-01 18:56:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: cleanups
Avoid some typechecks.
Avoid dereferencing appsrc->priv all the time.
2010-02-01 18:55:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
appsink: cleanups
Avoid some typecasting.
Avoid dereferencing appsink->priv all the time.
2010-02-01 15:09:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: avoid some typecasts
2010-01-29 16:34:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: ignore \n and \r as the first line
Be more forgiving for bad servers and ignore \r and \n when we are looking for
the response/request line.
See #608417
2010-02-10 16:05:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: fail gracefully on bad Content-Length headers
Be careful when allocating the amount of bytes specified in the Content-Length
because it can be an insanely huge value. Try to allocate the memory but fail
gracefully with a nice error when the allocation failed.
2010-02-10 10:12:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Add conversions from all ARGB formats to AYUV and back
2010-02-09 17:39:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Update segment duration and post a duration message if the duration changes
Fixes bug #609423.
2010-02-11 10:56:17 +0100 Benjamin Otte <otte@redhat.com>
* tests/examples/seek/Makefile.am:
build: link to libm in examples that use it
This fixes build failure in Fedora 13.
2010-02-11 01:11:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* MAINTAINERS:
Update MAINTAINERS, add myself
2010-02-11 23:57:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: back to development
Slushy freeze remains in effect.
=== release 0.10.26 ===
2010-02-10 20:17:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.26
2010-02-10 20:16:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2010-02-08 11:21:35 +0100 Benjamin M. Schwartz <bens@alum.mit.edu>
* ext/theora/gsttheoradec.c:
theoradec: PARs of 0:x, x:0 and 0:0 are all allowed and map to 1:1
Fixes #609252.
2010-01-24 12:31:04 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
* ext/ogg/gstoggstream.c:
oggdemux: use the default granpos functions for kate streams
Set timestamps on kate packets. See bug #600929.
2010-02-05 01:18:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.25.3 pre-release
2010-02-04 18:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/bg.po:
po: update translations
2010-02-04 18:32:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
Revert "playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler"
This reverts commit 7335ce5d3e03c126a417a721571cb6f3af136ecf.
Support abusing the uri property to configure the next uri to play
outside of the about-to-finish handler for the time being after all.
We also shouldn't use thread private structures for this, since it
should be possible to block the thread that emitted about-to-finish
while the main thread sets the uri property. See #607226.
2010-02-02 10:18:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Don't leak allocated buffers
This can happen if the combined flow return is not OK although the
allocation succeeded or if the packet in question is a BOS and we're
not going to push headers.
Fixes bug #608699.
2010-02-01 11:44:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: clean up decodebin properties
When reusing a decodebin2 element, clear the properties we might have changed,
to their default values or else we might end up with old configuration.
Fixes #608484
2010-01-29 13:56:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: when no uri is set, post an error message
When no uri is set, don't just return STATE_CHANGE_FAILURE from the
state change function, but actually post an error message.
2010-01-30 15:18:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 15d47a6 to 96dc793
2010-01-28 17:12:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
adder: don't hold object lock when calling peer elements
Do not hold the object lock while we call methods on peer elements as this can
lead to deadlocks.
Fixes #608179
2010-01-27 01:12:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
0.10.25.2 pre-release
2010-01-27 01:07:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/interfaces-enumtypes.c:
* win32/common/interfaces-enumtypes.h:
* win32/common/pbutils-enumtypes.c:
* win32/common/video-enumtypes.c:
win32: update generated files for non-autotools win32 builds
2010-01-27 00:56:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translation files
2010-01-27 00:41:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosrc.c:
audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
2010-01-26 16:47:40 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't skip an element when getting the topology
Fixes #608167
2010-01-24 14:41:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
* ext/ogg/gstoggdemux.c:
oggdemux: sparse streams aren't timed by end time, and their duration isn't implicit
Fixes timestamps and durations on Kate subtitle streams.
See http://www.xiph.org/ogg/doc/ogg-multiplex.html section 'start-time and
end-time positioning' for some more details, and bug #600929.
2010-01-23 20:15:08 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
* ext/ogg/gstoggstream.c:
oggdemux: properly set up the media type for kate streams
See #600929.
2010-01-25 18:57:52 +0100 Julien Moutte <julien@fluendo.com>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: relax caps template on sink pads
Allow any caps on sink pad templates as we could do passthrough with non raw
video caps.
2010-01-25 15:14:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.h:
oggdemux: use right type for the serialno
Use a consistent type for the serialno to avoid problems when comparing between
signed and unsigned variants.
Fixes #607926
2010-01-25 14:00:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: don't push headers twice
Don't push the stream headers twice but only in the activation of a chain.
Fixes #607929
2010-01-25 13:18:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
2010-01-25 12:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: rename a variable
Rename the 'seekable' variable to 'pullmode'. We might be able to seek in push
mode too eventually.
2010-01-25 12:22:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
Revert "inputselector: Protect g_object_notify() with the object's mutex"
This reverts commit a37426c41c80fd21e5017fea01a786c05bcd9661, it's
causing deadlocks with playbin2.
2010-01-24 20:55:26 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
* gst/playback/gstinputselector.c:
inputselector: Protect g_object_notify() with the object's mutex
This works around the thread unsafety of g_object_notify()
Fixes bug #607513.
2010-01-24 20:46:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add typefinder for ISO MP4 files
Fixes bug #607848.
2010-01-24 13:29:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix crash when freeing headers
Use _ogg_packet_free() instead of gst_mini_object_unref in one more
place now that the header list contains ogg packets and not buffers.
file: Stephen_Fry-Happy_Birthday_GNU-nq_600px_425kbit.ogv
2010-01-24 08:57:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Strip trailing \0 for subtitle OGM streams
Fixes bug #607870.
2010-01-23 22:09:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Correctly set DELTA_UNIT flag for OGM streams
2010-01-23 22:05:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Don't strip all 0-bytes from the end of OGM packets
This fixes broken packets pushed downstream by oggdemux for
MPEG4 streams for example.
2010-01-23 22:03:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Extract tags from OGM text streams and don't push them downstream
2010-01-23 14:46:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream
2010-01-23 15:25:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: optimise AC-3 typefinder a bit
Make AC-3 typefinder use the DataScanCtx stuff so we don't have to
do gst_type_find_peek() in the inner loop all the time. Also return
when we've suggested AC3 caps, instead of continuing with the loop.
2010-01-23 14:31:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
Revert "typefind: Reduce number of calls to gst_type_find_peek."
This reverts commit c661bfaa991c58f1fbd9fbc0dae90b8b2c27f92b.
This breaks AC-3 typefinding for all cases where the first frame
is at an offset > 0.
2010-01-23 15:35:05 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for Zip Block Motion Video
2010-01-23 15:34:54 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add mapping for Zip Block Motion Video
2010-01-23 15:26:37 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: YUNV is a fourcc which is also used for YUY2 raw video
2010-01-23 15:13:45 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: vp61 and VP61 are also valid On2 VP6 fourcc
2010-01-23 15:10:45 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add mapping for On2 VP5
2010-01-23 15:04:35 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add mapping for Sigma-Designs MPEG4
It's actually a xvid-compatible stream. both xviddec and ffmpeg handle it.
2010-01-23 14:35:28 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for LOCO Lossless codec
2010-01-23 14:35:16 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add mapping for LOCO Lossless codec
2010-01-23 14:08:39 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
2010-01-23 13:50:26 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for Autodesk Animator codec
2010-01-23 13:50:09 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add mapping for Autodesk Animator Codec
2010-01-23 13:20:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: ...and set caps on queued packet buffers too
2010-01-23 13:19:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Set caps on header buffers
2010-01-22 16:23:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: handle raw sources about-to-finish signals
When we are dealing with a source that produces raw audio/video, we don't use a
decodebin2 to decode the data and we thus don't have the drained/about-to-finish
signal emited. To fix this, we add a padprobe on the source pads and emit the
drained signal ourselves. This then makes playbin2 emit the about-to-finish
signal for raw sources such as cdda://
Fixes #607116
2010-01-22 16:15:54 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/typefind/gsttypefindfunctions.c:
typefind: include stdio.h for sscanf
2010-01-22 01:49:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: add PNM typefinder
Add PNM typefinder, so we can remove the one that's in the PNM plugin
in -bad (which btw uses different/wrong media types that don't match
the ones used by gdkpixbufdec) and people don't make fun of us for
loading image decoders when typefinding and playing back audio files.
2010-01-21 19:31:23 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: rename performance category
rename the performance category to ffmpegcolorspace_performance
as there is already a global GST_CAT_PERFORMANCE in core
2010-01-21 17:32:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: keep track of added pads
Keep track of the pads we added and removed.
Remove some unused fields.
Don't add pads for which we don't have caps.
2010-01-21 17:31:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: don't call NULL setup functions
If we find a known mapper but it doesn't have a setup function, simply skip it
instead of crashing.
2010-01-21 17:30:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: avoid division by 0 on bad annodex streams
2010-01-21 13:47:01 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for y4m container
2010-01-19 14:31:34 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppayload: ptime/maxptime should be unsigned
https://bugzilla.gnome.org/show_bug.cgi?id=607403
2010-01-18 21:16:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
basertppayload: ptime should be in nanoseconds
https://bugzilla.gnome.org/show_bug.cgi?id=607403
2010-01-20 00:53:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 14cec89 to 15d47a6
2010-01-19 13:33:06 -0800 David Schleef <ds@schleef.org>
* gst/typefind/gsttypefindfunctions.c:
typefind: rewrite h.264 detection
Make detection simpler: check for NALs, check that they make
sense, and report how certain we are that it's a raw H.264 stream.
Fixes: #583376.
2010-01-18 14:33:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppayload: Reject empty caps
https://bugzilla.gnome.org/show_bug.cgi?id=607353
2010-01-19 08:39:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: No need to subtract begin time
Last stop is already based on the chain start and there is no need
to subtract the chain start as it may lead to a negative overflow.
This was causing seeking issues when the target chain was not
the first one (that has chain start = 0)
Fixes #606382
2010-01-19 09:25:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/audio.h:
audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
Fixes bug #607381.
2010-01-18 15:22:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: granulepos is relative to its chain
When performing seeks, the granulepos should be offset by
its chain start time to avoid using wrong values to
update segment's last_stop. A sample file is indicated on
bug #606382
2010-01-18 17:57:16 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for MXF container format
2010-01-18 10:07:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: re-use iterator callback to avoid code duplication
2010-01-18 02:08:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: when looking for sink properties, make sure they have the right type
We don't want to end up setting values on elements where the property is of
a different type than we expect. Can't transform the value either, since we
can't really make assumptions about the scale and transform function.
Fixes crashes when using playbin2 with apexsink (#606949).
2010-01-18 09:30:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler
Changing the URIs in a state > READY results in unexpected behaviour,
i.e. the new URIs are only used after the current track has finished.
Fixes bug #607226.
2010-01-15 19:52:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: sprinkle some more locking
... to avoid races and ensure some data structure consistency.
See also #574289.
2010-01-14 18:26:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: mind blocked pads when shutting down
Fix regression in shutdown deadlock handling now that the
target of a ghostpad is blocked instead of ghostpad itself.
See also #574293.
2010-01-14 13:36:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Fix disabling of subtitles if subtitles were used before
In this case the video still goes through the text chain and
subtitles are still going in there, in case subtitles are
enabled again. This makes sure that re-enabling subtitles
happens instantly.
Fixes hanging video when disabling subtitles, caused by an
unliked video pad.
2010-01-14 10:43:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: fix pad ref leak
2010-01-12 21:42:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* docs/plugins/Makefile.am:
docs: fix out-of-source build
2009-04-29 11:50:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/icles/stress-playbin.c:
stress-playbin: fix error return check
2010-01-14 10:10:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/theora/Makefile.am:
* ext/theora/gsttheora.c:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
* ext/theora/theora.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
theora: Rename source files to have the same name as the headers
2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbis.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* ext/vorbis/gstvorbistag.c:
* ext/vorbis/vorbis.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
vorbis: Rename source files to have the same name as the headers
2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbiscommon.c:
* ext/vorbis/gstvorbiscommon.h:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
vorbis: Move channel layout definitions into a single separate file
...instead of having two copies.
2010-01-14 08:19:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
vorbis: Add official 6.1 and 7.1 channel mappings
These are in the Vorbis spec since 2010-01-13. Fixes bug #606926.
2010-01-13 23:05:45 +0100 Benjamin Otte <otte@redhat.com>
* gst-libs/gst/rtsp/gstrtspdefs.c:
rtsp: Don't define h_error ourselves
It's included from netdb.h and that header might define it differently,
which can lead to build failures.
2010-01-13 17:36:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: mp4 video is not parsed
2010-01-13 12:49:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: Add aac stream-format to caps
Also add the aac stream-format field on the caps when
detecting it.
2010-01-13 09:39:54 +0100 Brijesh Singh <brijesh.ksingh@gmail.com>
* gst/playback/gstplaysink.c:
playsink: Fix handling of the native audio/video flags
Fixes bug #606687.
2010-01-12 16:35:50 +0100 Edward Hervey <bilboed@bilboed.com>
* ext/ogg/gstoggdemux.c:
oggdemux: Fix unitialized variable.
If the package isn't handled, gracefully return GST_FLOW_OK.
2010-01-10 23:50:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/xoverlay.c:
docs: flesh out GtkXOverlay docs some more and add example for Gtk+ >= 2.18
Explain why the whole bus sync handler mess is needed. Add section about
how to use GstXOverlay in connection with Gtk+ and mention the Gtk+ API
break issue and how to work around it (see #601809).
2010-01-10 21:18:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
docs: minor netbuffer documentation fix
2010-01-10 20:41:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: update translated strings
Queue2 moved into core, so remove its strings.
2010-01-08 16:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.h:
oggdemux: push headers when activating chains
Keep a list of headers for each stream of a chain. When a chain is activated,
push the headers before pushing the data so that decoders can sync.
Fix seeking in chains, take the chain start time into account when comparing
timestamps.
See #606382
2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/lang-tables.dat:
* gst-libs/gst/tag/lang.c:
tag: fix up disting of lang-tables.c more correctly
lang-tables.c is included by lang.c and not really a proper source
file that should be compiled into its own object, so rename it to
lang-tables.dat and put it into EXTRA_DIST instead to ensure it
gets disted.
2010-01-07 13:50:03 +0000 Christian Schaller <christian.schaller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* gst-plugins-base.spec.in:
Add missing source file for tagger to Makefile and update spec file
2010-01-06 18:30:57 -0800 Mark Yen <mook@songbirdnest.com>
* gst-libs/gst/riff/riff-media.c:
riff-media: handle 32 bit raw RGB video.
2010-01-06 13:57:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: decide flac header packet by content rather than count
2010-01-06 13:56:26 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: reset header packet count at bos page
2010-01-06 13:39:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiopayload: add support for buffer-lists
2010-01-06 11:33:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
2010-01-05 17:17:58 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Ignore zero framerate
https://bugzilla.gnome.org/show_bug.cgi?id=606163
2009-12-29 18:45:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
basertpaudiopayload: Respect ptime if it is given
If the ptime is given in the caps, respect it and force the minimum
and maximum sizes to be exactly the requested ptime.
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2009-12-29 18:36:29 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
rtpbasepayload: Store ptime from caps
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2009-12-02 19:40:58 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppayload: Accept maxptime from caps
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 14:11:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggdemux: enhance flac packet duration calculation
2010-01-05 10:38:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
2010-01-04 09:49:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
* tests/icles/test-colorkey.c:
examples: use Gtk+-2.18 API conditionally
so the seek example and colorkey test work with older Gtk+ versions
as well.
Fixes #605960.
2009-12-29 00:53:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/icles/test-colorkey.c:
tests: fix colorkey test up for Gtk+ >= 2.18
Make test-colorkey work with newer versions of Gtk+.
See #601809.
2009-12-29 00:40:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/seek/seek.c:
examples: make seek example work with Gtk+ >= 2.18
Gtk+ broke API slightly with the introduction of
client-side windows in Gtk+ 2.18. Fix up seek
example to work with newer Gtk+ versions.
Fixes #601809.
2009-12-26 23:29:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/icles/stress-xoverlay.c:
tests: fix warning and memory leak in stress-overlay test
Not all messages have structures and we need to unref messages
when returning GST_BUS_DROP in the sync bus handler.
2009-12-26 18:46:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: correctly eat empty and dummy buffers
2009-12-24 19:56:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
adder: be a lot smarter with buffer management
Detect EOS faster.
Try to reuse one of the input buffer as the output buffer. This usually works
and avoids an allocation and a memcpy.
Be smarter with GAP buffers so that they don't get mixed or cleared at all. Also
try to use a GAP buffer as the output buffer when all input buffers are GAP
buffers.
2009-12-24 16:30:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
* tests/check/elements/adder.c:
adder: use collectpads clipping function
Install a clipping function in the collectpads and use the audio clipping helper
function to perform clipping to the segment boundaries.
Fixes #590265
2009-12-24 13:58:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
adder: fix juvenile comment
2009-12-23 21:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: fix typo in debug message
2009-12-23 18:18:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: avoid some type checks
2009-12-23 17:08:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: avoid leaking selector request pads
2009-12-23 15:46:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: avoid leaking queue and typefind
Don't leak the queue and typefind elements that we might link after the
source element.
2009-12-23 15:43:52 +0100 Jonathan Matthew <jonathan@d14n.org>
* gst/playback/gsturidecodebin.c:
uridecodebin: don't name the queue
There is no reason to name the queue.
Fixes #605219
2009-12-23 15:30:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtp.def:
defs: update defs with new symbols
2009-12-22 20:15:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
* gst-libs/gst/rtp/gstrtcpbuffer.h:
rtcpbuffer: add helper functions for SDES types
Add functions to convert SDES names to their types and back. Will be used later
to set SDES items using a GstStructure.
See #595265
2009-12-21 19:12:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* common:
Automatic update of common submodule
From 47cb23a to 14cec89
2009-12-21 18:45:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: add Since marker for the new tolerance property
2009-12-21 07:57:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/lang.c:
docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
2009-12-21 07:50:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
* tests/examples/app/appsrc-stream.c:
* tests/examples/app/appsrc-stream2.c:
tests: don't use deprecated GLib API g_mapped_file_free
Fixes #605100.
2009-12-20 17:34:46 -0800 David Schleef <ds@schleef.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theoraenc: Add encoder controls for libtheora 1.1
Added drop-frames, cap-overflow, cap-underflow, and rate-buffer.
2009-12-19 21:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: increase default drift tolerance to fix glitches with WMA
Increase default drift tolerance to 40ms to avoid glitches with decoders
or formats where there's a lot of timestamp jitter for some reason or
another (in this case: asf/wma), at least until we implement timestamp
smoothing.
2009-12-16 11:43:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: add some debugging
2009-12-15 18:41:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
* gst/audiorate/gstaudiorate.h:
audiorate: add a tolerance property
It may not be uncommon for the input timestamps to experience some jitter
around the 'perfect time'. As such, instead of regularly adding and dropping
samples, optionally allow for some tolerance in a more relaxed approach.
API: GstAudioRate:tolerance
2009-12-15 19:50:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/audiorate/gstaudiorate.c:
audiorate: add documentation
2009-12-15 16:52:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/Makefile.am:
* gst/audiorate/gstaudiorate.c:
* gst/audiorate/gstaudiorate.h:
audiorate: use separate header file
2009-12-14 21:17:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: set DISCONT when resyncing (e.g. newsegment)
2009-12-14 18:47:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: also fill up segments if possible
2009-12-15 19:29:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: fix segment handling
Do not compare a media (buffer) time to a (bogus) running time
(or their offset equivalents).
2009-12-15 19:22:45 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
audiorate: properly report truncated samples as dropped samples
2009-12-13 18:43:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/lang.c:
docs: mention that gst_tag_get_language_name() may return NULL
2009-12-13 18:42:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/tag.c:
checks: some more testing for the new language code functions
2009-12-12 18:58:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
docs: misc. mixer docs improvements
2009-12-12 18:16:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
docs: add short descriptions for API reference contents page
2009-12-12 17:43:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/mklangtables.c:
tag: make internal language names table static
2009-12-12 17:41:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/lang.c:
* gst-libs/gst/tag/mklangtables.c:
tag: don't use GLib 2.22 API
g_mapped_file_unref() was introduced in GLib 2.22, but we depend
only on GLib 2.18, so use g_mapped_file_free() when compiling
against older GLib versions until we bump the GLib dependency.
2009-12-11 23:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
* configure.ac:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/lang.c:
* gst-libs/gst/tag/mklangtables.c:
* gst-libs/gst/tag/tag.h:
* tests/check/libs/tag.c:
* win32/common/libgsttag.def:
tag: add some utility functions for language codes and tags
Add some utility functions for language tags and ISO-639
codes. These are useful for both GUIs and elements. The
iso-codes package is used for language name translations
if available.
API: gst_tag_get_language_codes()
API: gst_tag_get_language_name()
API: gst_tag_get_language_code()
API: gst_tag_get_language_code_iso_639_1()
API: gst_tag_get_language_code_iso_639_2B()
API: gst_tag_get_language_code_iso_639_2T()
2009-12-11 12:02:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: ogm video has constant packet duration
2009-12-10 22:47:53 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggstream.c:
oggdemux: implement old fLaC mapping
2009-12-10 17:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/tcp/gsttcpclientsrc.c:
tcpclientsrc: unset flushing state too
When unlocking, we set the flushing state on the fdset. Implement unlock_stop so
that we can use it to unset the flushing state again.
Fixes #577326
2009-12-10 16:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: remove redundant fields
2009-12-09 19:03:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/vorbisdec.c:
vorbisdec: adapt to new oggdemux
Remove all granulepos hacks and simply use the timestamps from the new oggdemux
like any other decoder.
2009-12-09 19:04:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/vorbisdec.c:
vorbisdec: fix peer query
2009-12-09 17:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: fix query
2009-12-09 16:55:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: small cleanups
2009-12-09 16:38:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/vorbisdec.c:
vorbisdec: use gst_pad_peer_query()
2009-12-09 12:10:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: fix video when subtitles disabled
When we have a source with subtitles but they were disabled with the flags,
still ghostpad the video pad instead of leaving it unlinked.
2009-12-09 09:47:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Only flush downstream on seeks for flushing seeks
2009-12-09 09:35:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad
2009-12-08 17:30:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: update slider only 25 times a second
don't update the slider a 100 times a second, it's likely higher than the screen
framerate and just wastes cpu.
2009-12-08 17:23:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
theora: remove granulepos hacks
Remove the granulepos hacking now that oggdemux outputs timestamps like any
other demuxer.
2009-12-08 13:40:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Fix stream-changed message list iteration
When iterating the list and removing the current element, first
get the next element and then remove the current one and not
the other way around.
2009-12-07 18:49:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: improve keyframe seeking
Improve keyframe seeking.
Fix reverse playback.
2009-12-07 15:42:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: implement keyframe seeking
Implement keyframe seeking in oggdemux by doing the double seek trick. First
seek to the required position, then read pages for all streams to grab the
granulepos (to know the timing of the keyframe) of each stream, then seek back
to the first keyframe.
2009-12-07 09:13:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Some minor cleanup
2009-12-06 18:05:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments
2009-12-04 16:35:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix timestamps after seek
After a seek, discard all packets before the packet with the granulepos on it so
that the output buffers contain valid timestamps.
Reorder some code so that we check the timestamps before allocating and pushing
an output buffer.
Do more checks on valid packets in ogm mode.
2009-12-04 15:39:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: add comment
2009-12-04 14:01:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: don't do math with invalid granulepos
When the current granulepos is unknown and set to -1, don't try to add durations
to it.
2009-12-04 13:14:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: guard against wrong granulepos
Clamp the initial granulepos to 0 instead of going negative for some badly muxed
ogg files.
2009-12-04 12:26:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: don't fail on bogus granulepos
Do some additional checks on the granulpos timestamp before using it for
calculating the duration because oggdemux generates wrong granulepos now.
Fixes seeking somewhat again.
2009-12-03 20:05:29 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: reimplement OGM support
OGM demuxing no longer requires helper elements. It's done internally
in oggdemux. Vorbis comments are still not handled because I don't
have anything to test with.
2009-12-03 17:02:11 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggstream.c:
oggdemux: fix for I-frame-only theora
2009-12-03 01:16:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: log when ogg mapper doesn't accept the setup header packet
2009-12-02 02:08:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: extract width, height and PAR from theora header and add to caps
2009-12-03 23:43:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: extract number of channels from FLAC, speex and vorbis headers
Because we can.
2009-12-03 22:14:34 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaybin2.c:
build: fix build with debug logging disabled.
2009-12-03 21:07:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
ogg: more print fixes
gstoggstream.c:419: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘gint64’
gstoggdemux.c:2253: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
gstoggdemux.c:2333: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
2009-12-03 16:57:48 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
ogg: Fixing some printf format strings
Fixes some printf format strings to make it build on mac.
2009-12-03 18:08:49 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstfactorylists.c:
* gst/playback/gstfactorylists.h:
* gst/playback/gstplaybin2.c:
playbin2: don't iterate the factory lists in non-debug mode
When debugging is disabled, we won't see anything printed anyway.
2009-12-02 23:55:55 -0800 David Schleef <ds@schleef.org>
* gst/videoscale/vs_4tap.c:
Build fix for MSVC
2009-12-02 23:27:55 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/subparse/qttextparse.c:
build: add missing includes for sprintf and atoi
2009-12-01 16:42:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/subparse/gstsubparse.c:
* gst/subparse/qttextparse.c:
subparse: Add support for some tags of qttext
Currently supporting timescale, timestamps, font, size,
textColor, backColor, plain, bold and italic
Fixes #603357
2009-12-01 13:13:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
* gst/subparse/qttextparse.c:
* gst/subparse/qttextparse.h:
subparse: add qttext support
Adds basic support for qttext subtitles, still lacks markup tags
to make it prettier, but the plain text already works.
Implemented according to:
http://www.apple.com/quicktime/tutorials/texttracks.html
http://www.apple.com/quicktime/tutorials/textdescriptors.html
Fixes #603357
2009-12-01 13:22:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: conditionally cleanup sami context
Only cleanup sami context if we are parsing sami subtitles,
otherwise we might have crashes.
2009-12-01 13:19:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Add missing caps to sink caps template
Some caps were missing from the sink caps template when
xml was disabled
2009-12-01 15:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 87bf428 to 47cb23a
2009-12-01 14:14:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From da4c75c to 87bf428
2009-11-30 10:22:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Fix some pad refcount issues
Fixes bug #603345.
2009-11-27 18:54:57 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From 53a2485 to da4c75c
2009-11-25 17:04:41 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: handle theora streams with 0 keyoffset
2009-11-25 16:53:26 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: Handle unknown streams
2009-11-26 14:30:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
Revert "textoverlay: First draw outline text and then the real text"
This reverts commit 60aa09d28c1f9fd29b56876d7ac6c0366d6cef4d.
First drawing the real text and then the outline produces ugly
text in lower resolutions. The outline line width needs to be somehow
changed relative to the resolution. Fixes bug #602924.
2009-11-26 10:30:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiofilter.c:
audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
...and fix code style a bit.
2009-11-26 10:31:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiofilter.h:
audiofilter: Add _CAST variants of the cast macros
2009-11-25 10:26:16 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
audiosink: add adjustement when slaving
Our calibration against the pipeline clock is done with the adjusted
ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
when reusing audio sinks after switching clocks and slaving methods in a
pipeline.
2009-11-25 16:17:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
Fixes bug #602834 and #350748.
2009-11-25 00:46:55 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: Reset last_granule during seeking
Fix case where we would reconstruct the wrong granulepos for
outgoing streams immediately after a seek.
2009-11-24 22:08:09 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
oggdemux: Fix timestamp generation for theora
Timestamp generation was broken by the last commit for formats
with a non-zero granule shift. Also keep track of the last keyframe
so that we can regenerate granulepos for theora.
2009-11-24 21:22:03 -0800 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
* ext/ogg/vorbis_parse.c:
oggdemux: Fix vorbis parsing
Add a granule to granulepos conversion function. Fix the duration
function for vorbis. Handle timestamps on header packets differently
and be more careful about calculating OFFSET and OFFSET_END. After
this change, timestamps for vorbis don't exactly match up with the
timestamps that vorbisparse outputs, but it's unclear if vorbisparse
is actually correct and it would add a lot more code to make oggdemux
match vorbisparse. Fixes #602790.
2009-11-19 19:28:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Transform QoS events to be meaningful for upstream elements
This is necessary because the sinks don't notice the group switches
and the decoders/demuxers have a different running time than the
sinks.
Fixes bug #537050.
2009-11-21 22:05:34 +0100 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
ogg: Fix generation of timestamps and durations
After changing some internal functions, I forgot to update
the code that puts the values on the buffers.
2009-08-29 10:51:48 -0700 David Schleef <ds@schleef.org>
* ext/ogg/Makefile.am:
* ext/ogg/dirac_parse.c:
* ext/ogg/dirac_parse.h:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
* ext/ogg/vorbis_parse.c:
ogg: Add ogg stream parsing
Adds code that parses headers of various formats encapsulated in
Ogg in order to calculate timestamps and durations of each buffer.
Removes the creation of helper decoder elements to do this calculation
via conversion queries.
Fixes: #344013, #568014.
2009-09-04 00:11:38 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggmux.c:
oggmux: don't overwrite object properties
2009-11-21 17:54:49 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/theora/theoradec.c:
debug: also cast packet.packetno to gint64 in debug log
We do this already for granulepos to handle ogg_int64_t mismatches.
2009-11-21 17:47:26 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
debug: fix format string that was missing a var
2009-10-10 00:32:04 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
* tests/check/elements/adder.c:
adder: make events succeed, if they succed on atleast one pad
2009-11-19 14:51:33 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: error when all streams have no buffers
In some cases (all buffers dropped by a parser) a decodebin2
chain might receive an EOS before it gets enough data to
expose a decoded pad. In the case that no streams can expose
a pad we should error out instead of hang.
Fixes #542758
2009-11-19 12:23:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Fix stupid bug introduced in last commit
2009-11-19 12:10:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Aggregate the stream-changed message by looking at the seqnum
Just counting how many messages were sent and how many were received
is not good enough because they might've been duplicated (e.g. by the
visualization audio tee). Comparing the sequence numbers should give
better results in that case.
2009-11-19 10:05:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Ignore async state changes of the uridecodebins
Otherwise the async state change from READY->PAUSED of the
uridecodebins will take playbin2 from PLAYING->PAUSED again
during gapless group switches.
Fixes bug #602000.
2009-11-19 10:30:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 0702fe1 to 53a2485
2009-11-18 14:50:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: set to buffer less on no-more-pads
When a decodebin2 receives no-more-pads of a group it
can set that group's multiqueue buffering thresholds to
'playing' buffering method, avoiding that it buffers
too long and cause problems when using with queue2.
See the associated bug for details.
Fixes #600787
2009-11-18 17:09:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: fix initial calibration
When we are calibrating the internal clock against the external clock take into
account the time offset applied to our internal clock because we will subtract
that in the render_function again.
2009-11-18 09:22:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Don't handle DURATION queries during group switches
During a group switch return the cached duration of the old group
because the old group still didn't finish playback. If we have no
cached duration return FALSE.
Fixes bug #585969.
2009-11-15 19:36:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Post a stream-changed message after activating a group
This is useful to detect when playbin2 has really switched to the next
group after about-to-finish for example.
Fixes bug #584987.
2009-11-18 12:27:19 +0000 Jan Schmidt <thaytan@noraisin.net>
* win32/common/libgstvideo.def:
win32: Add new still-frame API to the defs
Add gst_video_event_new_still_frame() and
gst_video_event_parse_still_frame() functions to the win32 defs files
2009-11-18 12:37:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: fix 'uninitialized' compiler warning
2009-11-18 10:14:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump core requirement to 0.10.25.1
We depend on new API that's only in git so far.
2009-11-15 17:34:37 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
* tests/check/libs/video.c:
video: Add functions to create/parse still frame events.
Add a new video event to mark the start or end of a still-frame
sequence, and a parser function to identify and extract info from
such events.
API: gst_video_event_new_still_frame()
API: gst_video_event_parse_still_frame()
Fixes: #601942
2009-11-17 16:39:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: make sure we always go to PAUSED async
Set the need_async_start flag before going to PAUSED so that we always post the
ASYNC_START message, even after reusing playsink.
2009-11-17 16:37:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: make sure we remain a sink
When we remove our elements, we could lose our sink flag. Make sure we remain a
sink by setting the flag again after removing elements.
2009-11-16 22:47:54 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: remove unused array
2009-11-16 09:57:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Use new double->fraction transformation function from core
2009-11-14 14:05:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Make subtitle error handling more robust and ignore late errors too
Make sure, to only "simulate" subtitle no-more-pads if it was still
pending and also handle errors in the subtitle pipeline as warnings
after the subtitles prerolled.
Don't set the suburidecodebin to READY after errors, handle_message
will usually be called from the streaming thread and doing that
from there is obviously not a good idea.
2009-11-14 13:21:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode
2009-11-13 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Don't leak the GError and debug string when parsing error messages
2009-11-13 11:16:44 +0100 Sreerenj B <bsreerenj@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: avoid crashing on SIGPIPE
Use send() instead of write() so that we can pass the MSG_NOSIGNAL flags to
avoid crashing with SIGPIPE when the remote end is not listening to us anymore.
Fixes #601772
2009-11-11 17:35:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Improve subtitle passthrough in uridecodebin
Now the caps property isn't set anymore for the subtitle caps
but instead in the autoplug-continue signal it is detected
if the caps belong to a supported subtitle stream.
This makes automatic use of newly installed plugins.
2009-11-11 17:08:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Only recreate factory caps if necessary and cache them
2009-11-10 18:27:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Only update the factory list when the registry has changed
Also don't free the list every time we go to NULL.
2009-11-08 15:04:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: Use gst_pad_get_caps_reffed()
2009-11-07 21:38:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
playbin2/playsink: Use new "silent" property instead of unlinking
This makes sure that subtitleoverlay still gets segment updates and
everything to pass on downstream. Without this segment problems happen.
2009-11-07 21:10:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Update segments after pushing the events downstream
This makes sure that we don't apply segments twice downstream. Also
always send our newsegment events downstream.
2009-11-07 21:09:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Add silent property to disable subtitles
This tries to disable subtitles in the overlay or renderer
and if that's not possible it goes into passthrough mode.
2009-11-07 11:46:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Set the video framerate on parsers if possible
Fixes bug #599649.
2009-11-07 11:31:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
subparse: Make fps a GstFraction typed property and use it properly
2009-11-07 11:08:19 +0100 Iago Toral <itoral@igalia.com>
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
subparse: Add property for the video framerate
2009-11-06 12:51:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Handle external subtitles better
First of all, make sure that suburidecodebin never
errors out because of not-linked in case external subtitles
are used but then subtitles are disabled.
And then make sure that external subtitles always start from
the correct position and are not racing until EOS if they
get unselected and selected again.
2009-11-04 17:29:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Flush the subtitles before switching to a new subtitle stream
This makes sure that all currently shown subtitles disappear
and new ones can be shown as soon as possible.
2009-11-03 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Set subtitle caps as raw caps for the uridecodebins
This will make sure that no subparse is ever plugged and subtitleoverlay,
that subpicture streams are handled the same was as subtitles and that
subtitle renderers are used if available.
Fixes bugs #595123, #570753, #591662, #591706.
2009-11-03 12:33:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2/playsink: Remove everything related to subpicture streams
These will soon be handled the same way as subtitle streams.
2009-11-02 15:50:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Add a queue before subtitleoverlay
This will improve playback, and the same thing is done
for subpicture streams too.
2009-11-02 15:05:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Use subtitleoverlay for subtitles
2009-11-02 07:43:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
subtitleoverlay: Add to the docs
2009-10-13 16:48:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstplayback.c:
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
subtitleoverlay: Add new element for generic subtitle overlaying
This autopluggs the required elements for parsing and rendering
different subtitle formats on a video stream.
Fixes bug #600370.
2009-11-11 19:32:01 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: Keep timestamp from incoming buffer if it is valid
Fixes bug #601627.
2009-11-11 14:00:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playback: Update factories list on every access if the registry has changed
This makes application's simpler because the element doesn't need to
go to NULL first to make use of newly installed plugins.
Fixes bug #601480.
2009-11-10 18:13:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playback: When going from NULL->READY check if the registry has new features
This makes it possible to use newly installed plugins after going back
to NULL instead of requiring a new instance.
Fixes bug #599266.
2009-11-10 13:55:26 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Clear the EOS state on a seek.
Allow seeking back into the stream after it hits EOS.
2009-11-10 12:21:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioresample/README:
* gst/audioresample/arch.h:
* gst/audioresample/fixed_arm4.h:
* gst/audioresample/fixed_arm5e.h:
* gst/audioresample/fixed_bfin.h:
* gst/audioresample/fixed_debug.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample_sse.h:
* gst/audioresample/speex_resampler.h:
audioresample: Update speex resampler to latest GIT
2009-11-10 00:48:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: assign chain->mute before using it
Fixes GObject warnings when starting totem.
2009-10-28 22:10:33 -0700 David Schleef <ds@schleef.org>
* ext/theora/theoradec.c:
theora: Fix alignment of frames when converting
Fix logic inversion in calculating the offset in the theora
frame when copying to a GStreamer frame.
2009-11-09 19:58:20 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstfactorylists.c:
playback: Fix the order in strcmp that I broke in previous commit.
2009-11-09 19:16:21 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: Reduce number of calls to gst_type_find_peek.
Shaves off a couple percents off typefinding
2009-11-09 17:49:51 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstfactorylists.c:
playback: Avoid expensive API calls in tight loop.
We know we're dealing with GstPluginFeature.
2009-11-09 18:11:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/cddabasesrc.c:
cddabasesrc: Add unit test for property settings
Also includes a regression test for bug #601104.
2009-11-09 18:04:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/cdda/gstcddabasesrc.c:
cddabasesrc: Never return a negative track number in get_uri()
2009-11-09 18:03:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/cdda/gstcddabasesrc.c:
cddabasesrc: Don't set the track to 1 every time a device is set
Fixes bug #601104.
2009-11-08 11:27:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
inputselector: Remove useless variables and fix a uninitialized variable compiler warnings
2009-11-06 17:01:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Add property to disable/enable posting of stream-topology messages
Most people don't need this messages and generating them is quite
expensive.
2009-11-06 15:12:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Protect subtitle elements and subtitle encoding by a new mutex
Using the object lock here can and will lead to deadlocks because
of deep-notifies of property changes: the deep-notify handler will
get the parent of objects, which will take the object lock again.
Fixes bug #600479.
2009-11-06 13:13:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
inputselector: Make sure that running_time->timestamp calculation never becomes negative
2009-11-06 13:25:05 +0200 Mart Raudsepp <leio@gentoo.org>
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
examples: Correct casting of g_signal* funcs first arguments
This completes the deprecated GTK API fix in commits 81a0a986 and
79adfa54 - unlike gtk_signal_connect and co, g_signal_connect and
co take a gpointer, not a GtkObject.
2009-11-06 12:25:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Improve all-raw-caps detection for pads
2009-11-06 12:19:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
basesrc: fix startup position in the ringbuffer
When we start and we need to produce the first sample, go to the next sample
that will be written into the ringbuffer instead of trying to go to sample 0.
We relied on rather small ringbuffer sizes to correctly go to the current
sample, which breaks whith large buffers.
Fixes #600945
2009-11-06 11:26:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
inputselector: Use the start time (i.e. timestamp) as the last stop
Using the end time makes it impossible to replace buffers, which is
a big problem for subtitles that could have very long durations.
2009-11-06 12:08:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Synchronize video/text based on the running time
Instead of simply using the buffer timestamps.
2009-11-06 09:30:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Clip text buffers to the text segment and reset segments properly
2009-11-06 09:01:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: Put the video segment into the instance struct instead of allocating it separately
2009-11-06 09:05:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Check if text timestamp/duration is valid before clipping
2009-11-05 23:33:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: printf format fix
2009-11-05 15:42:09 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
* gst/gdp/gstgdpdepay.c:
gdpdepay: Clear adapter on flush and state change
Fixes #600469
2009-11-05 13:12:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstinputselector.c:
inputselector: use _get_caps_reffed()
2009-11-05 13:00:27 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
pad: rename new api from _refed to _reffed.
Due to popular demand rename the new api as we still can.
2009-11-04 18:57:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playbin2: avoid copying caps
Use get_caps_refed() when we can.
2009-11-04 18:31:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: use new getcaps function to avoid copies
Use the gst_pad_get_caps_refed() to avoid some caps copy functions.
2009-11-04 17:50:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: use faster element_link_pads
Use the faster gst_element_link_pads because we know for sure the sinkpad name
and we don't need to have the function search for a suitable pad anymore.
2009-11-04 16:16:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: make drift tolerance configurable
Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
drift or timestamp drift instead of relying on the latency-time value for clock
drift and 500ms for timestamp drift.
Remove warning about discont timestamp and simply resync. The warning is in some
cases not correct and is triggered more frequently now that we lower the
tolerance value.
2009-11-04 10:52:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Return NOT_LINKED for unselected text pads from a demuxer
We want to return NOT_LINKED for unselected pads but only for pads
from the normal uridecodebin. This makes sure that subtitle streams
are not raced past audio/video from decodebin2's multiqueue.
For pads from suburidecodebin OK should always be returned, otherwise
it will most likely stop with an error.
2009-11-04 08:20:59 +0100 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstinputselector.c:
inputselector: also add inline to the proto to fix the build
Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.
2009-11-03 12:01:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Initialize caps property with the default raw caps
2009-11-03 11:48:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstrawcaps.h:
decodebin2: Use static caps for the default raw caps and put them into a separate header
This way we can use the same default raw caps everywhere.
2009-11-03 08:26:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: First draw outline text and then the real text
Improves the output a bit because no parts of the outline are
overwritten again.
2009-10-31 14:02:40 +0100 Josep Torra Valles <n770galaxy@gmail.com>
* gst/playback/gstplaybin.c:
playbin: Make sure to keep a reference on the volume element
Fixes null pointer dereferences under certain circumstances.
Fixes bug #595401.
2009-10-31 09:47:54 +0100 Edward Hervey <bilboed@bilboed.com>
* po/POTFILES.in:
po: queue2 has moved to core
2009-10-30 09:24:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Reset {mute,volume}-changed flags after setting the volume
These flags are there to make sure that the volume is set, if there
is no volume element yet.
2009-10-30 09:24:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state
2009-10-29 14:30:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
Fixes bug #600027.
2009-10-29 14:19:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2
2009-10-29 11:37:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/plugins/inspect/plugin-queue2.xml:
queue2: Remove inspect file
2009-10-29 11:29:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c:
queue2: Remove from gst-plugins-base
This is now in coreplugins.
2009-10-28 11:29:36 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
docs: include more indexes
2009-10-28 11:13:20 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
docs: turn entities into xi:includes
This is faster to process and easier to maintain. Its also less 80s.
2009-10-28 10:17:43 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtp: dump packets which we reject
2009-10-28 01:01:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/pipelines/.gitignore:
.gitignore: ignore basetime unit test binary
2009-10-28 00:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstqueue2.c:
* gst/playback/gststreaminfo.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstssaparse.c:
Remove GST_DEBUG_FUNCPTR where they're pointless
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
2009-10-27 15:23:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstqueue2.c:
queue2: add custom acceptcaps function
2009-10-27 15:22:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: implement low/high watermark property
2009-10-23 14:56:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: add checkbox to enable buffering
2009-10-23 14:54:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: don't use 2 buffering elements
Only use the multiqueue buffering when we don't have a stream (and thus are
using queue2 to do the buffering already).
2009-10-23 14:34:42 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplay-enum.c:
* gst/playback/gstplay-enum.h:
* gst/playback/gstplaybin2.c:
playbin2: add flag to enable decodebin buffering
Add a flag that enables buffering in decodebin.
2009-10-23 14:32:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: buffering is implemented now
2009-10-23 14:30:52 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: buffering is implemented now
2009-10-23 14:09:17 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: configure use-buffering on multiqueue
2009-10-23 13:58:25 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: use 0 for max buffer size
2009-10-23 13:53:21 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: set some reasonable defaults
2009-10-23 13:44:12 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: set buffering properties on decodebin2
Propagate the buffering properties on decodebin2 but only if we are not already
doing download buffering.
2009-10-23 11:52:09 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: add use-buffering property
Add a use-buffering property that will perform buffering on the parsed or
demuxed media.
2009-10-23 11:31:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: refactor queue size configuration.
Refactor the queue size configuration into a new method.
Use the same queue values for buffering as for preroll.
2009-10-23 11:08:50 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: move error path down
2009-10-23 11:02:40 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: implement max queue size properties
2009-10-23 10:42:23 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: add properties for buffering
Add properties that can be used to configure the multiqueue buffers and
buffering methods
2009-10-24 13:19:08 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/examples/app/Makefile.am:
* tests/examples/seek/Makefile.am:
* tests/examples/v4l/Makefile.am:
examples: fix linking order.
the uninstalled wrapper would create a LD_LIBRARY_PATH with system-wide
path before the local ones... resulting in the example applications picking
up the system-wide libraries and not the (potentially modified) uninstalled
libraries
2009-10-24 13:08:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Don't destroy the suburidecodebin on errors
It can still be reused
2009-10-24 13:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: If setting the state of the suburidecodebin fails just warn, don't error out
2009-10-24 12:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Don't set uridecodebin states to NULL before reusing them
This makes sure that the internal decodebin2 and everything else can
be reused without reinstantiation.
2009-10-18 17:28:22 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gsturidecodebin.c:
uridecodebin: Store unused decodebin2 instances for further usage.
This allows faster re-use of uridecodebin.
https://bugzilla.gnome.org/show_bug.cgi?id=599471
2009-10-23 17:49:15 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoraparse.h:
* ext/theora/theoraparse.c:
theora: Convert theoraparse to libtheora 1.0 API
2009-10-21 12:38:59 +0300 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
rtpaudiopayload: Only sent exact multiple of the frame size
Also align the maximum size with the frame size, not only the minimum
2009-10-22 09:12:03 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
* gst/audiorate/gstaudiorate.c:
audiorate: move debug calculation into debug macro
Remove in_duration and move its calculation to
GST_LOG_OBJECT macro. This way it will only be calculated
if we have debug enabled.
2009-10-22 09:06:02 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
* gst/audiorate/gstaudiorate.c:
audiorate: Removing unused variable
The in_stop variable was never read. Removing it.
2009-10-22 08:40:01 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
* gst/audiorate/gstaudiorate.c:
audiorate: be more accurate on offset math
Replace gst_util_uint64_scale_int for its rounding version
to improve accuracy and avoid inserting samples where
they aren't needed.
Fixes #499181
2009-10-22 10:17:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Optimize a bit more
...and add a FIXME for bug #598695 and explain
what we should do once Pango supports user fonts.
2009-10-22 10:02:11 +0200 Iago Toral <itoral@igalia.com>
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c:
subparse: Add support for DKS subtitle format
Fixes bug #598936.
2009-10-22 09:31:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Do shading as first operation
2009-10-22 09:08:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Only use a single cairo surface for drawing
... and comment/optimize what is going on here a bit better.
2009-10-21 16:24:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstinputselector.c:
inputselector: set output caps before pushing
Set the output caps on the srcpad before pushing the buffer because else core
will do a rather expensive check to see if we can actually accept those caps on
the srcpad.
2009-10-21 15:58:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstinputselector.c:
inputselector: install an acceptcaps function
Install a custom acceptcaps function instead of using the default expensive
check. We accept whatever downstream accepts so we pass along the acceptcaps
call to the downstream peer.
2009-10-21 20:35:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: fix typo in previous mxf typefinder change
2009-10-21 20:44:33 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: speed up mxf_type_find over 300 times for worst case scenarios
* memcmp is expensive and was being abused, reduce calling it by checking
the first byte.
* iterating one byte at at time over 64 kbites introduces a certain overhead,
therefore we now do it in chunks of 1024 bytes
And I do mean over 300 times. The average instruction call per mxf_type_find
was previously 785685 and it's now down to 2458 :)
2009-10-20 17:13:39 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstfactorylists.c:
decodebin2: avoid type checks
2009-10-20 09:00:28 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
gst/decodebin2: Ensure we get fixed caps for topology message
There are some corner cases (like with dvdemux amongst others) where
the caps won't be negotiated, but the pad has fixed caps.
2009-10-20 08:52:36 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
gst/decodebin2: Don't expose chains if we're shutting down.
This avoids adding flushing pads to ourself
2009-10-17 21:16:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* ext/pango/gsttextoverlay.c:
pango: bump pango requirement to stable version and remove ifdefs
Bump pango requirement from an ancient development version to an
ancient stable version.
2009-10-17 21:11:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/.gitignore:
.gitignore: update after files got renamed
2009-10-16 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppayload: small comment fix
2009-10-16 10:50:35 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtp/gstbasertppayload.c:
rtp: Correct timestamping of buffers when buffer_lists are used
The timestamping of buffers when buffer_lists are used failed if
a buffer did not have both a timestamp and an offset.
2009-10-16 10:56:56 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp-marshal.list:
* gst-libs/gst/rtsp/gstrtspextension.c:
* gst-libs/gst/rtsp/rtsp-marshal.list:
* gst-libs/gst/video/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
build: fix previous commit to fully accomodate the glib-gen.mak changes
I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
2009-10-16 10:18:45 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114
The build rules in glib-gen.mak were using pattern rules in a non save way.
2009-10-16 10:14:36 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 85d1530 to 0702fe1
2009-09-10 11:39:18 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoradec.c:
theora: Make theoradec use gstvideo for image conversion
Vastly simplifies code.
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-10 09:36:31 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoradec.c:
theora: Don't always round to even width/height
Previously, the code always rounded to even sizes. Now it only ensures
that pic_x and pic_y are multiples of 2 if the output format requires
it.
Also inlcudes fixes to take pic_x/y into account properly when copying
the buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-10 00:00:44 +0200 Benjamin Otte <otte@gnome.org>
* configure.ac:
theora: Don't check for theora.pc anymore
THe new APIs from theoradec and theoraenc are used now.
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
theora: Convert theoradec to libtheora 1.0 API
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-09 23:44:36 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/Makefile.am:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Port encoder to new Theora API
Includes ripping out the old buffer copy code to fill up to frame size.
This is not necesary with the new encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-09 21:59:31 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Disable sharpness property
It's ignored by libtheora
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-09 21:57:08 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Disable noise-sensitivity property
It is ignored by libtheora
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-09 21:50:57 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Disable keyframe-mindistance property
It's ignored by the current Theora library
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-09 21:48:08 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Disable keyframe_threshold property
It's ignored by the current theora encoder
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-09 20:26:47 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Get rid of "quick" property
The proeprty is not used by libtheora at all
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-08 15:12:23 +0200 Benjamin Otte <otte@gnome.org>
* configure.ac:
* ext/theora/theoraenc.c:
theora: remove support for outdated granulepos hack
This is in preparation to switching to switching to the new Theora API
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-08 13:23:04 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Ignore border property
Always make the video use black as padding color.
The output will be identical to previous versions.
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-09-08 13:18:26 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theora: Ignore the center property, always set video to top left
This is not a necessary property, the output will be identical no matter
what.
https://bugzilla.gnome.org/show_bug.cgi?id=594729
2009-10-15 16:34:28 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/Makevars:
po: Don't create backup .po files
As well as preventing creation of useless backup files, it works
around a bug in gettext 0.17 on OS/X
2009-10-15 13:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Post a element message on the bus with the stream topology
Fixes bug #598533.
2009-10-15 13:01:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Store the "endcaps" of a chain
This are the caps that either resulted in a deadend if
no plugin for them could be found or raw caps.
2009-10-15 11:38:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Store for every chain, which pad resulted in its creation
2009-10-15 10:28:39 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/pipelines/basetime.c:
check: Don't fail the basetime test when no audiosrc is available
On OS/X the DEFAULT_AUDIOSRC is not going to be available, because
it isn't in gst-plugins-base. Just defer the test, instead of
failing it.
2009-10-14 10:41:03 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From a3e3ce4 to 85d1530
2009-10-14 08:36:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Use gst_object_has_ancestor() instead of our own implementation of it
2009-10-13 19:14:41 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: fix timestamp comparission, Fixes #597407
2009-10-13 13:52:02 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
* tests/check/Makefile.am:
* tests/check/pipelines/basetime.c:
tests: new test for baseaudiosrc base_time comparison
This test reveals a bug in comparison operation between timestamp and
GstElement's base_time in GstBaseAudioSrc.
2009-10-08 19:55:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Don't stop completely on initialization errors from subtitle elements
Instead disable the subtitles and play the other parts of the stream.
Fixes bug #587704.
2009-10-13 16:50:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Ignore no-more-pads from non-demuxer elements
instead of printing an error that no corresponding group could
be found. no-more-pads from non-demuxer elements doesn't give
any additional information because there can only be a single srcpad.
Fixes bug #598288.
2009-10-12 21:30:15 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: track active conversion in perf log
2009-10-12 15:48:46 +0200 Patrick Radizi <patrick.radizi at axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: handle socket errors
gstrtspconnection.c:gst_rtsp_connection_receive() can hang when an error occured
on a socekt. Fix this problem by checking for error on 'other' socket after poll
return.
Fixes #596159
2009-10-06 14:08:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c:
audioclock: whitespace fixes
2009-10-06 14:07:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: avoid confusing error
2009-10-09 22:00:45 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
vorbis: fixes warings in macosx snow leopard
2009-10-09 18:52:12 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/theora/theoradec.c:
* ext/theora/theoraparse.c:
theora: fixes warnings on macosx snow leopard
2009-10-09 16:56:29 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
ogg: fixes warnings on macosx snow leopard
2009-10-09 16:19:17 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/ogg/gstoggdemux.c:
oggdemux: fix a warning in macosx
2009-10-08 14:16:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/tags.c:
tag: use BOM to recognize UTF-16/32 encoding and convert accordingly
2009-10-09 15:11:16 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/gst-plugins-base.supp:
check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.
2009-10-09 15:32:45 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/gnomevfs/gstgnomevfssrc.c:
audioconvert: change the format instead of cast as ensonic asked
2009-10-09 15:29:15 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/audioconvert/gstchannelmix.c:
audioconvert: fixes warning: format not a string literal and no format arguments
redo of valid part of my previous revert.
2009-10-09 15:19:42 +0200 Josep Torra <n770galaxy@gmail.com>
* common:
* gst/audioconvert/gstchannelmix.c:
Revert "audioconvert: fixes warning: format not a string literal and no format arguments"
Revert this commit as unintentionally I've changed common.
This reverts commit 49ea0138223ec5f9e53780635cbcc70f33778667.
2009-10-09 14:28:42 +0200 Josep Torra <n770galaxy@gmail.com>
* ext/gnomevfs/gstgnomevfssrc.c:
gnomevfssrc: fixes warnings in macosx
warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'GnomeVFSFileOffset'
warning: format '%lld' expects type 'long long int', but argument 9 has type 'guint64'
2009-10-09 14:23:36 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/videorate/gstvideorate.c:
videorate: fix warning in macosx
2009-10-09 14:20:47 +0200 Josep Torra <n770galaxy@gmail.com>
* gst/audiorate/gstaudiorate.c:
audiorate: fix warning in macosx
2009-10-09 14:14:15 +0200 Josep Torra <n770galaxy@gmail.com>
* common:
* gst/audioconvert/gstchannelmix.c:
audioconvert: fixes warning: format not a string literal and no format arguments
2009-10-09 14:07:24 +0200 Josep Torra <n770galaxy@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
audio: fix warnings building on macosx
2009-10-08 18:08:22 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: chwck formats just once per _chain()
2009-10-08 17:49:39 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: add perf-log-category and log suboptimal operation
Log if we use an intermediate colorspace for conversion.
2009-10-08 10:59:36 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From 19fa4f3 to a3e3ce4
2009-10-08 00:17:21 +0100 Jan Schmidt <jan.schmidt@sun.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Fix type-punning warning
2009-09-26 12:56:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Chains with an exposed endpad are complete too
This allows partial group changes, i.e. demuxer2 in the example below
goes EOS but has a next group and audio2 stays the same.
/-- >demuxer2---->video
demuxer--- \--->audio1
\--->audio2
2009-09-26 12:47:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads
2009-09-24 14:56:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Don't post missing plugin messages twice
decodebin2 already posts them after emitting the unknown-type signal,
there's no need to post another one.
2009-09-26 12:17:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Rewrite autoplugging and how groups of pads are exposed
This now keeps track of everything that is going on, creates
a tree of chains and groups to allow "demuxer after demuxer" scenarios
and allows chained Oggs with multiple streams (needs oggdemux or playbin2 fixes).
Also document everything in detail and give a general overview of what
decodebin2 is doing at the top of the sources.
Fixes bug #596183, #563828 and #591677.
2009-10-07 17:45:33 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
ximagesink: only start event thread if needed
The event thread is doing 20 wakeups per second to poll the events. If one
runs ximagesink with handle-events=false and handle-expose=false then we can
avoid the extra thread.
2009-10-07 16:56:28 +0200 Edward Hervey <bilboed@bilboed.com>
* ext/theora/theoraenc.c:
theoraenc: Make the default quality property 48.
This guarantees that people who use theoraenc without modifying any
properties will end up with a reasonably good quality output.
48 is also the default of the encoder_example application shipped with
libtheora.
2009-10-07 11:48:37 +0200 Benjamin Otte <otte@gnome.org>
* tests/check/libs/video.c:
tests/check/libs/video.c: Update strides for Y41B
2009-10-07 10:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: we can use GLib 2.18 API unconditionally now
2009-10-07 10:13:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump GLib requirement to 2.18
Bump required GLib version as per the release planning docs.
2009-10-05 00:33:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/tuner.c:
docs: clarify GstTuner docs in two places
2009-09-25 15:32:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* sys/v4l/gstv4lelement.c:
v4l: fix compiler warning
Fix 'variable may be used uninitialized' compiler warning (which is
true in theory, but can't actually ever happen, since we always
call the function with check=FALSE).
Fixes #596313.
2009-10-07 11:56:35 +0300 Stefan Kost <ensonic@users.sf.net>
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstogmparse.c:
* gst/subparse/gstsubparse.c:
* gst/subparse/mpl2parse.c:
* gst/subparse/tmplayerparse.c:
build: sprintf, sscanf need stdio.h
2009-09-15 15:26:06 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: only start event thread if needed
The event thread is doing 20 wakeups per second to poll the events. If one runs
xvimagesink with handle-events=false and handle-expose=false then we can avoid
the extra thread.
2009-10-07 09:58:27 +0200 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/video/video.h:
Update Since tags for NV12/NV21
They are added in 0.10.26 now, not 0.10.25
2009-09-23 15:31:50 +0200 Benjamin Otte <otte@gnome.org>
* gst/videotestsrc/videotestsrc.c:
[videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles
2009-09-23 11:03:57 +0200 Benjamin Otte <otte@gnome.org>
* gst/ffmpegcolorspace/imgconvert_template.h:
[ffmpegcolorspace] Fix NV12 and NV21 with odd width and height
2009-09-23 10:25:02 +0200 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add NV12 and NV21 formats
2009-09-21 18:49:42 +0200 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/video/video.c:
[video] Fix Y41B
Chroma components should be aligned on 4byte boundaries.
https://bugzilla.gnome.org/show_bug.cgi?id=595849
2009-09-21 18:49:06 +0200 Benjamin Otte <otte@gnome.org>
* gst/videotestsrc/videotestsrc.c:
[videotestsrc] Fix Y41B
Chroma components should be aligned on 4byte boundaries.
https://bugzilla.gnome.org/show_bug.cgi?id=595849
2009-10-07 07:28:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* gst-libs/gst/interfaces/streamvolume.c:
streamvolume: Define cbrt() if it's not available
Fixes build on Win32, bug #597537.
2009-09-24 16:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstfactorylists.c:
factorylist: Use gst_caps_can_intersect() instead of _intersect()
This is faster and results in less allocations.
2009-09-26 12:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't set the external ghostpads blocked but only their targets
Pad blocks should never be done on external pads as outside elements
might want to use their own pad blocks on them and this will lead to
conflicts and deadlocks.
2009-09-26 12:04:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Only use the object lock for protecting the subtitle elements
Using the decodebin lock will result in deadlocks if the subtitle encoding
is accessed from a pad-added handler.
2009-09-26 18:11:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Improve debugging of pad blocks
2009-09-23 16:07:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately
2009-10-06 19:59:11 -0700 David Schleef <ds@schleef.org>
* configure.ac:
configure: Add an 'else' to pangocairo check
Otherwise it exits if it fails.
2009-10-06 19:35:50 -0700 David Schleef <ds@schleef.org>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: add pattern with out-of-gamut colors
Adds a pattern with out-of-gamut colors in a checkerboard
pattern with in-gamut neighbors. Useful for checking YCbCr->RGB
color matrixing. Correct matrixing and clamping will cause the
checkerboard pattern to be invisible.
2009-10-06 19:17:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: use CLOSE_SOCKET() instead of close()
Use CLOSE_SOCKET instead of directly calling close() because it does the right
thing for windows.
Fixes #597539
2009-10-01 14:19:41 +0200 Robert Swain <robert swain gmail com>
* gst/audioresample/gstaudioresample.c:
audioresample: fix printf variable type
Change printf variable type from %lu to %" G_GUINT64_FORMAT " as it
should be for guint64.
Fixes #596981
2009-09-30 23:22:35 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
ffmpegcolorspace: Use the ffmpegcolorspace debug category
Move gstffmpegcodecmap debug to the ffmpegcolorspace category
2009-09-22 11:58:26 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/gdp/gstgdppay.c:
gdppay: Don't repeat tags buffers for every new segment
Only send a tag buffer when one is received, not after every new segment
event/update.
2009-09-28 20:25:35 -0700 David Schleef <ds@schleef.org>
* gst/typefind/gsttypefindfunctions.c:
typefind: detect 'ftypqt ' as video/quicktime
2009-10-06 19:47:00 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
back to development -> 0.10.25.1
=== release 0.10.25 ===
2009-10-05 13:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
Release 0.10.25
2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.24.4 pre-release
2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: make the lock recursive for now
Fixes #583255
2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: fix the vis property getter
2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Add missing file to spec file
2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/cdda/gstcddabasesrc.c:
* tests/check/libs/cddabasesrc.c:
cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
* gst-libs/gst/cdda/gstcddabasesrc.c:
* tests/check/libs/cddabasesrc.c:
cddabasesrc: ignore URI fragments that look like device paths
Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
worked before the fix for bug #321532.
Also adds a check for negative track numbers and some unit tests for URI
parsing.
Fixes bug #595454.
2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.24.3 pre-release
2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: don't ever return NULL in list of strings.
2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaysink.c:
playsink: Expose mute,volume,vis-plugin and font-desc properties
https://bugzilla.gnome.org/show_bug.cgi?id=594623
2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaysink.c:
GstPlaySink: Expose 'reconfigure' as an action signal.
2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaysink.c:
GstPlaySink: Expose flags as a gobject property.
2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplayback.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playback: Register playsink as an element.
This allows using playsink from outside the playback plugin.
Add code to be able to request the sink pads using standard GStreamer API.
TODO : expose GObject properties/signals.
2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs.types:
docs: add new gst_stream_volume_get_type to types file
This is needs to get Gobject features to show up in the docs.
2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
* ext/ogg/gstoggdemux.c:
oggdemux: Fix duration calculation for truncated files
If the last page of a stream has a granulepos of -1, that is,
it doesn't complete a packet, we need to continue to search
for the last granulepos.
2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
* ext/theora/theoraenc.c:
theoraenc: Fix a string leak in _getcaps()
2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/LINGUAS:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.24.2 pre-release
2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/elements/audioresample.c:
check: Improve audioresample test
Make the audioresample test work with CK_FORK=no, and
turn a g_print into a GST_INFO.
2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix crashes with even widths
The fix for green lines introduced by commit
35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
for even widths. This patch fixes it.
2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Implement GstStreamVolume interface
2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
* tests/check/Makefile.am:
* tests/check/elements/volume.c:
volume: Implement GstStreamVolume interface
2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/streamvolume.c:
* gst-libs/gst/interfaces/streamvolume.h:
* gst/playback/Makefile.am:
* win32/common/libgstinterfaces.def:
interfaces: API: Add GstStreamVolume interface
Fixes bug #567660.
2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: properly fix the HTTP manual mode
When we're not parsing HTTP, return EPARSE when we get an HTTP
message.
2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/mixertrack.h:
mixertrack: add READONLY and WRITEONLY flags
Should really have been READABLE and WRITABLE, but those are hard to
add whilst maintaining backwards compatibility. See #343615.
API: GST_MIXER_TRACK_READONLY
API: GST_MIXER_TRACK_WRITEONLY
2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: fix build against core that has debugging disabled
The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videorate/gstvideorate.c:
videorate: Add Since marker for the new skip-to-first property
2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: Make videorate work with a live source
Add a property that makes videorate skip to the first buffer it
receives instead of padding the stream from segment start to the
first real buffer.
Fixes bug #567928.
2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/fft/gstfft.h:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.h:
fft: Mark one function as const and add notes that the structs should be private in 0.11
2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: add human readable format names when logging
Add string array with human readable names for format and type to be used in log
statements.
2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppay: don't print RTP timestamps as clocktime
Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
Fixes #594757
2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
playbin(2): Document that the volume property uses a linear scale
Fixes bug #571610.
2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: don't return EPARSE
Don't blindly return EPARSE when http mode is disabled.
Restore old http mode after temporarily setting it to TRUE.
2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: add ugly backward compat hack
Check for pulsesink < 0.10.17 because it includes code that is now included in
baseaudiosink. Disable that code in baseaudiosink to be compatible with the
older version.
2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
A green border could be visible when converting to Y444 or RGB, because
the last chroma samples weren't copied correctly
2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix YVU9 and YUV9
- Buffer sizes were computed different from ffmpegcolorspace
- Green bar on right size for widths not divisable by 4
2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix image for odd widths in some formats
videotestsrc rounds chroma down. This causes it to omit the last chroma
value completely for odd widths when the chroma is downsampled.
This patch special cases the last pixel to not be rounded down.
2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: Handle kate and cmml as sparse streams too
2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: Better handling of sparse streams by sending segment updates
Fixes bug #397419.
2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gsturidecodebin.c:
docs: tell a biit more about uri-decodebin and buffering
2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: take clock time in setcaps
Take the time of the clock so that the last_time field is set. This is important
for sinks that restart their internal ringbuffer after a caps change and need to
know the last know position.
2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c:
audioclock: add some more debug
2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/theora/theoraenc.c:
theoraenc: Print a debug message with supported formats
2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoraenc.c:
theora: Check supported input formats in getcaps function
We want to fail early when an older libtheora release is used that does
not support Y444 or Y42B formats, so use a getcaps function that does
this.
2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoraenc.c:
theora: Implement support in theoraenc for Y444 and Y42B
Fixes bug #594165.
2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoraenc.c:
theora: Refactor the buffer copy code
2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoraenc.c:
theora: Split yuv_buffer creation into its own function
2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoraenc.c:
theora: Split out buffer resize in its own function
2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
* ext/theora/theoraenc.c:
theora: Add assertions that functions don't fail
Some functions in libtheora can return an error, but that error cannot
ever happen inside theoraenc. In those cases assert that it doesn't.
2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: make stop state configurable
Make it easy to experiment with different stop states (NULL and READY)
2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: correct for clock reset
When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
also make sure that the clock is updated with the elapsed time so that it
alsways increments even when the ringbuffer goes back to 0. When this happened
we need to adjust the sample position for the reset ringbuffer.
Fixes #594136
2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.h:
baseaudiosink: whitespace fixes
2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: add more debug
2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/mixer.h:
whitespace fixes
2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/gstvideosink.c:
* gst-libs/gst/video/gstvideosink.h:
videosink: add "show-preroll-frame" property
Add a property to disable rendering of video frames during preroll. This
will only work for videosinks that use the new ::show_frame() vfunc instead
of overriding basesink's preroll and render vfuncs directly.
API: GstVideoSink:show-preroll-frame
2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/gstvideosink.c:
* gst-libs/gst/video/gstvideosink.h:
video: add GstVideoSinkClass::show_frame()
Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
vfuncs and add some gtk-doc chunks.
API: GstVideoSinkClass::show_frame()
2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/navigation.c:
navigation: don't do stuff inside g_return_val_if_fail() statements
Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/interfaces/navigation.c:
navigation: Fix compiler warning with MSVC
Fixes bug #594275.
2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
basertpdepayload: fix event forwarding
2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/rtp/gstrtcpbuffer.c:
rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
Fixes #594258
2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
fix whitespace
2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: improve slave skew resync
The old one did the mistake of not actually advancing the ringbuffer, it just
adjusted the segbase, introducing the whole lenght of the ringbuffer as an
extra delay in the pipeline.
Also make sure that the resync can never go back in time, producing the same
timestamps that has already been produced, as this can cause severe problems
for sinks and other synching mechanisms.
Fixes #594256
2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: disable typefinder for headerless flac
Disable headerless flac typefinder as long as it happily typefinds anything
including /dev/urandom as flac and as long as it's not particularly useful
given that such streams don't really exist in the wild.
Also fix up some comments so that gtk-doc doesn't complain about them.
2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
* sys/ximage/ximagesink.c:
ximagesink: fix small memory leak when setting window title
2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
* sys/xvimage/xvimagesink.c:
xvimagesink: fix small memory leak when setting window title
2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* .gitignore:
introspection: Add *.gir and *.typelib to .gitignore
2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/video/Makefile.am:
introduction: Fix out-of-tree build
2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/rtsp/Makefile.am:
rtsp: Fix introspection build by ordering sources/headers in dependency order
2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/Makefile.am:
audio: Remove debug echo
2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/Makefile.am:
audio: Fix build of introspection data by using dependency order for the headers/sources
2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
introspection: Strip Gst prefix from all types/functions
2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
introspection: Fix build if gir-repository is not installed
2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/Makefile.am:
video: Add gobject-introspection support
2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
tag: Add gobject-introspection support
2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/sdp/Makefile.am:
sdp: Add gobject-introspection support
2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
libs: Add nodist headers and sources to the introspection files
2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/rtsp/Makefile.am:
rtsp: Add gobject-introspection support
2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/rtp/Makefile.am:
rtp: Add gobject-introspection support
2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/riff/Makefile.am:
riff: Add gobject-introspection support
2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/pbutils/Makefile.am:
pbutils: Add gobject-introspection support
2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/netbuffer/Makefile.am:
netbuffer: Add gobject-introspection support
2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/Makefile.am:
interfaces: Add gobject-introspection support
2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/fft/Makefile.am:
fft: Add gobject-introspection support
2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/cdda/Makefile.am:
cdda: Add gobject-introspection support
This is disabled for now until gobject-introspection is fixed
2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/Makefile.am:
audio: Add gobject-introspection support
2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* gst-libs/gst/app/Makefile.am:
app: Add gobject-introspection support
2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 00a859e to 19fa4f3
2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: fix midi typefinding
We already have a audio/midi typefinder so don't override it with the midi in
RIFF typefinder or else we fail to detect plain midi files.
2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: do buffering for more uris
Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
buffering.
Fixes #594020
2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add typefinder for Midi inside RIFF
This is a standard Midi file format that should be supported by
all Midi decoders and also has the mimetype audio/mid according to
the Midi specification homepage.
Fixes bug #594094.
2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: add some debugging
2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: handle gaps
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: fix frame duration calculations
Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
rtppay: add some debugging
2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: use offsets for RTP timestamps
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: move function around
2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: fix sample duration calculation
2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppay: more refactoring
Unify the sample/frame buffer handling code by making the functions plugable.
2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
audiortppayload: refactor some more
Refactor getting the packet min/max size and alignment code.
Refactor converting bytes to time.
change some variable to something shorter.
2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* win32/common/libgstrtp.def:
audiortppayload: refactor and cleanup
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.
API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Update common
2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppay: add property to disable perfect RTP time
Add a property to disable the generation of perfect RTP timestamps. By default
it is active.
API: GstBaseRTPPayload::perfect-rtptime
2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
basertppay: allow subclasses to influence RTP time
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.h:
basertppay: add macro to cast
2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiopayload: code cleanups
2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
audiortppayload: don't check adapter
the adapter is never NULL so we don't need to check it.
Use _scale functions to avoid overflows.
2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* gst/typefind/Makefile.am:
* gst/typefind/gsttypefindfunctions.c:
typefinding: move gio-based xdg mime typefinder from -bad to -base
Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
reporting a 20% probability and somesuch). Won't be registered if
the gio plugin has been disabled via ./configure --disable-gio.
2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* sys/v4l/v4lsrc_calls.c:
v4lsrc: fix timestamping for when we do not have a clock yet
Should fix #559049.
2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* sys/v4l/v4lsrc_calls.c:
v4lsrc: don't log not-yet-initialised integer value
2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* sys/v4l/v4lsrc_calls.c:
v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
And reflow code to be more indent friendly.
2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
basertppayload: Make instance init faster by not reading /dev/urandom 3 times
... which is the default seed when creating a new GRand. Because
GLib in older versions used buffered IO this would take a lot of time.
Instead use the global GRand for getting random numbers and keep the
three instance GRand for backward compatibility with a simple seed.
Fixes bug #593284.
2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: improve caps filter functionality. Fixes #590146.
Also use the capsfilter if there is no src-peer as the caps constrain what
we can do. Don't create any_caps as a default, as we check for NULL to skip the
filtering. This is a (small) performance regression as we always intersect
otherwise.
2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Post missing plugin messages before any error messages
2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/cdda/gstcddabasesrc.c:
cddabasesrc: safely handle the indexes
2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtsp.def:
def: add new rtsp symbols
2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.h:
basertppayload: whitespace fixes.
2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
* gst/gdp/gstgdppay.c:
Bug 593035 - set IN_CAPS for streamheader buffer
2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
playbin: The internally linked pad of the selector might be NULL in some cases
2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
playbin: Fix iterate internal linked pads functions for the stream selectors
This now used the new gst_iterator_new_single() function and as a side effect
fixes bug #592864.
2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-read.c:
riff: Add support for AVF files
AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
Fixes bug #593117.
2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Detect AVF files as RIFF files too
AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
Partially fixes bug #593117.
2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/audioresample.c:
audioresample: Add unit test for checking for timestamp drifts
This also checks for perfect timestamping and offsetting.
2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
audioresample: Fix drain processing
In case we have to convert internally don't process output length input samples
but history length input samples.
2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/audioresample.c:
audioresample: Improve debugging a bit in the unit test
2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
audioresample: On the first buffer we need discont handling
Otherwise we won't get upstream timestamps and everything and all
output buffers would have -1 timestamps.
2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
* configure.ac:
* gst/subparse/gstsubparse.c:
subparse: Remove dependency on regex.h as it's not used anyway
Fixes bug #592544.
2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
* gst/audioresample/gstaudioresample.c:
audioresample: Fix buffer overflow when pushing the drain
2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
audioresample: Fix timestamp drift
Fixes bug #591934.
2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gsttextrender.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
Remove Ronald Bultje from Authors field
Replaced with "GStreamer maintainers
<gstreamer-devel@lists.sourceforge.net>" or just removed,
depending on the number of other authors.
2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: fix refcounting of _get_sink()
g_value_set_object() increases the refcount of the sink, which is not needed
because the object should already be refcounted. Make sure this is always the
case and use g_value_take_object().
Fixes: #592884
2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspdefs.c:
rtsp: Mark Transport as supporting multiple values.
2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
rtsp: Added missing Since tags.
2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: Improve audiosink startup performance
When we start the ringbuffer, immediatly continue processing samples if the
writer prepared some for us.
Fixes #545807
2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Added new API for sending using GstRTSPWatch.
The new API to send messages using GstRTSPWatch will first try to send the
message immediately. Then, if that failed (or the message was not sent
fully), it will queue the remaining message for later delivery. This avoids
unnecessary context switches, and makes it possible to keep track of
whether the connection is blocked (the unblocking of the connection is
indicated by the reception of the message_sent signal).
This also deprecates the old API (gst_rtsp_watch_queue_data() and
gst_rtsp_watch_queue_message().)
API: gst_rtsp_watch_write_data()
API: gst_rtsp_watch_send_message()
2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Made gst_rtsp_watch_queue_data() thread safe.
2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Added gst_rtsp_connection_set_http_mode().
With gst_rtsp_connection_set_http_mode() it is possible to tell the
connection whether to allow HTTP messages to be supported. By enabling HTTP
support the automatic HTTP tunnel support will also be disabled.
API: gst_rtsp_connection_set_http_mode()
2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
then just setup the base64 decoding context for the first connection.
2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Write as much as possible in gst_rtsp_source_dispatch().
Try to write as much as possible if there are multiple messages queued.
2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Add error_full callback to GstRTSPWatchFuncs.
The error_full callback is similar to the error callback, but allows for
better error handling. For read errors a partial message is provided to
help an RTSP server generate a more correct error response, and for write
errors the write queue id of the failed message is returned.
2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Made read_line() support LWS.
Rewrote read_line() to support LWS (Line White Space), the method used by
RTSP (and HTTP) to break long lines. Also added support for \r and \n as
line endings (in addition to the official \r\n).
2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: Do not split headers which should not be split.
From RFC 2068 section 4.2: "Multiple message-header fields with the same
field-name may be present in a message if and only if the entire
field-value for that header field is defined as a comma-separated list
[i.e., #(values)]." This means that we should not split other headers which
may contain a comma, e.g., Range and Date.
2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Parse WWW-Authenticate headers correctly.
Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
allows commas both to separate between multiple challenges, and within the
challenges themself, we need to take some extra care to split these headers
correctly.
2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Improve parse_line().
Make parse_line() handle keys with multiple values on one line correctly.
2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Rewrote setup_tunneling().
Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
coded strings and duplicates of the message parsing code.
2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: Rewrote gen_tunnel_reply().
Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
than a hard coded string.
2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Ignore the Content-Length for POST requests.
The Content-Length for POST requests with an x-sessioncookie header should
be ignored as the length is bogus and only there to fool proxies.
2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Normalize lines (remove extra whitespace) before parsing.
2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Made parse_string() return a result.
This will catch parsing errors when a too long string is received.
2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Improved parsing of messages.
Do not abort message parsing as soon as there is an error. Instead parse
as much as possible to allow a server to return as meaningful an error as
possible.
2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c:
* gst-libs/gst/rtsp/gstrtspmessage.h:
rtsp: Added support for HTTP messages
2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Added gst_rtsp_connection_create_from_fd().
API: gst_rtsp_connection_create_from_fd()
2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Add initial buffer support.
The initial buffer contains data for a connection which should be used
before starting to actually read anything from the socket.
2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
appsink: don't block in paused
When we are asked to unlock we should either leave the render function or call
the wait_preroll method to release the stream lock.
Fixes #592657
2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: fix includes for appsrc/appsink
2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: Add support for the Authentication-Info header.
The Authentication-Info header is defined in RFC 2617 (Digest Access
Authentication).
2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* tests/check/pipelines/oggmux.c:
oggmux: don't drop the streamheader field from the output caps
Revert previous 'fix' for bug #588717 and fix it properly, whilst
maintaining the streamheader field on the output caps. Also make
sure we don't leak header buffers we couldn't push when downstream
is unlinked. Add unit test for the presence of the streamheader
field on the output caps and for the issue from bug #588717.
2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Avoid duplicated headers.
Remove any existing Session and Date headers before adding new ones
when sending a request. This may happen if the user of this code reuses
a request (rtspsrc does this when resending after authorization fails).
2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Corrected the HTTP digest authorization computation.
Do not use sizeof() on an array passed as an argument to a function and
expect to get anything but the size of a pointer. As a result only the
first 4 (or 8) bytes of the response buffer were initialized to 0 in
auth_digest_compute_response() which caused it to return a string which
was not NUL-terminated...
2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: Also send SEEK events directly to a subpicture sink
2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: If a custom text sink is used, send events to it too
Before, SEEK events would be sent to the video sink, which wouldn't
be linked in any way to the subtitle part of the pipeline and
subparse would never see the SEEK event. This would then seek
the audio/video but the subtitles would continue from the old
position instead.
Fixes bug #591664.
2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Make missing plugins emit a warning message, not an error message
The problem with an error message is, that it will stop playback completely
while it could be that only a audio decoder plugin is missing and the video
could be played with the available plugins.
See bug #591677.
2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Post a correct error message for unknown types
Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
because a plugin is missing and nothing else is wrong.
Also make it an error instead of a warning.
Really fixes bug #591677.
2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Post a missing plugin message additional to the error message on unknown types
Fixes bug #591677.
2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
playbin2: fix error message string
Fixes #591577.
2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/riff/riff-read.c:
riff: align API doc of gst_riff_parse_chunk with reality
2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: avoid assertion failure on empty/NULL caps
2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Also detect SVG by the <svg> starting tag
Not all SVG images have the DOCTYPE specified.
2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: don't use GLib-2.18 function
g_checksum_reset() was added only in GLib 2.18, but we still require
only 2.16, so work around that if we only have 2.16. Fixes #591357.
2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/pipelines/streamheader.c:
streamheader: Fix caps leak in the vorbisenc unit test
2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/pipelines/streamheader.c:
checks: fix stream header unit test hanging in gst_task_cleanup_all()
Set pipelines to NULL state and unref when done.
2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
rtsp: Use GLib's GChecksum instead of our own MD5 implementation
2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
* gst-libs/gst/interfaces/navigation.c:
navigation: Fix doc blurb typo for gst_navigation_send_key_event
2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Allow . instead of , as millisecond delimiter in srt subtitles
Fixes bug #591207.
2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
Revert inlines that cause compiler warnings and are not needed anyway
2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/riff/riff-media.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/video/gstvideofilter.c:
* gst-libs/gst/video/gstvideosink.c:
gst-libs: Remove dead assignments and resulting unused variables.
2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gsttextrender.c:
* ext/vorbis/vorbisenc.c:
ext: Remove dead assignments and resulting unused variables.
2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstfactorylists.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaysink.c:
* gst/playback/gststreamselector.c:
* gst/tcp/gsttcpclientsink.c:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_image.c:
* gst/videotestsrc/gstvideotestsrc.c:
gst: Remove dead assignments and resulting unused variables
2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
* docs/design/draft-va.txt:
docs: add draft for generic introduction of video acceleration APIs idea
2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
Revert "theora: Convert theoradec to libtheora 1.0 API"
This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
Temporarily revert until we have a workaround for debian/ubuntu
packaging failure (see http://bugs.debian.org/528710).
2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add typefinders for many game sound console formats supported by gme
These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: fix warning when we're not linked downstream and error out properly
Fix caps warning when there's no element linked downstream, and pass
not-linked flow return value correctly up the chain, so we error out
correctly. Fixes #588717.
2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
theora: Convert theoradec to libtheora 1.0 API
2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
textrender: Fix blitting of text over the output buffer and cairo painting
2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/icles/test-colorkey.c:
colorkey-test: fix xsync error
2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaysink.c:
playbin2: smarter sink selection. Fixes #588523
Don't do fallbacks if application specified a sink element. When doing the
fallback use configured default elements instead of hardcoded linux only
elements. Improve error messages accordingly.
2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstqueue2.c:
queue2: post error message when pausing task if so appropriate
If a downstream element returns an error while upstream has already
put all data into queue2 (including EOS), upstream will no longer
chain into queue2, so it is up to queue2 to perform some
EOS handling / message posting in such cases. See #589991.
2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: change default slave method
Set the default slave method to the much better skew slaving algortihm.
2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: make buffer writable
Make the input buffer writable before changing its contents.
2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: fix postscript typefinder probability
Two bytes for a rare format hardly warrants MAXIMUM typefinding
probability, POSSIBLE seems more appropriate.
2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
pango: Send queries from the srcpad directly to the video sinkpad
2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
subparse: Implement POSITION query
2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
subparse: Implement SEEKING query
2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
Require latest core for this.
Fixes bug #590430.
2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
pango: Add support for xRGB and BGRx formats
2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
pango: Fix endianness issues from the pangocairo switch
cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
and BGRA on little endian architectures.
2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
pango: Re-add shading support which was dropped by a previous patch
2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* ext/pango/gsttextoverlay.c:
pango: Check if pangocairo supports vertical rendering and fix properties
2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
textrender: Use PROP_X instead of ARG_X consistently
2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
pango: Some minor cleanup
2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
pango: Check for pangocairo instead of pangoft2
2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
* ext/pango/gsttextrender.c:
* ext/pango/gsttextrender.h:
pango: Use pango-cairo instead of pango-ft2
pango-cairo will always use the native font rendering backend
of the platform and provides better results.
Fixes bug #340887.
2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add SVG typefinder
2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Add postscript typefinder
2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Use static caps again for MPEG4 typefinding
2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Implement better & more flexible MPEG4 typefinding
This detects more MPEG4 streams as MPEG4.
Fixes bug #556537.
2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/cdda/gstcddabasesrc.c:
cddabasesrc: Allow to specify the device name in the URI
The allowed URI scheme is now:
cdda://(device#)?track
Also allow every combination of uppercase and lowercase
characters for the protocol part.
Fixes bug #321532.
2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Restrict width/height to 2^15 - 1
Otherwise integer overflows will happen, resulting in segmentation faults.
Fixes bug #590243.
2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Fix indention of template header
2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Clarify documentation about caps and linkage
Fixes bug #589095.
2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Fix typefinding of SDP files
Fixes bug #589574.
2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
* gst/audioresample/gstaudioresample.c:
audioresample: Take the output offsets from the input if possible
Fixes bug #588915.
2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Make sure to allocate enough memory for the temporary buffer
and fix scaling of odd-height interlaced video.
2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Fix interlaced scaling for I420
...and some other minor mistakes in the previous change.
2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Include interlacing information in the AVPicture
This later allows to handle interlaced AVPicture different than
progressive ones which is needed for horizontally subsampled YUV
formats, see bug #589242.
2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
videoscale: Add support for interlaced content
videoscale is not mixing content of two seperate fields anymore
and does scaling on every field separately.
Fixes bug #588761.
2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
back to development -> 0.10.24.1
2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst-plugins-base.doap:
Add 0.10.24 release to the doap file
=== release 0.10.24 ===
2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Release 0.10.24
2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
* tests/check/gst/typefindfunctions.c:
typefinding: fix detection of fLaC id packet in broken flac-in-ogg
There are flac-in-ogg files without the usual flac packet framing
and these files just have a 4-byte fLaC ID packet as first packet.
We need to recognise the type just from these four bytes if we
want oggdemux to recognise these streams correctly.
2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/LINGUAS:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.24.5 pre-release
2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/audio/gstaudiofilter.c:
audiofilter: Don't assert on slightly different caps
Plugins should not assert on incompatible caps, caps negotiation will
fail anyway.
2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
The gio mount example needs GtkMountOperation, which is new in 2.14.
2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
* ext/alsa/gstalsasrc.c:
alsasrc: set alsasrc->handle back to NULL when closing device
Fixes crashes in gst_alsa_find_device_name() when probing or
reading the device-name property (e.g. when doing a dot-file
dump). Fixes #589797.
2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gststreamselector.c:
playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
Rename the GType of the pads of playbin's internal stream selector
element so they don't use the same type name as input-selector's
pads. Fixes #589622.
2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.23.4 pre-release
2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/v4l/.gitignore:
ignores: Ignore v4l probing example binary
2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: recognise Kate spu subtitles as well
Recognise spu-subtitles, SUB and K-SPU as valid categories for
Kate subtitles as well.
2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From fedaaee to 94f95e3
2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Update spec file with latest changes
2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/_stdint.h:
* win32/common/audio-enumtypes.c:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/interfaces-enumtypes.c:
* win32/common/video-enumtypes.c:
0.10.23.3 pre-release
2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: call send_event directly
We can't call gst_element_send_event() from a streaming thread as it gets the
state lock. Instead call the send_event method directly until we have a nice API
for this in basesrc.
Fixes #588746
2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosink.c:
audiosink: Add stream-status messages
Fixes #587695
2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosrc.c:
audiosrc: Add stream-status messages
See #587695
2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/adder/gstadder.c:
gstadder: Don't forget to free pending events on flush/dispose.
Fixes #588747
2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/elements/adder.c:
tests/adder: Add stream consistency checking. Fixes #588748
2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: Make sure tags are properly serialized. Fixes #588746
We do this by letting the basesrc base class handle the tags.
2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
adder: Collect incoming tag events and send them after newsegment. Fixes #588747
2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
* ext/vorbis/vorbisdec.c:
vorbisdec: Check for empty tag strings. Fixes #588724
2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstqueue2.c:
queue2: fix leak and improve buffering
Keep track of the max requested position and compare this to the write position
in the temp file to get the current amount of buffered data.
Fix memleak of all incomming buffers.
Fixes #588551
2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/Makefile.am:
* gst/playback/gstinputselector.c:
* gst/playback/gstinputselector.h:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
playbin2: use private copy of input-selector
We shouldn't really depend on elements from -bad for stream
selection in playbin2, so use a private copy of input-selector
until the selector plugin is ready to be moved to -base or -good.
Fixes #586356.
2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstinputselector.c:
* gst/playback/gstinputselector.h:
playback: add private copy of the input-selector from gst-plugins-bad
Not hooked up yet though. See #586356.
2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
* tests/examples/v4l/Makefile.am:
examples: fix v4l probe example build
Fixes bug #588550.
2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.23.2 pre-release
2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/LINGUAS:
* po/tr.po:
Add Turkish translations
2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/elements/adder.c:
adder: One more attempt to fix the adder test
Give up and discard and recreate the alsasrc after checking it can
be opened, due to some strange crash inside alsa when we don't.
2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/elements/adder.c:
adder: Perform get_state() in the unit test
Wait for the alsasrc to return to NULL after setting it to PAUSED for
testing, otherwise it leads to segfaults later on.
2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/elements/adder.c:
adder: Don't fail when alsasrc is unavailable
Make the liveadder test succeed silently when it can't be completed
either because alsasrc is unavailable, or because the device is
inaccessible.
2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
Differentiate subtitle streams and lyrics/cracktastic/complex streams via
the category string in the headers. This seems like a useful distinction
to make, and also seems more future-proof. See #525743.
2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
* ext/ogg/gstoggmux.c:
oggmux: add Kate caps to the list of accepted types
See #525743.
2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gsturidecodebin.c:
uridecodebin: treat uri-schemas incasesensitive
Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
Fixes not showing buffering messages e.g. for HTTP://...
2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/interfaces/navigation.c:
navigation: simplify docs
Make short-desc short - its used in the toc. Strip uneeded markup.
2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
* win32/common/libgstnetbuffer.def:
* win32/common/libgstvideo.def:
win32: Fix exports
Remove methods from video base classes that have moved to -bad.
Add gst_netaddress_to_string
2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/examples/gio/.gitignore:
ignores: ignore the giosrc-mounting example binary
2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/interfaces/navigation.c:
navigation: Add some partial documentation
Add a general documentation blurb for the GstNavigation functionality.
Still lacks some example code and detail on how to implement it.
2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for Siren codec and make two descriptions non-translatable
2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* common:
Automatic update of common submodule
From 5845b63 to fedaaee
2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
riff: add siren to the RIFF parser
Add siren7 caps to the RIFF parser.
2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/v4l/Makefile.am:
* tests/examples/v4l/probe.c:
v4lsrc: add a simple test case for device probing
2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
* configure.ac:
* sys/v4l/Makefile.am:
* sys/v4l/gstv4lelement.c:
v4lsrc: optional support for device probing with gudev
Enumerate v4l devices using gudev if available.
Fixes bug #583640.
2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: add since tags to docs
2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: don't automatically start pipeline in DB
Keep the pipeline paused when we detect download buffering. The user has to
manually start the pipeline for now because we can't estimate when the buffering
will finish or when we have underrun.
2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstqueue2.c:
queue2: flush differently, avoiding deadlocks
Don't flush the file by closing and opening it but instead use g_freopen. This
avoids a deadlock in shutdown because we emit the temp-location property change
with the wrong lock held.
2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: add a checkbox for progressive download
2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: Fix template construction
Fix the construction of the temporary filename construction as the application
name can be NULL and we don't want a separator between the prgname and the
template.
2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplay-enum.c:
* gst/playback/gstplay-enum.h:
* gst/playback/gstplaybin2.c:
playbin2: add support for progressive download
Add a new playbin2 flag (initially disabled) to enable progressive download
buffering in uridecodebin.
2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: add download property
Add a download property that will attempt to configure queue2 into progressive
download buffering.
Make sure we only enable download buffering for quicktime and flv formats.
2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstqueue2.c:
queue2: add temp-template property
Add a new temp-template property so that queue2 can securely allocate a
temporary filename. Deprecate the temp-location property for setting the
location but still use it to notify the allocated temp file.
2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
adder: add a caps-property to avoid to need to plug a capsfilter afterwards
Adder can only handle one common format accross the pads. Thus one needed to add
a capsfilter afterwards and manage the caps. Now one can simply set the caps on
the property.
2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
* tests/check/elements/adder.c:
adder: skip live-seek text if we have no audiosrc, add new test
The seek-test needs a real audiosrc. Also add a test that checks that adder is
reusable. Finaly handle warnings as warnings to fix a assertion.
2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiosink.c:
gio: Also post a "not-mounted" message from giosink
2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/gio/giosrc-mounting.c:
gio: Remove workaround for playbin2 bug in the sample application
The playbin2 bug was #588078.
2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
If READY->PAUSED failed in the source element we would've swapped
the current and next group already. To allow READY->PAUSED to succeed
after the first failure we have to swap the current and next group
back again. This also ensure that we're again in the same state
as before the failed state change and not at the next group.
This was especially a problem for playbin2 pipelines that use the
new mounting support in giosrc as the source would fail for READY->PAUSED
the first time, the application mounts the location and then tries
to go READY->PAUSED again (and this time it would succeed).
Fixes bug #588078.
2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/gio/Makefile.am:
* tests/examples/gio/giosrc-mounting.c:
gio: Add example application that shows how to handle the "not-mounted" message
2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
gio: Remove the experimental status from the GIO plugin
Fixes bug #510417.
2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
gio: Add documentation for the new "not-mounted" and "file-exists" messages
2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesrc.c:
gio: Make sure that we have the correct stream position when starting
2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesink.c:
gio: Make sure to flush the output stream if it shouldn't be closed
Otherwise there might still be unwritten data after the element
has stopped.
2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c:
* ext/gio/gstgiobasesrc.h:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
gio: Don't close the GIO streams for the giostream{src,sink} elements
This makes it possible to do something useful with the streams
after the element has stopped. Fixes bug #587896.
2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/pipelines/gio.c:
gio: Try to reuse the pipeline with the same stream objects
2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesrc.c:
gio: Improve the error message if a stream is already closed before usage
2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiosink.c:
gio: Post a custom file-exists message on the bus if the file already exists
An application can handle this message, remove the file in question
and restart the pipeline again without showing an error.
This fixes bug #529300.
2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiosrc.c:
gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiosink.c:
gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiosrc.c:
gio: Post a custom "not-mounted" message on the bus
This allows applications to mount the GFile if possible and restart
the pipeline instead of simply giving an error.
2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
* gst/audioconvert/gstchannelmix.c:
audioconvert: Fix compilation when debugging is disabled
Fixes bug #587980.
2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.h:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosink.h:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsink.h:
gio: Add vfunc for requesting the stream for the sinks too
2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
gio: Some more random cleanup
2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgio.c:
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesrc.c:
* ext/gio/gstgiobasesrc.h:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiosrc.h:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gio/gstgiostreamsrc.h:
gio: Update my mail address and copyright
2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gio/gstgiobasesrc.c:
* ext/gio/gstgiobasesrc.h:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gio/gstgiostreamsrc.h:
gio: General clean up and simplification
The GInputStreams are now requested by a vfunc from
the subclasses instead of relying that the subclass
sets it until it's needed.
This might also fix bug #587896.
2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: keep sending newsegments after seeking
Adder sends with timestamps from 0 upwards. After seeking we need to send
new-segments to get correct positions-queries.
2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
* tests/check/elements/adder.c:
adder: make test more robust
Add audioconverts to the live-seeking test to make it negotiate.
2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: use core performance log category
2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/adder/gstadder.c:
adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
This ensures that collectpads' cookie is properly updated so that when the streaming
threads will restart and be checking for the flushing status of all pads there will
be no inconsistent state.
2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
* ext/pango/gstclockoverlay.c:
pango: Call tzset() before localtime_r()
POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
required to set the state variables that define the current timezone. Indeed,
glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
if the system timezone is changed for a running program between two calls to
gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
timezone equals /etc/localtime being modified.
Fixes bug #587676.
2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
* ext/Makefile.am:
build: remove spurious schroedinger reference
2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
* configure.ac:
* ext/Makefile.am:
* ext/schroedinger/Makefile.am:
* ext/schroedinger/gstschro.c:
* ext/schroedinger/gstschrodec.c:
* ext/schroedinger/gstschroenc.c:
* ext/schroedinger/gstschroparse.c:
* ext/schroedinger/gstschroutils.c:
* ext/schroedinger/gstschroutils.h:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/gstbasevideocodec.c:
* gst-libs/gst/video/gstbasevideocodec.h:
* gst-libs/gst/video/gstbasevideodecoder.c:
* gst-libs/gst/video/gstbasevideodecoder.h:
* gst-libs/gst/video/gstbasevideoencoder.c:
* gst-libs/gst/video/gstbasevideoencoder.h:
* gst-libs/gst/video/gstbasevideoparse.c:
* gst-libs/gst/video/gstbasevideoparse.h:
* gst-libs/gst/video/gstbasevideoutils.c:
* gst-libs/gst/video/gstbasevideoutils.h:
basevideo: send basevideo back to remedial school
Move basevideo classes and schroedinger plugin to -bad.
2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/netbuffer/gstnetbuffer.h:
netaddress: add constant for max len
2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
* gst-libs/gst/netbuffer/gstnetbuffer.h:
netbuffer: add gst_netaddress_to_string
Add function to serialize a net address to a string.
API: GstNetAddress::gst_netaddress_to_string()
2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: make fd:// uri use buffering too
fd:// usually operate in push mode only and are thus suitable for buffering.
2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaybin2.c:
* gst/volume/gstvolume.c:
volume: include "1.0=100%" in property description
2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaysink.c:
playsink: remove unused property defs
2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/multichannel.c:
multichannel: rewrite the new doc comment a bit
Its part of the audio lib.
2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaysink.c:
playsink: Avoid a segfault when the video sink fails to start
Don't attempt to display the subpictures and segfault when the
video sink failed to start (and hence the videochain is NULL).
2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/audio/gstringbuffer.h:
ringbuffer: add vmethod to clear the ringbuffer
Add a vmethod so that subclasses can be notified when they should clear the data
in the ringbuffer.
2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/riff/riff-media.c:
riff-media: Fix the fourcc caps property for VC-1/WMVA
The caps property for carrying fourccs is 'format', not 'fourcc'
2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: include in.h for FreeBSD compat
Fixes #586920
2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstapp.def:
defs: add defs for new appsink buffer-list method
2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
appsink: add docs and signals
Add docs for the new callback.
Add signals for the new buffer-list support.
2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
* tests/check/elements/appsink.c:
Added unit tests for buffer list support in appsink.
2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
* gst-libs/gst/app/gstappsink.c:
Added buffer list support.
2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
* gst-libs/gst/app/gstappsink.h:
Added buffer list support.
2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: Include winsock2.h after defining WINVER.
Similar to bug #587080.
2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Moved a comment.
2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/multichannel.c:
docs: add basic section docs for multichannel and relocate the ones for audio
Add section docs for multichannel, so that it has a short desc in the toc too.
Move the section docs in adio up, so that the follow the copyright like
elsewhere.
2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c:
v4l: open/close device in ready.
Simillar change like in v4l2src. This allows probing feature in paused, where
streaming is noit yet started.
2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
* gst/playback/gstplaysink.c:
playbin2: fix initial volume handling also when reusing the element
This is a follow-up to commit 452988, making it work correctly when the audio
chain is reused.
2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Define WINVER before including any win headers
Fixes bug #587080.
2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
* gst-libs/gst/riff/riff-read.c:
riff: prevent crash if rounded up tag size exceeds data size
When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
and an invalid read past the buffer data follows.
2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/gstbasevideocodec.c:
basevideocodec: By default don't allow caps changes on the srcpad
This fixed playback of Dirac files with schrodec when upstream wants
a different width/height, basevideocodec accepts this and then
pushes buffers with new caps but content of the old caps.
In the best case this will just result in wrong unit size and a
failure in basestransform elements.
2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
* autogen.sh:
autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
Check for more automake command variants. Use printf instead of 'echo -n'
for portability
2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From f810030 to 5845b63
2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstscreenshot.c:
screenshot: don't leak message
2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: lower the h264 typefinder's probability
A NEARLY_CERTAIN is absolutely not warranted given the kind
of things it checks for. Even a LIKELY is probably not entirely
appropriate.
2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
* common:
Automatic update of common submodule
From f3bb51b to f810030
2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for multipart
So we get slightly nicer error messages when multipartdemux is missing.
2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
adder: only unflush when we flushed before
Ass suggested by Stefan Kost:
Keep track of when the sinkpad was set to flushing and unflush the pad when an
upstream flushing seek failed.
2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: fix leak when the source fails to change state
2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/subparse/gstssaparse.c:
ssaparse: avoid leaking all buffers
2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/check/elements/adder.c:
adder: test seek handling in adder
This tests seeking on an adder that has a normal and a live source connected.
Wheter the current behavior is the desired one needs to be discussed still
(see #586033)
2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: pass the xwindow along to not look at the yet unset var.
When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
x(v)imagesink: catch tags and show title in own window
Refactor the code that sets the window title. Catch tag-events and use title
metadata for the window title.
2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
Also make all the function arrays constant.
2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
audiotestsrc: Add support for generating gaussian white noise
This patch adds support for stationary white Gaussian noise.
The Box-Muller algorithm is used to generate pairs of independent
normally-distributed random numbers.
Fixes bug #586519.
2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Fix NV12 and NV21 transformations
Fix some stride problems, fix the nv12 to nv21 direct transformation,
and implement a direct conversion to yuv444 to save CPU.
2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Fix NV12 painting for odd strides/heights
2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/cdparanoia/gstcdparanoiasrc.c:
cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
Finally fixes #531035.
2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/cdparanoia/gstcdparanoiasrc.c:
cdparanoia: try to guess a good cache size if it's set to -1
Try to guess from the paranoia-mode setting whether playback or
ripping is wanted, and use a smaller cache size if we're likely
to be doing playback, to avoid a long startup delay. Since this
was the value used in older cdparanoia versions, it should be
fine in any case. See #586331.
2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
* configure.ac:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.h:
cdparanoia: expose cache size setting
This setting was added in cdparanoia 10.2. The default value is good
for audio extraction, but lower values (previous versions of cdparanoia
used 150) are better for realtime playback.
Fixes #586331.
2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Make build of schro plugin conditional
2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
* win32/common/libgstrtp.def:
basertppayload: add support for bufferlists
Based on patch from Ognyan Tonchev.
See #585559
2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtpbuffer: use new convenience functions
New core convenience functions makes the list getters and setters trivial.
Maybe even too trivial...
2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtp.def:
defs: add new symbol to win32 defs file
Based on patches by Ognyan Tonchev.
See #585559
2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtp: cleanups, add _list_get_seq() too
Clean up the docs a little.
Add missing _list_get_seq method.
Add new symbols to the docs
2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
* win32/common/libgstrtp.def:
rtp: cleanups
Add Since tags to docs
Move some code around
Add win32 symbols
2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c:
rtp: add bufferlist support
2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
rtp: pass data to macros instead of GstBuffer
2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
* win32/common/libgstrtsp.def:
win32: Add gst_rtsp_watch_queue_data() to the exports
Fix the tests by exporting the new symbol from the win32 dlls
2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: appname might be NULL
Don't set title if appname is unknown.
2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: set window title from application name
2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspurl.c:
rtsp: Made the parsing of the RTSP URL scheme more generic.
2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Added gst_rtsp_watch_queue_data().
gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
but allows for queuing any data block for writing (much like
gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
API: gst_rtsp_watch_queue_data()
2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Only extract the session ID from RTSP responses.
2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspurl.c:
rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Improved base64 decoding in fill_bytes().
The base64 decoding in fill_bytes() expected the size of the read data to
be evenly divisible by four (which is true for the base64 encoded data
itself). This did not, however, take whitespace (especially line breaks)
into account and would fail the decoding if any whitespace was present.
2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
audiosrc: fix get_offset
When we need to jump to the most recently captured sample, jump to where the
next sample will be written instead of to some old data.
Fixes #581460
2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
audiosink: free the ringbuffer when going to NULL
Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
audio: correctly handle short read/writes
2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: add some extra logging for buffer timestamps
2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
adder: more seeking fixes.
When a seek failed upstream, make sure the adder sinkpad is set unflushing again
so that streaming can continue.
We only have a pending segment when we flushed.
Set the flush_stop_pending flag inside the appropriate locks and before we
attempt to perform the upstream seek.
Add some more comments.
Use the right lock to protect the flags in flush_stop.
See #585708
2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: Free iterator after removing all groups
2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/gstvideofilter.c:
videofilter: Add a default get_unit_size function
This returns the correct values for all formats that are handled by
GstVideoFormat and makes all the custom get_unit_size functions in
many elements unnecessary.
2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: add Timestamp header field
fixes #585994
2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: set smarter target state on uridecodebin
Set the target state of the newly added uridecodebins to somthing else that
PAUSED so that we keep their state in sync with the playsink state.
Fixes #585268
2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: set the sink flag on the element
2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: add debug message
2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
audiosink, audiosrc: do the class_ref()s in the right class_init functions
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
audiosink,audiosrc: ref the audio ring buffer class and type in class_init
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
audiosrc: return FALSE when receiving a SEEK event
When receiving a seek event, return FALSE as we don't implement seeking.
2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/seek.c:
Don't use deprecated GTK API
Fixes bug #585758.
2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: send flush_stop when seeking failed
At least do the fix to sent the flush_stop when seeking failed to ensure we
keep no pads flushing. before it was send when the seeking worked which is just
plain wrong and was not the intention.
2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Use a more consistent naming of GstRTSPRec variables.
2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Call message_sent() callback for all sent messages.
Previously the messages_sent() callback was only called for messages
which had a CSeq, which excluded all data messages. Instead of using the
CSeq as ID, use a simple index counter.
2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/theora/theoradec.c:
* ext/vorbis/vorbisdec.c:
oggdemux: post/send tags with the container-format tag
For this to work properly, theoradec and vorbisdec need to put
tag events received from upstream into the pending_events list
so they get pushed out after any newsegment event, not before.
2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
* tests/old/examples/seek/cdplayer.c:
Don't use deprecated GTK API
Fixes bug #585758.
2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/adder/gstadder.c:
adder: send flush-stop earlier
When no flush-stop has been sent by upstream, we have to send one ourselves to
continue playback. Do this as soon as the collect function is called instead of
after we possibly pushed segment events (that got then flushed out)
2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: add shuttle controls
2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/stepping2.c:
example: fix compile
2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/Makefile.am:
examples: build the stepping2 example
2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: update for new step API
2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: do reverse seeks more accurate
For reverse seeking with the accurate flag set, try to be more precise by
seeking a little bit after the requested position.
2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstogmparse.c:
* gst/subparse/gstssaparse.c:
* gst/subparse/gstssaparse.h:
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
Make subtitle parsers post a taglist with codec tags, so the application
knows what kind of subtitle a subtitle stream is. Fixes #576552.
2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
ringbuffer: handle border cases in resampler
2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
docs: Update common. Use upload-doc.mak instead of upload.mak
2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
docs: fix typo
2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: reset accum when dropping samples
When we are resampling and we drop samples because we paused, reset the accum
counter because it's now invalid.
2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/video/gstbasevideodecoder.h:
docs: Fix a couple of warnings from the docs build.
2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/testchannels.c:
Don't include config.h multiple times when build audio testchannel app.
Fixes build problem on win32 (#585075).
2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
playbin2/uridecodebin: Fix connection-speed propagation
uridecodebin expects the passed connection-speed value in kbps, so we
need to divide the value stored in bps by 1000. Also, lower the upper
limit on the properties to the value that we can actually store in our
internal guint (which is plenty high enough)
2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/subparse/gstsubparse.c:
* tests/check/elements/subparse.c:
subparse: recognise more subrip timestamp variants
Be even less restrictive in what we accept for .srt timestamps when
typefinding and parsing subrip subtitles and add a unit test for
the 'new' format. Fixes #585197.
2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.h:
rtsp: add some more docs
2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspmessage.c:
rtsp: Avoid a compiler warning.
2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: Updated documentation for GstRTSPResult.
Moved GST_RTSP_ELAST to be last in the documentation to match the actual
enum values.
2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* autogen.sh:
autogen: remove -Wno-portability from here
as it is in configure.ac now.
2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Plug a memory leak.
Free memory related to any partially read and/or written RTSP messages.
2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: no need to cause discont when clipping
Remove the discont-when-clipping hack now that basesink provides us with
correctly clipped samples when stepping.
2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
audiosink: don't align when we clip
Don't align samples when they were clipped. Not entirely correct but better than
nothing for now.
2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/.gitignore:
* tests/examples/seek/stepping2.c:
examples: add stepping example in PLAYING
Add stepping example in PLAYING, audio is a bit distorted because basesink does
not provide good clipping info yet.
2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for hdv/aux-* formats.
2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
* ext/schroedinger/Makefile.am:
Added libgstbase to schro's LIBADD
Fixes #585079
2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstid3tag.c:
libgsttag: don't extract genres from empty ID3v1 tags
If we don't have any other info, don't try to interpret the
genre field. In particular we don't want to interpret a genre
of 0 as 'Blues' if no other fields are set and the entire tag
is just empty.
2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: make sure varargs are of right type
Explicitly cast the variables to g_object_set to their right types.
2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: increase stream probing queues
When we are probing for streams, we want to set the queue size in such a way
that we can scan a maximum amount of data without consuming too much memory.
Therefore, remove the time limit on the queue and only stop scanning after 2MB
of data.
See #584104.
2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Fixed a typo.
2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Remove an unused variable.
2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Removed duplicate initialization of conn->writefd.
2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Use #defined status codes.
2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Correct gen_tunnel_reply().
Prevent gen_tunnel_reply() from generating an incomplete response
in case an error response code is given.
2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/video-enumtypes.c:
configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
See #584835. Also update win32 files while we're at it.
2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: API: Add {audio,video,text}-tags-changed signals
Fixes bug #584686.
2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/vorbis/vorbisdec.c:
vorbisdec: don't put invalid bitrate values into the taglist
Bitrates are stored as 32-bit signed integers in the vorbis
identification headers, but seem to be read incorrectly,
namely as unsigned 32-bit integers, into the vorbis structure
members which are of type long, which makes our check for
values <= 0 fail with files that put -1 in there for unset
values.
2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/.gitignore:
ignore: add new stepping app to ignore
2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/Makefile.am:
* tests/examples/seek/stepping.c:
examples: add stepping example.
Add an example of using playbin2 and frame stepping to simulate variable rate
playback based on a sine wave.
2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.h:
playbin2: also set custom text and subp sinks
Set the custom subpicture and text sinks along with the custom audio and video
sinks when needed.
Fix a little docs blurb too.
2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: add G_LIKELY because we can
2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Fix caps for ogg typefinder.
2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: remove some cruft from -sections.txt file
2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
* tests/examples/seek/seek.c:
add framestepping to playbin2 and seek
2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Avoid compiler warnings with -Wextra.
2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: Remove an unused variable.
2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaybin2.c:
playbin2: Have playbin recognise PGS subpicture streams
Recognise PGS subpicture streams and connect them to the SPU pad
in playsink. Unfortunately this fails badly with negotiation errors
if the SPU is not recent enough to support the stream. I'm not sure
how to add format negotiation in yet.
2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstdecodebin2.c:
* gst/playback/gsturidecodebin.c:
decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix volume handling for audio sinks without "volume" property
When using an audio sink without a "volume" property, volume control
would only work for the first song. For the next song, we'd try to
re-use the existing audio chain, but inadvertently set chain->volume
to NULL instead of to the existing volume element.
2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: cosmetic change to avoid unnecessary line breaks
Looks nicer and works around gst-indent silliness.
2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: don't lose the ref to the volume element
Only release the ref to the volume element when it is controled by a sink. For
software volume we never have to fear that it will change.
2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
playbin2: actually use configured audio/video sinks
playbin2 inadvertently used autoaudiosink and autovideosink up to now,
since it would overwrite the sinks configured via the "audio-sink"
and "video-sink" properties with the stream-specific group sinks when
configuring the outputs. Those are usually NULL however, so that would
overwrite the configured sinks with NULL which makes playbin2 then
default to the auto sinks. Fix this by keeping a reference to each
configured sink in playbin2 and setting up the right sinks depending
on whether there is a stream-specific sink or not.
Fixes #584020.
2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/examples/seek/seek.c:
seek: add volume label and sync with sink volume
Look at the volume and have the pulsemixer open at same time. Unfortunately
playbin2 does not emit notify on volume right, so this polls for now.
2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: remove leftover elements
Remove all of the elements inside decodebin2 when goint to READY and NULL.
Makes decodebin2 reusable.
Fixes #583750
2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2; release refs to volume/mute properties
Release the refs to the volume and mute property elemens before setting the
child elements to READY or NULL.
Fixes #583318
2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/gdp/gstgdppay.c:
gdppay: set caps on outgoing buffers
Set caps on outgoing buffers because NULL caps confuse basetransform.
Fixes #583867
2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
netbuffer: also note the order of IP4 addresses
IP4 addresses are also stored in network byte order. Make a note of this in the
docs.
2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
* ext/theora/theoraparse.c:
theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
We now require GLib 2.16.
2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
Update common
2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
netbuffer: document that the port is network order
Document the fact that we store the port number in network order in
GstNetAddress and that the caller should byteswap appropriately.
2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* gst/videoscale/vs_image.c:
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.c:
* gst/videoscale/vs_scanline.h:
videoscale: Add support for 16 bit grayscale in native endianness
2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
videotestsrc: Add support for 16 bit grayscale in native endianness
2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
add can-activate-pull property to baseaudiosink
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
to baseaudiosink.
2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix boundary case for seeking.
When we have exactly 0 bytes left to search, make sure we stop instead of going
into an infinite loop.
2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/cdda/sha1.c:
* gst-libs/gst/cdda/sha1.h:
cddabasesrc: Remove copy of sha1 digest
Remove our copy of sha1 digest now that we depend on glib 2.16.
Fixes #536313
2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
* gst-plugins-base.spec.in:
Update spec file
2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/gstbasevideodecoder.c:
* gst-libs/gst/video/gstbasevideoparse.c:
* gst-libs/gst/video/gstbasevideoutils.c:
* gst-libs/gst/video/gstbasevideoutils.h:
* win32/common/libgstvideo.def:
video: don't expose internal gst_adapter_get_buffer() helper function
If it's really needed it should go into GstAdapter in core.
2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/gstbasevideodecoder.c:
basevideo: Fix memleak
2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
* ext/schroedinger/gstschrodec.c:
* ext/schroedinger/gstschroparse.c:
schro: Fix usage of adapter_masked_scan_uint32
Because *somebody* changed the API without telling me.
2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
* ext/schroedinger/gstschro.c:
schro: Change package name to GST_PACKAGE_NAME
2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/gstbasevideoencoder.c:
basevideo: Add preset interface to encoder
2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
* gst/audioresample/gstaudioresample.c:
Run liboil benchmark multiple times
The statistics function requires multiple runs, otherwise
it causes a divide by zero error.
2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* m4/gst-fionread.m4:
m4: fix 'suspicious cache value' warning for gst-fionread.m4
.. here as well (should really be moved to common, but I'm too lazy).
2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/vorbisdec.c:
vorbisdec: detect and report errors better
Check the return values of a couple more libvorbis functions and post an error
when something is wrong instead of continuing and crashing.
2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaysink.c:
playbin2: fix initial volume and mute handling
Use two flags to remember volume/mute changes at times when we don't have the
audiochain yet (e.g. construction). Only set values when they were actualy
changed. This makes pulseaudio's stream restore functional.
2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From d3a8fab to 888e0a2
2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
* win32/common/libgstvideo.def:
win32: Remove gst_adapter_masked_scan_uint32 from the exports
2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
audiosink: improve debug message
2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
* gst-libs/gst/tag/gstid3tag.c:
gstid3tag: Don't extract a track number unless present.
In ID3v1, a track number is present only if byte 125 is null AND
byte 126 is non-null. If the track number is not present, don't add
a track number tag with value 0.
2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/video/gstbasevideoutils.c:
* gst-libs/gst/video/gstbasevideoutils.h:
videoutils: remove adapter methods
Remove adapter methods now that they are in core.
2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstvideo.def:
defs: add new symbols
2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
autogen: pass -Wno-portability to automake to suppress warnings
GNU make is needed.
2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/.gitignore:
gitignore: remove bogus *.sgml wildcard - these files are tracked in git
2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/tcp/gsttcpclientsrc.c:
tcpclientsrc: this is not a live source
Don't mark us as a live source because we are not.
2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: only send flush_stop when seek failed
This is still not the ultimate fix. Added some comment to explain the troubles.
2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
audiosink: return the return value of wait_preroll
Return the value that _wait_preroll() returned instead of always WRONG_STATE.
2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
adder: send flush_stop to match flush_start
Adder was relying that something else sends a flush stop. When using adder with
a livesource it was not getting a flush_stop and thus all pads downstream where
keept flushing. Mark a pending flush_stop and send it when we are working on
the new segment back in the streaming thread.
2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/examples/seek/seek.c:
seek: ui improvements
Repaint the window black on expose, as this looks nicer when resizing or using
the expander. Also show time after slider, as this saves a whole line (nice on
small displays).
2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstdecodebin.c:
decodebin: use iterators instead of list
The list api is deprecated. Use threadsafe iterators instead.
2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: configure caps on decodebin2
Implement the caps property by setting the configured caps on new decodebin2
objects.
Fixes #582749
2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: avoid some _caps_ref in some cases
Only mess with the caps refcount when we configure different caps.
2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: fix potential caps leak
Free the user-configured caps in finalize.
2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: add queue after cdda://
Add a queue2 after the raw output pads of certain sources such as those for uris
like cdda://
No tuning of the queue is done yet as the defaults seem to work fine for me.
Fixes #582528
2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: don't loop when at EOS
When we try to read the last page, don't try to read past the upper boundary, as
this might cause endless loops.
See #582942
2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/audioresample/gstaudioresample.c:
audioresample: Don't drain remaining buffers after a flush.
If we were resetted (due to a flush), we can not drain the remaining
buffers since they would be pushed before a valid new newsegment event.
2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
* ext/theora/theoradec.c:
theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: add more logging and return value checking
2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: handle the return value from iterator_fold
2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: use the pad in logging as objects
Helps to differenciate between source and sinks pads.
2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
* tests/examples/seek/seek.c:
seek: use parser for mp3 and rename variable
2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: add playbin2 options in expander
Add the playbin2 stream selection options inside an expander to preserve some
space on screen.
2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add support for v210 and v216 formats
2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/gstbasevideocodec.c:
* gst-libs/gst/video/gstbasevideodecoder.c:
* gst-libs/gst/video/gstbasevideoencoder.c:
* gst-libs/gst/video/gstbasevideoparse.c:
video: remove // comments
2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add Y444, v210, v216 formats
2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
* configure.ac:
* ext/Makefile.am:
* ext/schroedinger/Makefile.am:
* ext/schroedinger/gstschro.c:
* ext/schroedinger/gstschrodec.c:
* ext/schroedinger/gstschroenc.c:
* ext/schroedinger/gstschroparse.c:
* ext/schroedinger/gstschroutils.c:
* ext/schroedinger/gstschroutils.h:
schro: Move schro plugin from Schroedinger
Previous history is in Schroedinger. Depends on, and is an example
of using, GstBaseVideo* base classes.
Code was reindented, and an #ifdef HAVE_ENCODER removed.
2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/gstbasevideocodec.c:
* gst-libs/gst/video/gstbasevideocodec.h:
* gst-libs/gst/video/gstbasevideodecoder.c:
* gst-libs/gst/video/gstbasevideodecoder.h:
* gst-libs/gst/video/gstbasevideoencoder.c:
* gst-libs/gst/video/gstbasevideoencoder.h:
* gst-libs/gst/video/gstbasevideoparse.c:
* gst-libs/gst/video/gstbasevideoparse.h:
* gst-libs/gst/video/gstbasevideoutils.c:
* gst-libs/gst/video/gstbasevideoutils.h:
video: Copy BaseVideo classes from Schroedinger
2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
* gst/tcp/gstmultifdsink.c:
multifdsink: add num-fds property
multifdsink::num-fds
2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/vorbisenc.c:
vorbisenc: Implement Preset interface
2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/theora/theoraenc.c:
theoraenc: Implement Preset interface
2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: Implement Preset interface
2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaysink.c:
playbin2: Fix cdda:// playback
Don't send async-start when the playsink has already been configured
before changing state.
2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: require core CVS for gst_adapter_prev_timestamp()
which is used in the libvisual plugin.
2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* AUTHORS:
AUTHORS: fix my email
2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c:
audioclock: make our internal time monotonic
Make the internal time increase monotonically.
2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
visual: remove next_ts variable
We can remove the next_ts variable as we don't use it anymore.
2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
visual: use new adapter timestamp code
Use the new adapter timestamp tracking code to make things easier and produce
vastly better output timestamps.
2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* po/Makevars:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: avoid conflicts of local *.po files with files in git
Make it so that filenames and line numbers are only stored in the *.pot file
(which is not in git), but not in the individual *.po files. This information
is hardly useful for translators in our case, and it should avoid the constant
conflicts of local *.po files with the ones in git which are caused by the
source files changing and the line numbers being updated. This commit might
cause one last merge conflict for you, which you can work around with
"git checkout po/*.po" before merging or pulling. After that there should
(hopefully) not be any more local modifications of these files (unless
someone committed additions or changes to translated strings and the
*.po files haven't been updated yet, that is).
2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/elements/.gitignore:
* tests/check/elements/audioresample.c:
tests: fix audioresample unit test on big endian architectures
Don't hardcode endianness=1234 in the filtercaps, it will cause
pad link failures which will result in the test timing out.
2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: fix broken enum nick - it should have a hyphen
The enum nick should be 'sine-table', not 'sine table'. Technically this is
an API/ABI change I guess, but anyone who was using this and didn't report
it deserves this.
2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: seek to the requested byte offset, not the expected byte offset
2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
audiotestsrc: support more than just one channel
2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/propertyprobe.h:
propertyprobe: Fix typo in the docs
2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/theora/theora.c:
* ext/vorbis/vorbis.c:
Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: handle invalid timestamps better
Handle buffers with -1 timestamps better by keeping track of the en time of the
previous buffer and assuming the -1 timestamp buffer goes right after the
previous one.
when we have two buffers that are equally good, output the oldest buffer once to
minimize latency.
don't try to calculate latency when the input framerate is unknown.
2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggmux.c:
oggmux: small debug statement in DISCONT
2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
oggdemux: fix abuse of ogg API, handle broken oggs
When we feed the ogg sync layer, we need to feed it contiguous data even if the
sync layer did not consume all of it yet. This makes sure that it always finds
the next page even for more corrupted files. Use a different read_offset for
this purpose. since we now keep track of the sync layer, we don't have to reset
after finding a start of a page.
Add some more debug info for the error paths.
Only reset the sync layer when we perform a seek operation.
Avoid failure when the next chain has no bos pages but instead simply ignore it.
when we receive unknown page serial numbers mid stream, don't fail but post a
warning and hope that we get back on track later.
Fixes #579642
2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: make subpictures a raw output format
Subpictures are a raw format, we want those pads exposed so that playbin2 can do
the subpicture mixing.
2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
rtpdepay: add some more comments
2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c:
audioclock: make sure values are ever increasing
2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: make fallback identity silent
Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
element so that it consumes less CPU.
2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
playbin2: handle custom audiosinks differently
Keep track of the autoplugged custom sinks and configure them in the playsink
element when we have collected all streams.
Also make sure that we only select one custom sink.
When unreffing the internal sink, we don't need to change the state to NULL.
2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2: unify custom sink get/set functions
Use one function to set/get all of the different sink types.
cleanup up the subpicture chain too.
Allow setting a custom subpicture sink.
2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/tunernorm.h:
interfaces: Seperate some more struct definitions from typedefs
2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/navigation.h:
* gst-libs/gst/interfaces/videoorientation.h:
* gst-libs/gst/interfaces/xoverlay.h:
interfaces: Seperate some more struct definitions from typedefs
2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* win32/common/libgstinterfaces.def:
Add new functions to win32 exports
2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
Add new functions to the docs
2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
interfaces: API: Add gst_mixer_get_mixer_type()
This is a convenience function that returns the mixer_type
of the interface struct.
2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/colorbalance.c:
interfaces: Add docs for gst_color_balance_get_balance_type()
2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
* autogen.sh:
Run libtoolize before aclocal
This unbreaks the build in some cases. Fixes bug #582021
2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
textrender: Correctly initialize the background for ARGB too
2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
* ext/pango/gsttextrender.h:
textrender: Use libgstvideo functions to create caps
Also check if downstream wants ARGB always when we get
new caps.
2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
textrender: Don't always use ARGB if downstream supports it but take it's preference
2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
* ext/pango/gsttextrender.c:
* ext/pango/gsttextrender.h:
textrender: Add support for ARGB and alignment properties
Fixes bug #581571.
2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextrender.c:
textrender: Add ; after GST_BOILERPLATE to fix indention
2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: made mp3_type_find less aggressive
mp3_type_find could suggest already when only a single valid header
was found, if it ran out of data before the end of the next frame.
Therefore, ignore the last found frame if it was incomplete.
Fixes bug #579692.
2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: Store cover art in vorbiscomments
Fixes bug #513373.
2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
interfaces: API: Add gst_color_balance_get_balance_type()
This is a convenience function that returns the balance_type
of the interface struct.
2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.h:
* gst-libs/gst/interfaces/tunerchannel.h:
interfaces: Separate struct definitions from typedefs
2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* pkgconfig/gstreamer-app-uninstalled.pc.in:
Fix libdir for uninstalled gstreamer-app library
2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add description for APE tag caps
2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump core requirement to last release
as that's more likely to be true than that we need
only 0.21.1.
2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
* configure.ac:
configure: rename CVS -> git in a couple of places
2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump GLib requirement to GLib >= 2.16
as per the New Regime (see wiki).
2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: cache events from upstream and re-send them once we have a source pad
Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
Fixes #580318.
2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
* gst-libs/gst/riff/riff-media.c:
riff: support UYVY raw 4:2:2 in riff.
2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
Back to development -> 0.10.23.1
2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
* ext/theora/theoradec.c:
theoradec: fix buffer overrun on 422 decode.
2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
* ext/theora/theoradec.c:
theoradec: 444 support.
2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
* ext/theora/theoradec.c:
theoradec: handle 422 images (as YUY2).
2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
theoradec: rearrange code in preparation for 422 and 444 support.
=== release 0.10.23 ===
2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 0.10.23
2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.22.6 pre-release
2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix resume after pause
Don't ignore the state change of the children, they might be doing an ASYNC
state change.
2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.22.5 pre-release
2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcp-marshal.list:
multifdsink: fix signature of the add-full signal
The second parameter is a GstSyncMethod enum, not a boolean.
2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: initialize variable too
2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: make playsink go ASYNC to PAUSED
Make playsink go async to the PAUSED state instead of relying on uridecodebin
for async behaviour in playbin. This solves some problems (mainly with DVD)
where the pipeline would go to PLAYING before preroll completed, failing to
select the audiosink clock.
Fixes #581727
2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.22.4 pre-release
2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisenc.c:
vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
With vorbisenc, compute the granulepos with running time and clip incoming
buffers to segment.
With theoraenc, drop out of segment buffers.
2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst/audioresample/gstaudioresample.c:
audioresample: Fix buffer size transformations
When calculating the input/output buffer sizes in the transform_size function,
take the number of channels into account, so we don't end up calculating
a buffer size that only contains a partial number of audio frames.
Also, when going from output size to input size, round down rather than
up, so as to calculate the minimum number of samples that *might* yield
a buffer of the intended destination size.
Fixes: #580470 and #580952
2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
* ext/vorbis/gstvorbisenc.h:
* ext/vorbis/vorbisenc.c:
vorbisenc: Ensure output buffers fall within the segment
Add the start position of the first segment to the running time
used to generate buffer timestamps in vorbisenc. This avoids generating
buffers which fall outside the initial segment. The element segment
handling requires more extensive fixing, but this at least prevents
regressions. Fixes: #580020
2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
* gst-libs/gst/audio/gstbaseaudiosink.c:
Revert "add can-activate-pull property to baseaudiosink"
This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
* gst-libs/gst/audio/gstbaseaudiosink.c:
Revert "[baseaudiosink] add docs for can-activate-pull"
This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
[baseaudiosink] add docs for can-activate-pull
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
can-activate-pull.
2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
add can-activate-pull property to baseaudiosink
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
to baseaudiosink.
2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
videorate: clear discont on duplicated buffers
When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
the first pushed buffer but fails to clear it for subsequent buffers. This
causes theoraenc!oggmux and possibly other elements to consider this a discont
stream.
Fix videorate to produce discont as the first buffer and after a flushing seek.
Fixes #580271.
2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/Makefile.am:
check: Disable the playbin2 for this release, as it is a bit racy.
Disable the test, as per the discussion in #580120. Needs re-enabling
after the release, when playbin2 is fixed.
2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
The 2s limit is way too small for a lot of files (which have an interleave
in time of between 3 and 5s). Instead, leave it to the initial 5s value
and reduce the other limits (allowing us to stay memory-efficient).
2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.22.3 pre-release
2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
* gst/audioresample/gstaudioresample.c:
audioresample: Fix unused variable in compilation with --disable-gst-debug
Fixes: #579668
2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From b3941ea to 6ab11d1
2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybasebin.c:
playbin: only use raw_decoding_mode when it's true
First check the pad caps if they are raw before setting the raw_decoding_mode to
TRUE. Fixes playback of transport streams and other streams that require large
queues.
Fixes #579734
2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/cdda/gstcddabasesrc.c:
* tests/check/libs/cddabasesrc.c:
cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes #579463.
2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: don't use GLib-2.16 API, we require only 2.14
Fixes #579267.
2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: don't unparent the ringbuffer
when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c:
RTCP: don't fail when retrieving invalid PT
We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.
Fixes #579192.
2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/app/gstappsink.h:
* gst-libs/gst/app/gstappsrc.h:
app: add trivial cast macros
Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
and add the macros to the standard macros in the docs.
Fixes #579130
2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
pkgconfig: add the app/ directory to Libs
Add the appsrc/appsink directory to the Libs in the uninstalled
pkgconfig file so that one can build against it.
Fixes #579129
2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
0.10.22.2 pre-release
2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
ChangeLog: regenerate changelog with the gen-changelog script
2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: Update po files from TP
2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/interfaces-enumtypes.c:
* win32/common/interfaces-enumtypes.h:
* win32/common/video-enumtypes.c:
win32: Update win32 build files
2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/libs/video.c:
check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/elements/playbin2.c:
check: Fix the input uri in playbin2 test.
Don't try and use a random file in wim's home directory as a test input
2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.h:
video: Fix typo in the docs
2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add support for YVYU YUV colorspace
2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* gst-libs/gst/fft/gstfft.c:
docs: fix hyperlink and move fft attribution to the right place
2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstbaseaudiosink.c:
log: use G_GUINT64_FORMAT instead of llu
2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
RTSP: add missing headers for WMS RTSP
Add missing headers related to Windows Media RTSP extension.
Fixes #578942
2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
* docs/design/draft-keyframe-force.txt:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theoraenc: implement upstream keyframe force
Implement handling of upstream keyframe forcing.
Update the design documents too.
Fixes #578656
2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
* ext/theora/theoraenc.c:
theoraenc: factor out keyframe forcing
See #578656
2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* AUTHORS:
* gst-libs/gst/fft/gstfft.c:
Give credit to Mark Borgerding (kissfft author)
and add myself to AUTHORS as well. Fixes #575638.
2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
multifdsink: add property to resend streamheaders
Adds a new property in multifdsink, resend-streamheader.
If this property is false, the multifdsink will not send the streamheader if
there's already one set for a particular client.
There are some formats in which every stream needs to start with a certain
blob, but you can't inject this blob at leisure. If the producer wants to
change the blob in question and sets in as the streamheader on the outgoing
buffers' caps, new clients of multifdsink will get the new streamheader, but
old clients will break, because they'll see the blob in the middle of the
stream.
The property is true by default, so existing code will not see any difference.
Fixes #578118.
2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
multifdsink: add property to handle client write
Add a property to disable listening to client writes. This property is usefull
when other code will deal with reading from the client socket.
API: GstMultiFdSink::handle-read property
2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
* gst-libs/gst/rtp/gstrtcpbuffer.h:
* win32/common/libgstrtp.def:
RTCP: add beginnings of Feedback messages
Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610.
2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: clear the target
Clear the target of our ghostpads before we remove the pad from the element.
This to make sure that the internal pad is not left linked to whatever pad we
were ghosted to. This should only be a problem when we leak the ghostpads.
Also release our subpicture pads.
Fixes #577288.
2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
* sys/ximage/ximagesink.c:
ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
Fixes #570768.
2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
baseaudiosrc: adjust the internal timestamp
Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: use new clock time methods
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.
When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
* win32/common/libgstaudio.def:
audioclock: add methods for the internal offset
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().
Add a debug category and some debug lines to the audio clock.
API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: fix up the debugs and warnings
Use _OBJECT variants because we can. Go over some log statements and put them in
the right category.
Fixes #567740.
2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
* gst/tcp/gstmultifdsink.c:
multifdsink: fix error in sync-method
Multifdsink did not handle sync-method=latest-keyframe correctly when the
soft-limit is set to -1 (unlimited).
Fixes #578583.
2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: use the internal clock time
We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: handle clock-lost messages
When we receive a clock-lost message we need to pause and play to select a new
clock.
2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/elements/playbin2.c:
check: add a unit test for playbin2
Add unit test for playbin2 and include the refcount test in #577794.
2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix refcounting of visualisations
See #577794.
2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playsink: fix refcounting of custom elements
Sink the custom sinks, let other elements we create be sunken by the bin we add
them to.
Fixes #577794.
2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/appsink.c:
check: fix appsink test
Fix the appsink test now that the method signature changed.
2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: handle missing input-selector
Gracefully degrade and disable stream selection when input-selector is
missing.
2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
appsink: make callbacks return GstFlowReturn
Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
errors can be reported properly.
Fixes #577827.
2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/audio/gstringbuffer.h:
ringbuffer: allow for custom commit functions
Allow subclasses to override the commit method.
2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
baseaudiosink: fix a small glitch after pause
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaysink.c:
playbin2: better error message on sink failure
If we could create the sinks, but the don't work, don't send the missing plugin
message and report that the state-changed failed.
2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstaudiofilter.c:
audiofilter: don't leak pad-template
gst_element_class_add_pad_template() does not take ownership.
2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
* common:
Automatic update of common submodule
From d0ea89e to b3941ea
2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/interfaces/navigation.c:
* sys/v4l/v4lsrc_calls.c:
navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
* ext/theora/theoradec.c:
theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
This fixes most seeking issues when used with gnonlin.
Fixes #543591
2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From f8b3d91 to d0ea89e
2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
playbin2: don't leak selector when getting current stream numbers.
2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: use fully qualified urls when using a proxy
Use a fully qualified url when specifying the url for tunneled requests through
a proxy.
See #573173
2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/navigation.c:
* gst-libs/gst/interfaces/navigation.h:
* tests/check/Makefile.am:
* tests/check/libs/.gitignore:
* tests/check/libs/navigation.c:
* win32/common/libgstinterfaces.def:
navigation: Extend the navigation interface
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
playbin: Add simple 'raw decoding mode'.
Raw decoding mode removes almost all buffering in video and audio queues
when a source providing already decoded video/audio is detected, on the
possibly bogus assumption that such a source should provide sufficient
internal queueing. Fixes playback on some DVDs, and improves it
on all.
2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/elements/.gitignore:
ignores: Ignore the videoscale check binary
2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
* win32/common/libgstrtsp.def:
win32: Add gst_rtsp_connection_set_proxy to the win32 exports
2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/alsa/gstalsamixer.c:
alsamixer: don't forget to release locks in a few places
Might fix #576585.
2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
videoscale: Don't read over line ends when taking the last Cr or Cb
2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
videoscale: Don't write to few pixels and don't mix Cr and Cb
Fixes bug #577054.
2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
* tests/check/elements/audioresample.c:
audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
If one side has a preference for a particular sample rate or set of sample rates, we
should honour this in the caps we advertise and transform to and from, so that elements
actually know about the other side's sample rate preference and can negotiate to it
if supported. Also add unit test for this.
2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstplaybin2.c:
docs: add a blurb about redirect messages to playbin2 docs
2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: fix little typo in the comments
2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
People might queue messages from a thread other than the thread in which
the main context which this watch is attached is iterated from, so use
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
over list nodes just freed in the other thread. This just fixes issues
I've had with gst-rtsp-server. We might need more locking in various
places here.
2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspmessage.c:
rtsp: clear the entire builder structure
And use structure instead of variable with sizeof when
clearing the rtsp message structure, for clarity.
2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
docs: fix typo in gst_rtsp_message_unset() API docs
2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: add support for proxies
Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
format the colorkey depending on xcontext->depth. This is what they will use to
interprete the value. The max_value in turn is usualy a constant regardless of
the depth.
2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtsp/gstrtspmessage.c:
rtsp: reset whole message (was sizeof pointer instead of sizeof type)
2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/interfaces/mixer.c:
doc: Fix a typo in the GstMixer docs
2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_scanline.c:
videoscale: Fix linear scaling for one byte components
Fixes bug #577054.
2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
videoscale: Fix 4tap scaling of YUYV and friends
2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_image.c:
* gst/videoscale/vs_scanline.c:
* gst/videoscale/vs_scanline.h:
videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
Partially fixes bug #577054, there's just one issue left now.
2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/elements/videoscale.c:
videoscale: Add some more unit tests
2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Use bilinear instead of 4tap scaling for heights < 4
Partially fixes bug #577054.
2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_scanline.c:
videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
This case is for upscaling a frame with width=1
Partially fixes bug #577054.
2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_scanline.c:
videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
Partially fixes bug #577054.
2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: Initialize buffer memory with zeroes
This prevents valgrind warnings when accessing the "x" parts
of xRGB and friends in other elements that handle (and can handle)
xRGB like ARGB (for example videoscale).
2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/elements/videoscale.c:
videoscale: Add a lot of unit tests
2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videocale: Add support for video/x-raw-gray with bpp=depth=8
2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Add support for v308 YUV colorspace
2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
videoscale: Add my copyright to the 4tap scalers
2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
videoscale: Enable 4-tap scaling for all supported formats
2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
videoscale: Implement 4-tap scaling for RGB565 and RGB555
2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
videoscale: Implement 4-tap scaling for UYVY
2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
videoscale: Implement 4-tap scaling for YUY2 and YVYU
2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
videoscale: Implement 4-tap scaling for RGB and BGR
2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pango/gsttextoverlay.c:
textoverlay: Fix drawing of UYVY text borders
2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
textoverlay: Add support for UYVY colorspace
Fixes bug #378094.
2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: do some more cleanup
Free the groups when we go to READY.
Allow for NO_PREROLL elements.
2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: start CSeq counting from 1 instead of 0
Start counting from 1 instead of 0 as this is what most other clients
seem to do.
2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: add ETag and If-Match headers
Add new headers, we need them for RealMedia support.
2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: scale the colorkey components in case of 16bit visuals
Use a default that won't be scales to 0,0,0
2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
audiosrc: improve 'Dropped n samples' warning message
2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
examples: use new method to set flags
Use the new core method for setting object enum properties by name.
2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2: add more support for subpictures
2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2: first support for subpictures
Add beginnings of subpicture support.
2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: print tags from the different tracks
2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: blacklist subpictures for now
Blacklist the subpictures until we add support for them.
Add some small debug info.
See #576408.
2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: expose more media types
Expose more media types from a raw source, such as the subpicture and various
text pads.
Small cleanups and add some more debugging.
See #576408.
2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: rescan audio sinks for volume/mute
Rescan the audio sinks for the mute and volume properties.
fixes #576180.
2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix reuse of the video chains
When reusing playbin with visualisations, reset the async property on the video
sink because some sinks might dynamically recreate their sinks.
Fixes #576188
2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: allow dynamic swtiching of subtitles
When we have the textpad configured, enable and disable the subtitles by setting
the silent flag on the overlay element instead of trying to remove elements.
See #576187
2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/playbin-text.c:
tests: print some more info in the text example
Print both the position and the running_time when the subtitle becomes available
in the application.
2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix dynamic switching of visualisations
Fix the switching of visualisations by requesting and releasing the tee request
pads on demand.
See #576187.
2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/tcp/README:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
docs: add examples for tcp elements, also use correct section name. Fixes #564139
Updated the examples in the README to actually work. Add them to api docs. Tests
the api-docs and fix the section names to make the docs actualy show up.
The example for "tcpserversrc" needs review (might be an element bug).
2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/videoscale/gstvideoscale.c:
indent: fix damange that gst-indent did some time ago
2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix linking order
Link after doing the state change and unlink before shutting down. Makes the
window for causing races in toggling the visualisations smaller.
See #576187.
2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
uridecodebin: reset counter
reset the number of pending dynamic operations back to 0 when we reuse
uridecodebin.
Fixes #576190
2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
* ext/theora/theoradec.c:
theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
The problem was that previously we didn't check whether _theora_granule_frame
returned a negative framecount or not, resulting in bogus timestamps.
2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
* ext/vorbis/vorbisenc.c:
vorbisenc: Set caps on non-header ouput buffers.
Fixes #576142.
2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/seek/seek.c:
seek: Add some more debug
Add some more info about the selected streams.
2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: a pad starts out being not drained.
Mark a new pad as not drained until we get EOS on it.
2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
* gst/playback/gstqueue2.c:
win32: fix seeking in large files
Fix Seeking in large files by using the 64-bit seek functions.
Fixes #576019
2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: recover from failing to add a pad
When we cannot add a pad to the decodebin2 for some reason, print a warning but
continue adding the remaining pads.
2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: more cleanups and docs.
Add some more comments and use g_list_prepend().
2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: refactoring and race fixes
Refactor some code so that we can take the right locks and in the right order.
Fixes quite a bit of races already.
2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: remove the group cond + cleanups
Remove the group GCond that we used for waiting for groups to finish because we
use pad blocking on the selectors and counters instead for waiting for the
groups to complete.
remove the obsolete about_to_finish variable set while emiting the
about-to-finish signal and fix some old comments.
We don't need to take the playbin lock when querying the uridecodebin.
2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/playbin-text.c:
icles: print better error and warning messages
--
2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspbase64.c:
* gst-libs/gst/rtsp/gstrtspbase64.h:
rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
This also fixes another instance of CVE-2008-4316.
2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: report -1 for duration in push mode
In push mode we must return TRUE from the duration query with a value of -1
meaning that we know that we don't know the duration.
2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: add extra dynamic ref for demuxers
When we make a group connected to a demuxer, keep an extra dynamic refcount for
the group which is only decremented when no_more_pads or a multiqueue overrun is
detected. This way we avoid a race between exposing the group while more dynamic
refs are added from new pads.
Fixes #575588.
2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: sync state of the sink correctly
Sync the state of the newly added chains to the state of the parent sink element
to avoid lost async-start messages. Fixes cdda:// async-done message storm.
2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: return NOT_LINKED for unselected streams
When streams are not selected in the selector, return NOT_LINKED so that
upstream elements can skip decoding. Only do this for audio and video pads
because for text streams the overhead is smaller and they could come from
external files.
2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin: set custom text sink properties
Set the custom sink async=FALSE to not make it participate in preroll because we
are dealing with sparse streams.
Try to set sync=TRUE on the custom text sink.
2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/playbin-text.c:
example: use appsink instead of fakesink
Use appsink instead of fakesink to get the subtitles.
Make things more pretty.
2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/playbin-text.c:
examples: add example of intercepting subtitles
Add an example of how to install a custom sink for receiving subtitles in
playbin2.
2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/appsink.c:
tests: fix include in the appsink test
Fix dist by doing the right include.
2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: don't try to set invalid stream numbers
Fix a problem with setting the stream numbers because we check for the wrong
range.
See #575239.
2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: release the shutdown lock
Release the shutdown lock when we wait for other groups to complete or else we
have a deadlock when the other group completes and tries to grab the shutdown
lock.
Fixes #575550.
2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
* tests/examples/app/appsrc-stream.c:
* tests/examples/app/appsrc-stream2.c:
examples: fix g_object_set() value type.
Make sure we cast the length value as a gint64 to the vararg g_object_set() just
incase sizeof(gsize) != sizeof(gint64).
2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: make flac typefinder return lower probability for frame headers
The flac frame header typefinder overstates the likelihood of a match, leading
to false positives with e.g. aac streams and PDF files. Reduce probabilty
returned from LIKELY to POSSIBLE for the frame header matchin code.
Fixes #574939.
2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: improve image/bmp typefinder
Detect more variations and also bail out in more cases where the values
don't make sense. Furthermore, add width/height and bpp to the caps,
because we can.
2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
* tests/check/Makefile.am:
check: Ignore alsamixer in the states test too
2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
* sys/v4l/v4l_calls.c:
v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: fix resolving of hostnames
We were returning a pointer to a stack variable with the resolved hostname,
which doesn't work.
return a copy of the resolved ip address instead.
Fixes #575256.
2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/vorbisparse.c:
vorbisparse: be smarter when queueing headers
Look at the first buffer byte to see if a buffer is a header instead of counting
packets.
2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/gsttheoraparse.h:
* ext/theora/theoraparse.c:
theoraparse: be smarter when queuing headers
Look at the first byte of the buffer data (if we can) to decide if the packet is
a header packet or not instead of counting packets.
2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: add some debug info
Add some debug info to log when the seek worked.
2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
appsrc: release lock in _eos flushing case
Release the mutex when we are flushing in gst_app_src_end_of_stream()
Fixes #574964.
2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
* ext/vorbis/vorbisdec.c:
vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
* ext/theora/theoradec.c:
theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
playbin2: fix raw elements like cdda://
Fix a fixme with a one liner and make cd playback work again.
2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playbin2: improve subtitle handling
Add property to playbin2 to configure a custom sink that receives the raw
subtitle buffers instead of using a textoverlay.
Improve the property finding code to make it more usable.
Use property find code to find async properties in custom sinks that are bins.
Improve text overlay code to gracefully handle missing elements.
2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: Protect memory allocation calculation from overflow.
Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
* gst-plugins-base.spec.in:
Spec: fix up deps
2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: fix parsing of the timeout parameter
--
2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
rtsp: fix g_return condition
when parsing a data message, we require a data message.
2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefinding: flac typefinder fixes
Use scan context for initial peek as well. Peek 6 bytes in the initial
peek rather than 5 bytes, to match the length of the memcmp we're doing
on that data later. Return immediately when we found caps from looking
at the beginning of the data - no point in continuing to scan the next
64kB for something matching a frame header.
2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
rtsp: free the right string.
Free the key value before we remove the header item from the array. The item we
retrieved from the array is only valid until we remove it from the array.
2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: keep track of amount of decoded bytes
Keep track of the actual amount of decoded bytes, which can be less than 3 when
we decode the last bits of a base64 message.
2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: log details in getcaps like in setcaps
2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/MANIFEST:
win32: update MANIFEST, fixing 'make dist'
2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From 7032163 to f8b3d91
2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
* gst/typefind/gsttypefindfunctions.c:
typefind: add photoshop typefind functions
Add photoshop typefind functions.
Fixes #574516.
2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin2: only remove pads that were added
Flag pads that were added so that we can see if we need to remove them later or
not.
2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.c:
rtsp: only add ports when not using TCP
Only add the port numbers in the transport string when we are using udp or
multicast.
2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
rtsp: use gstreamer dump mem
--
2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: use glib base64 encoder
--
2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
Unblock blocked ghostpads when shutting down. Fixes #574293.
2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
Riff: Add mapping for Fraps video codec.
Found through insanity testrun. Confirmed mapping in libavformat.
2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
riff: Add the 'DVR ' mapping for mpeg2video.
Found this in 3 files from the insanity suite and mapping is also present
in libavformat.
2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: Use the proper data pointer instead of poking random memory.
2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: fix compilation on windows.
Remove unused variable when building for windows.
Fixes #574443.
2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From ffa738d to 7032163
2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 3f13e4e to ffa738d
2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 3c7456b to 3f13e4e
2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 57c83f2 to 3c7456b
2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/theoradec.c:
theoradec: parse and use codec_data in the caps
Parse the codec_data in the caps and use this as the headers.
Fixes #574169.
2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: add theora mapping
Add theora mappings. See #574169.
2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
* win32/common/libgstrtsp.def:
rtsp: Add methods for getting the read/write fds
API:gst_rtsp_connection_get_readfd()
API:gst_rtsp_connection_get_writefd()
2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* Makefile.am:
* win32/common/audio-enumtypes.c:
win32: indent copied *-enumtypes.c files in make win32-update
2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/MANIFEST:
win32: update MANIFEST
2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* win32/common/config.h:
win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/interfaces-enumtypes.c:
* win32/common/multichannel-enumtypes.c:
* win32/common/pbutils-enumtypes.c:
* win32/common/video-enumtypes.c:
* win32/common/video-enumtypes.h:
win32: update windows files via make win32-update
Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
which fixes the build of pbutils on windows (#574319).
2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
gitignore: ignore more
2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fix build on Mac OS X
2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstdecodebin2.c:
decodebin2: don't stay connected to notify::caps after negotiation
Disconnect the notify::caps signal in our callback (it'll be re-added
if we're not, in fact, finished getting complete caps). Ensures that
caps changes mid-stream (e.g. from an mp3 that changes from
stereo->mono mid-file) don't cause us to try to add a new pad.
2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsprange.c:
rtsp: fix parsing of 'now-' ranges.
--
2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/dynamic/.gitignore:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/sprinkle.c:
* tests/examples/dynamic/sprinkle2.c:
* tests/examples/dynamic/sprinkle3.c:
examples: add some more sprinkle examples
Add some more sprinle examples and add some more comments.
See #574160.
2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/plugins/gst-plugins-base-plugins-sections.txt:
docs: add appsrc symbols to standard section
--
2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
adder: add variants for unsigned to fix warnings for unneeded check
For unsigned int out+in can't be < 0.
2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/subparse/gstsubparse.c:
subparse: use the right variable in debug log, encoding is not yet initialized
2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/v4l/v4l_calls.c:
v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/audioresample/gstaudioresample.c:
audioresample: add missing break in event handling, remove dead code
2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: do some more cleanup in _close
Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
unconnected state as it was allocated.
2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: fix the memory management of the url
Constify the url parameter in _create.
Make a copy of the url stored in the connection.
Free the url when the connection is freed.
2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
* win32/common/libgstrtsp.def:
RTSP: Add support for server tunneling
Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
that a server can store and match the id against other tunnel requests.
Fix the URI in the tunnel requests so that they contain the absolute uri and the
query string if any instead of just the hostname.
Transparently base64 decode the input stream when tunneling.
Add method to set the connection ip address so that it can be included in the
tunnel response.
Add method to connect the two tunnel requests.
Add two callbacks for the async mode to notify a tunnel start and tunnel
complete event.
Add method to reset the watch after the connection has been tunneled.
Various little refactoring to make more stuff reusable.
API: RTSP::gst_rtsp_connection_set_ip()
API: RTSP::gst_rtsp_connection_get_tunnelid()
API: RTSP::gst_rtsp_connection_do_tunnel()
API: RTSP::gst_rtsp_watch_reset()
2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
rtsp: add new defines for tunneling
Add two more result codes for tunneling support.
2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.h:
rtsp: remove , from last enum member
Remove , from last enum member to improve compatibility with other compilers.
2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
* gst/subparse/gstsubparse.c:
subparse: Convert regex code to GRegex code
Fixes: #572993. Patch author prefers to use an alias, contact
ds if you actually need a real name.
Signed-off-by: David Schleef <ds@schleef.org>
2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: remove debugging g_message
--
2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
* win32/common/libgstrtsp.def:
RTSP: add support for Quicktime tunneled RTSP
Add support for tunneling RTSP over HTTP.
Fix documentation some more.
See also #573173.
API: RTSP:gst_rtsp_connection_is_tunneled()
API: RTSP:gst_rtsp_connection_set_tunneled()
2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.h:
* gst-libs/gst/rtsp/gstrtspurl.c:
RTSP: parse rtsph uris as RTSP tunneled over HTTP
Add transport define for RTSP tunneled over HTTP.
Parse rtsph:// uris as tunneled HTTP over TCP.
API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
See also #573173.
2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstrtsp.def:
win32: Add gst_rtsp_connection_get_url definition
No, I'm not wim's buildslave, seriously.
2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtsp: add _get_url method and separate sockets
Add gst_rtsp_connection_get_url() method.
Reserve space for 2 sockets, one for reading and one for writing. Use socket
pointers to select the read and write sockets. This should allow us to implement
tunneling over HTTP soon.
API: RTSP::gst_rtsp_connection_get_url()
2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/gstapp-marshal.list:
app: force automatic rebuild of gstapp-marshal.[ch] after previous change
The previous change to appsrc/appsink requires people to 'make clean'
to get the marshallers rebuilt (causing a build failure otherwise).
Change some lines in the .list file around to force a rebuild of
these files automatically.
2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
* configure.ac:
Bump glib requirement to 2.14
2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
* ext/gio/gstgiobasesink.c:
gio: Use correct format modifier for size_t
Fixes bug #573528.
2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Use correct types for some functions on Win32
Fixes bug #573529.
2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Fix warning about using unitialized value.
2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
riff: Add more codec mappings.
This comes mostly from a review of ffmpeg/libavformat/riff.c
2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/alsa/gstalsa.c:
alsa: release pcminfo after the strdup
2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtsp/gstrtsprange.c:
rtsprange: don't leak the range in case of parsing error.
Free the gstRTSPTimeRange if we don't return it. Also simplify
gst_rtsp_range_free() as it is valid to pass NULL to g_free().
2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/alsa/gstalsa.c:
alsa: cleanup name lookup.
We can break, once we have a name to make sure, we won't read it ever twice.
2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/subparse/gstsubparse.c:
subparse: don't leak line, if flushing
2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
* ext/gio/gstgiosink.c:
giosink: reflow error handling to not leak uri
2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: remove unused code/variables
2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* win32/common/libgstapp.def:
app: add callbacks to appsrc, cleanups
Add a uri handler to appsink.
don't emit signals when we have installed callbacks on appsink.
Add callbacks to appsrc to replace the signals.
Add property to disable callbacks in appsrc, default to TRUE for backwards
compatibility but disable when callbacks are installed.
API: GstAppSrc::emit-signals
API: GstAppSrc::gst_app_src_set_emit_signals()
API: GstAppSrc::gst_app_src_get_emit_signals()
API: GstAppSrc::gst_app_src_set_callbacks()
2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/app/gstappsink.h:
* tests/check/elements/appsink.c:
Appsink: add padding for callbacks + docs
Add some padding to the callbacks structure just to be safe.
Remove the now invisible marshaller methods from the docs.
Fix a comment in the unit test.
2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstapp.def:
win32: Add new libgstapp symbol
2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/gst-plugins-base-plugins-sections.txt:
docs: clean section.txt file.
Add appsrc/sink symbols to private, as they are covered in the libs docs.
2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaybasebin.c:
docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst/playback/gstplaybin2.c:
docs: playbin2 has no stream-info
2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/video/video.h:
docs: fix newly added interlace constants and plug holes in video format docs
2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
docs: don't put random stuff in tags.
Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
tag to append text again to the documentation body.
2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/ximage/ximagesink.c:
ximagsink: do not access uninitialized height variable.
Exit like in xvimagesink, if we have partial caps.
2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
* Makefile.am:
* configure.ac:
* win32/common/config.h.in:
Change how win32/common/config.h is updated
Generate win32/common/config.h-new directly from config.h.in,
using shell variables in configure and some hard-coded information.
Change top-level makefile so that 'make win32-update' copies the
generated file to win32/common/config.h, which we keep in source
control. It's kept in source control so that the git tree is
buildable from VS.
This change is similar to the one recently applied to GStreamer,
except that it adds a few -base specific defines.
2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* win32/common/libgstapp.def:
app: add win32 .def file and only export functions we want exported
Add a .def file for win32 builds (and make check-exports).
Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
Make sure private marshaller functions aren't exported by prefixing them with __gst;
also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
a comment why we're not using glib-genmarshal for this one.
2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/dynamic/.gitignore:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/sprinkle.c:
sprinkle: Add another example app
Add an example app that dynamically adds and removes audiotestsrc elements from
adder.
2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fixed a typo.
2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst/tcp/gstmultifdsink.c:
rtsp, multifdsink: Unify the use of union gst_sockaddr.
2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
* common:
* configure.ac:
build: Update shave init statement for changes in common. Bump common.
2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
xvimageink: protect buffer_alloc from shutdown
Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
crashes when the sink is shutdown.
2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin: use flushing pads instead of fakesink
Use the flushing pads on playsink to terminate on shutdown instead of plugging
fakesinks. this should be a little cheaper.
2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
playsink: Add FLUSHING pad type
Make it possible to request a flushing pad from the playsink. We can eventually
use these flushing pads to quickly terminate the dataflow when we are shutting
down.
2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From 9cf8c9b to a6ce5c6
2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
Fixes: #565777
2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/stress-playbin.c:
stress-playbin: print the current uri
Print the current uri so that we can more easily see what uri caused a crash or
error.
2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/stress-playbin.c:
Print the errors more clearly
Print some more verbose messages when dealing with errors.
2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
Release the group lock when setting states
Release the group lock while we perform the state changes on the uridecodebins
because that might trigger callbacks that we need to handle with the group lock
taken. Avoids a possible deadly embrace in some id3/flac files.
Fixes #567396.
2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
Combine finding and creating groups
Combine the search for the current group and optionally creating one into one
function so that we can avoid taking the lock multiple times.
2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/playback/gstplaybin2.c:
Playbin2: Don't leave unused parameters in debug statements.
Fixes build on macosx
2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
Add some G_UNLIKELY because we can
Add a G_UNLIKELY when checking the shutdown variable.
2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixertrack.h:
mixer interface: Add flags to enhance mixer interfaces
This patch adds a few flags to the mixer and mixerctrl interface to
better support OSSv4 (and potentially other backends).
Patch By: Garret D'Amore <garrett.damore@sun.com>
Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
API: GST_MIXER_TRACK_WHITELIST
2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst/tcp/gstmultifdsink.c:
multifdsink: Fix strict aliasing error using a union
2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: Fix a strict aliasing warning
Fix strict aliasing warnings from casting a sockaddr_storage and
using it as a sockaddr_in6. Use a union instead.
2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
* docs/libs/.gitignore:
* docs/libs/tmpl/.gitignore:
* docs/plugins/.gitignore:
* docs/plugins/tmpl/.gitignore:
Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/plugins/Makefile.am:
* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/gstvorbisenc.h:
* ext/vorbis/gstvorbisparse.h:
* ext/vorbis/gstvorbistag.h:
* ext/vorbis/vorbis.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisenc.h:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbisparse.h:
* ext/vorbis/vorbistag.c:
* ext/vorbis/vorbistag.h:
vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Add conversion from/to YVYU colorspace
Fixes bug #572872.
2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
* gst/ffmpegcolorspace/imgconvert.c:
ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
The conversion from UYVY to RGB24 and then to GRAY8
is quite slow. Fixes bug #569655.
2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin2: fix deadlock when shutting down. Fixes #572577.
2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/icles/stress-playbin.c:
stress-playbin: make more flexible, e.g. also useful for playbin2
2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Match WSAStartup and WSACleanup correctly
Don't randomly call WSAStartup and WSACleanup but instead call the startup when
we create a connection and cleanup when we free it again. Because the internal
datastructure is refcounted, this should not cause any refcounting leaks when
the connection is managed correctly.
Fixes #562794.
2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
* pkgconfig/gstreamer-app-uninstalled.pc.in:
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
* pkgconfig/gstreamer-video-uninstalled.pc.in:
Add srcdir to includes for out-of-source builds
When you use gstreamer uninstalled and build outside
the source tree, the includes need to be specified for
both the source tree and the build tree.
Signed-off-by: David Schleef <ds@schleef.org>
2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Use shave for the build output
2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstrtsp.def:
win32: Add new symbol to libgstrtsp.def
2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspextension.c:
* gst-libs/gst/rtsp/gstrtspextension.h:
Add method for handling server requests
Add a receive_request so that extensions can react to server requests.
2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/check/libs/netbuffer.c:
Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/theora/theoraparse.c:
theoraparse: Use the correct unref functions
2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: Unref the actual buffer instead of the memory address of the buffer
2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
* common:
Automatic update of common submodule
From 5d7c9cc to 9cf8c9b
2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
* win32/common/libgstrtsp.def:
* win32/common/libgstvideo.def:
win32/common: Update .def files for recent API addition
2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
* tests/check/libs/rtp.c:
tests: Fix indentation
2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/video/video.c:
libs/video: Fix gst_video_format_new_caps* functions.
Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
don't add anything.
2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From 80c627d to 5d7c9cc
2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
Improve key/value parsing
Improve header field parsing by keeping a ref to the key/value instead of
copying it into a local variable.
2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Add trailing \0 to message length
We always put a trailing 0 at the end of the message body. Reflect this fact in
the length of the message.
2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Don't parse headers for data messages
Don't try to parse the headers on a data message because they don't have
headers.
2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
theoraenc: Add property for speed level control
Add property "speed-level" to control the amount of motion searching
the encoder does. This is only available in libtheora >= 1.0 and
will silently fail with earlier libraries. Fixes: #572275.
Signed-off-by: David Schleef <ds@schleef.org>
2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Fix 'Since' tags
2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
video: Add flags for interlaced video along with convenience methods for interlaced caps.
These three flags allow all know combinations of interlaced formats. They should
only be used when the caps contain 'interlaced=True'.
Fixes #163577 (yes, it's a 4 year old bug).
2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
Make RTSPConnection opaque and rename RTSPChannel
Make the RTSPConnection object opaque so that we can extend it in the future.
Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/riff/riff-media.c:
Add some more mappings for h264 in riff
2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtsp.def:
Add new RTSP symbols to def files
Add the new RTSP symbols to the windows def file.
2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* tests/check/Makefile.am:
* tests/check/elements/.gitignore:
* tests/check/elements/appsink.c:
Add method to install callbacks on appsink
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes #571299.
Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
performant alternative to connecting to the signals.
Add a unit test for appsink.
Clean up some of the appsink docs.
API: GstAppSink::gst_app_sink_set_callbacks()
2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add RTSP accept method
Add a method to accept a connection on a socket and create a GstRTSPConnection
for it.
API: gst_rtsp_connection_accept()
2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add RTSP channel object for async io
Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
that the connection can be monitored from a maincontext. This allows us to
operate in ASYNC mode, which is handy when building a server.
Rework the old code to use the async code under the hood.
API: gst_rtsp_channel_new()
API: gst_rtsp_channel_unref()
API: gst_rtsp_channel_attach()
API: gst_rtsp_channel_queue_message()
2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioresample/gstaudioresample.c:
audioresample: Add locking to protect the resampling context
When setting the quality/filter-length while PLAYING the
resampling context will be destroyed and created again in
some cases, which will cause crashes in the transform function
if it's called at that time.
2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/videotestsrc/videotestsrc.c:
ffmpegcolorspace/videotestsrc: Use v308 instead of V308
2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
Only conversions from/to are implemented, which
gives (indirect) support for all possible conversions.
Partially fixes bug #571147.
2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
Partially fixes bug #571147.
2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: don't abort when downstream pulls a buffer of size 0
Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
aborting. Fixes #571009 (wma file with ID3v2 tag).
2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/riff/riff-read.c:
riff: error out on nonsensical chunk sizes instead of aborting
When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
in g_malloc() or crash.
Fixes #553295, crash with fuzzed AVI file.
2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
Make git ignore backup files.
2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
* gst/playback/gstplaybin2.c:
Revert "Remove pad-removed handlers after setting the decodebins to NULL."
This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
This brought back some deadlocks. A small leak is better, for now. Need to
figure out a way to fix the leak properly.
2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
playbin2: Fix segfault on notify after group change.
If our group has been switched, then we get a selector active-pad
notification, we don't need to notify.
2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaysink.c:
playbin2: Look for volume/mute properties recursively in audio element.
Rather than only checking for volume property on the audio sink
directly, recursively look for it on sinks within it (if it's a bin).
Allows use of sink-as-volume-control where the application has supplied
an audio-sink bin that includes a real audio sink internally.
2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
* gst-plugins-base.spec.in:
Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videotestsrc/videotestsrc.c:
videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
Partially fixes bug #571147.
2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
* gst-libs/gst/rtsp/gstrtspmessage.c:
gstrtspmessage: Minor documentation correction.
Corrected documentation about what needs to be freed after calling
gst_rtsp_message_new(), gst_rtsp_message_new_request(),
gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
* ext/alsa/gstalsamixer.c:
alsamixer: Fix race condition that made alsamixer not working properly
This is due to race conditions between functions that
modified the mixer like set_volume and
snd_mixer_handle_events since the handle_events
can now be called at any time.
Fixed by adding locking around any snd_mixer call
since even read functions can modify the mixer stucture, since
alsa likes to clear it's values before reading new ones.
The favorite race condition seemed to be that set_volume
called read_elem (in alsalib) that reset the volumes to
0 and then read them with read_x_volume. This read looped
on each channel and as the race condition occured the
channels value could be anything , most of the time
it was 0. Thus no value was read or only the value of
one channel was and the volume was reset to 0.
Fixes bug #478512.
2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Bump revision to use for common submodule.
2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
xvimagesink: do not call _xwindow_clear on ready->paused.
Calling clear at that transition does things like stopping xvideo (which is not
running at that time) and also clearing anything what the application might have drawn.
This breaks handle-expose and autopaint-colorkey features.
2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtsprange.c:
* gst-libs/gst/rtsp/gstrtsprange.h:
RTSPRange: Add method to serialize ranges
Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
be used by a server.
API: GstRTSPRange::gst_rtsp_range_to_string()
2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspurl.c:
* gst-libs/gst/rtsp/gstrtspurl.h:
GstRTSPUrl: Add some const to methods
Add const to the methods that do not modify the object.
2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/playback/gstplaysink.c:
playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
The flags where present but actually not been taken into account.
2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/audioresample/gstaudioresample.c:
audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
The comment will ensure that is is marked properly in the docs and the
GParamSpecflag was causing a duplicated initialisation of the same value.
2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
Add more g_return_if_fail() calls
Check that we have a valid file descriptor before entering certain functions in
order to avoid undesirable situations.
Add some more debugging in the connect method.
2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* gst/audioresample/Makefile.am:
* gst/audioresample/gstaudioresample.c:
audioresample: Only pull in liboil if its actualy used.
Liboil still has quite significant startup overhead especialy on embedded
platforms. In audioresample it was only used for the profiling timer.
2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
* gst/typefind/gsttypefindfunctions.c:
typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
Add comments about the flac format. Tighten the check to not allow values that
refer to headers.
2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* win32/common/libgstrtsp.def:
Add new methods
Add new methods to the windows def file.
2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/pbutils/install-plugins.c:
* tests/check/libs/pbutils.c:
pbutils: remove duplicate detail strings when calling the external codec installer
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosink.h:
Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* gst/audioresample/gstaudioresample.c:
Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* sys/ximage/ximagesink.c:
Fix buffer_alloc in ximagesink
Remove some useless debug info that reported wrong image sizes.
When upstream does not accept out suggested size, fall back to allocating an
image of the requested width/height instead of the currently configured size.
The problem is that an image is reused from the pool because the width/height
match but the caps on the new buffer are the requested caps with possibly
different height/width resulting in errors.
2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
* gst/playback/gsturidecodebin.c:
Fix documentation for autoplug-select
fix the documentation strings for the autoplug-select signal.
Fixes #570142.
2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspmessage.c:
Fix string leak in rtspmessage
when we remove a header field from a message we must free the value associated
with the key to avoid a memory leak.
2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
Its "Base Library" and not just "Library".
2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstaudiofilter.c:
Link to the class, as we can't link to the members yet.
2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Remove pad-removed handlers after setting the decodebins to NULL.
They do needed cleanup; without this we leak selector requestpads.
2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Unref selector request pad even if we no longer have a selector.
During destruction, we won't have a selector any more, but we still need
to unref the pad to avoid leaking it.
2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Unref source in playbin2's finalize method
2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaysink.c:
Fix more leaks of pads and elements in gstplaysink.
Don't keep extra references to volume and mute elements; we don't need
to do so.
Ensure we unref pads that we have references to, and release request
pads.
2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaysink.c:
Avoid leaking all playsinks. Fix some internal leaks.
Playsink was holding references to itself. Don't do that, it's not cool.
Also, free all chains in dispose.
2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Unref peer request pad after releasing it, since we hold a reference.
2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Fix caps leak in playbin2.
2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Unref active pad from selector when finding active stream.
2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c:
Free uris when finalizing playbin2 instance.
2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gsturidecodebin.c:
Unref pads when iterating over them in analyse_source.
Fixes leak of source's srcpad when using uridecodebin.
2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
Add releaseinfo with online url.
2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
* gst/playback/gstplaybasebin.c:
Fix compilation warning on Forte
2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
* gst/adder/gstadder.c:
Don't do void pointer arithmetic.
2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
* common:
Bump common
2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
* autogen.sh:
* common:
Use a symbolic link for the pre-commit client-side hook
2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
* .gitignore:
Add more files/directories to ignore
2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.c:
fix some typos
Fix some typos in the doc string of the new
gst_rtsp_options_as_string() method.
2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspmessage.c:
* gst-libs/gst/rtsp/gstrtspmessage.h:
Add new RTSP message method to set header
Add gst_rtsp_message_take_header() that takes ownership of the passed header
value. This allows us to avoid an allocations and memory copy in some
situations.
API: GstRTSPMessage::gst_rtsp_message_take_header()
2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
Add new method to docs
Add the new gst_rtsp_options_as_text() method to the docs.
2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
Add method to serialize RTSP options
Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
string.
API: GstRTSP::gst_rtsp_options_as_text()
2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
* gst/typefind/gsttypefindfunctions.c:
Ensure we have sufficient data when using data scan contexts.
Fixes crashes typefinding things that look like they might contain AAC
data (but probably aren't actually AAC).
2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
* ext/gio/Makefile.am:
Fix include order for gio plugin
2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
* win32/common/config.h:
Update win32 config.h for 0.10.22.1 dev cycle
2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
* .gitignore:
* docs/libs/.gitignore:
* gst-libs/gst/audio/.gitignore:
* gst-libs/gst/video/.gitignore:
* po/.gitignore:
* tests/examples/dynamic/.gitignore:
Extend and clean up git ignores
2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst/audioresample/Makefile.am:
* gst/audioresample/README:
* gst/audioresample/arch.h:
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/fixed_arm4.h:
* gst/audioresample/fixed_arm5e.h:
* gst/audioresample/fixed_bfin.h:
* gst/audioresample/fixed_debug.h:
* gst/audioresample/fixed_generic.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
* gst/audioresample/resample_sse.h:
* gst/audioresample/speex_resampler.h:
* gst/audioresample/speex_resampler_double.c:
* gst/audioresample/speex_resampler_float.c:
* gst/audioresample/speex_resampler_int.c:
* gst/audioresample/speex_resampler_wrapper.h:
* gst/speexresample/Makefile.am:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_arm4.h:
* gst/speexresample/fixed_arm5e.h:
* gst/speexresample/fixed_bfin.h:
* gst/speexresample/fixed_debug.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/resample_sse.h:
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
* gst/typefind/gsttypefindfunctions.c:
* tests/check/Makefile.am:
* tests/check/elements/audioresample.c:
* tests/check/elements/speexresample.c:
Rename files and types from speexresample to audioresample
Rename files and types from speexresample to audioresample
to finish the move and to prevent any confusion.
2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* sys/xvimage/xvimagesink.c:
Add some more debugging to the Xv strides
Add some more debugging to the strides as they are received from the server and
the expected strides.
2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
Add typefind function for gsm
Because core now supports typefindfactories without a typefind function we can
register a factory fo GSM that will --if all else fails-- assume the file is a
GSM file based on the registered extension.
Fixes #566661.
2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
Use more performant link function
We can use gst_element_link_pads() instead of the more generic
gst_element_link() function because we know the pads. This saves some cycles
because the more generic function needs to search for possible compatible caps
etc.
2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
Add more codec ids for RIFF formats
Handle codec ID for various other AAC formats.
Sync the list of possible codec ids with that of ffmpeg.
Fixes #567255
2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/theora/theoradec.c:
Use rounded values for image strides and sizes
Round up the height before calculating the expected size and
strides of the output image.
2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/alsa/gstalsasink.c:
Improve debug message
Improve the debug message when alsa returns an error.
2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/app/gstappsrc.c:
Reset queued_bytes counter when flushing
Set the amount of queued bytes in the internal queue back to 0 when we clear the
queue.
Fixes #567982
2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
* gst/typefind/gsttypefindfunctions.c:
Add typefinder for Mobile XMF. Fixes bug #568707.
2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
* configure.ac:
Fix linking on Solaris. Fixes bug #568482.
Check for nsl and socket libraries and add them to
LIBS if they're found. They're needed for socket()
and gethostbyname() on Solaris.
2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
* gst/playback/gstplaybasebin.c:
Fix use-after-unref problem noticed by Josep Torra Valles, and run gst-indent
2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Update common snapshot.
2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
* common:
Fix pre-commit hook
2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts32.c:
Reduce the number of allocations for creating FFT contexts
Reduce the number of allocations from 2 to 1 for every FFT
context by allocating enough memory for the FFT context
and passing parts of it to the kissfft allocation functions.
2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
Back to devel -> 0.10.22.1
2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
* autogen.sh:
* common:
Install and use pre-commit indentation hook from common
2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Avoid overflows in the padding checks by doing the check slightly differently. Add a unit test to check for correct behaviour.
2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
* autogen.sh:
autogen.sh : Use git submodule
=== release 0.10.22 ===
2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* po/LINGUAS:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/config.h:
Release 0.10.22
Original commit message from CVS:
Release 0.10.22
2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
Original commit message from CVS:
Update .po files
2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
Original commit message from CVS:
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
Use correct struct alignment everywhere to prevent unaligned
memory accesses, resulting in SIGBUS on sparc and probably others.
Fixes bug #500833.
2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Forward unknown events upstream to allow latency configuration.
Fixes bug #567960.
2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (groups_set_locked_state):
Provide the right arguments to a debug line.
2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Don't reset the colorkey when element is reused. Fixes #567511.
2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: 0.10.21.3 pre-release
Original commit message from CVS:
* configure.ac:
0.10.21.3 pre-release
2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
Store the returned signal id in the right slot when
registering the pull-buffer signal.
Fixes #567168
Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c:
Small docs addition to clarify that one really mustn't free
the constant GList returned (#566812).
2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
Add GType for GstRTSPUrl and expose a copy function because we can.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
(gst_rtsp_url_get_type), (gst_rtsp_url_copy):
* gst-libs/gst/rtsp/gstrtspurl.h:
* win32/common/libgstrtsp.def:
Add GType for GstRTSPUrl and expose a copy function because we can.
API: gst_rtsp_url_copy()
Fixes #567027.
2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add plugin dependency for the GIO and GVfs modules.
Original commit message from CVS:
* configure.ac:
* ext/gio/gstgio.c: (plugin_init):
Add plugin dependency for the GIO and GVfs modules.
Fixes bug #566876.
2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add plugin dependency for the gnomevfs modules.
Original commit message from CVS:
* configure.ac:
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
Add plugin dependency for the gnomevfs modules.
Fixes bug #566875.
2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
Original commit message from CVS:
* win32/common/libgstcdda.def:
Add new symbol to the list of exported symbols.
2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Fix some comments and docs.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_uri), (gst_play_bin_set_suburi),
(no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
(activate_group), (deactivate_group), (groups_set_locked_state),
(gst_play_bin_change_state):
Fix some comments and docs.
Post an error message when we fail to link the selector to the sink.
Remove pushing of EOS, this seems unneeded.
Lock the state of deactivated groups so that they don't accidentally
reactivate when the playbin2 state changes.
Reuse uridecodebins.
Unlock and relock state of groups when playbin goes to NULL.
Fixes #566654.
Fixes #566341.
* gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
Only do something in the pad removed callback when we are dealing with
our sourcepads because the sinkpads don't have a ghostpad.
2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Make the GType of GstCDDABaseSrcMode public for bindings.
Fixes bug #566837.
2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (plugin_init):
Use new core API to make registry re-scan the plugin
whenever visualisations are added or removed (see #350477).
2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
* gst-libs/gst/audio/gstaudioclock.h:
Make gst_audio_clock_new use const gchar* to ease the wrapping of
C++ bindings. Fixes #566723.
2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add pkg-config files for libgstapp. Fixes bug #566761.
Original commit message from CVS:
* configure.ac:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-app-uninstalled.pc.in:
* pkgconfig/gstreamer-app.pc.in:
Add pkg-config files for libgstapp. Fixes bug #566761.
2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Make debug categories static. Use _element_class_set_details_simple().
2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
(gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
(gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
(gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer)::
* gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
* gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_finalize), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_is_seekable), (gst_app_src_check_get_range),
(gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
(gst_app_src_set_caps), (gst_app_src_get_caps),
(gst_app_src_set_size), (gst_app_src_get_size),
(gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
(gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full),
(gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
* gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
Move private data into a private instance struct. Add padding to
instance and class structures exposed in public headers. Add
Since markers to the gtk-doc blurbs (#566750).
2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/app/appsrc_ex.c: Some comments.
Original commit message from CVS:
* tests/examples/app/appsrc_ex.c: (main):
Some comments.
When pulling a buffer we can get NULL when the element is EOS, don't try
to unref this NULL buffer.
2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Fix up build flags and include statement for the new generated
enumtypes files, to fix dist.
2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-app.xml:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* tests/examples/Makefile.am:
* tests/examples/app/Makefile.am:
Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
Disconnect signal handlers before destroying a previous decodebin so
that we don't end up causing deadlocks. Fixes #566586.
2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_check_get_range),
(gst_audio_test_src_set_property),
(gst_audio_test_src_get_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add property to control pull/push based scheduling.
2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
Make the seek and colorkey examples depend on gtk+-x11 as they use
Original commit message from CVS:
* configure.ac:
* tests/examples/seek/Makefile.am:
* tests/icles/Makefile.am:
Make the seek and colorkey examples depend on gtk+-x11 as they use
GDK_WINDOW_XID.
Fixes the build with gtk+-quartz.
2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
win32/common/: Add new exports to win32 files.
Original commit message from CVS:
* win32/common/libgstaudio.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
Add new exports to win32 files.
2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
* gst-libs/gst/tag/gsttagdemux.h:
Add GType for GstTagDemuxResult enum.
2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
This will help bindings to use it.
2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c:
* win32/MANIFEST:
* win32/common/audio-enumtypes.c:
(gst_audio_channel_position_get_type),
(gst_ring_buffer_state_get_type),
(gst_ring_buffer_seg_state_get_type),
(gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
* win32/common/audio-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
* win32/common/multichannel-enumtypes.h:
* win32/vs6/grammar.dsp:
* win32/vs6/libgstaudio.dsp:
* win32/vs7/libgstaudio.vcproj:
* win32/vs8/libgstaudio.vcproj:
Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
audio- in order to wrap all enums declarations of that library.
This modification should not matter since that header file is not a
public header (it will be included by public headers).
Modify win32 crap^Wfiles accordingly.
2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Complete Sebastien's commit from the 13th by exporting the
_slave_method_get_type() methods.
2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_query),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full):
* gst-libs/gst/app/gstappsrc.h:
Add properties and methods to configure and retrieve the min and max
latencies.
2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/: Implement URI query. Fixes bug #562949.
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
(gst_gio_base_src_query):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
(gst_gnome_vfs_src_query):
Implement URI query. Fixes bug #562949.
2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Add some debug info.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Add some debug info.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_reconfigure), (gst_play_sink_request_pad),
(gst_play_sink_release_pad):
Add some more debug info.
Reconfigure the audio chain when we switch between raw and encoded audio
in gapless playback.
2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Pause the write thread before deactivating and releasing the ringbuffer
to avoid a deadlock when we do gapless playback with different sample
rates in playbin2. Fixes #564929.
2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
Make GstAudioSrcSlaveMethod get_type() function non-static
as it's public now.
* win32/common/libgstaudio.def:
* win32/common/libgstnetbuffer.def:
Add some missing functions to the list of exported symbols.
2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
Original commit message from CVS:
Patch by: Andrew Feren <acferen at yahoo dot com>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
(gst_netaddress_get_address_bytes),
(gst_netaddress_set_address_bytes):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Make gst_netaddress_get_ip4_address fail for v6 addresses.
Make gst_netaddress_get_ip6_address either fail or return the v4
address as a transitional v6 address.
Add two convenience functions:
API: gst_netaddress_get_address_bytes()
API: gst_netaddress_set_address_bytes()
Fixes #564896.
2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
Add appsrc and appsink documentation.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
Add appsrc and appsink documentation.
2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
Original commit message from CVS:
* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
Cleanup variable names to make the adder-loop easier to understand.
Also try to use liboil to spee it up, but ifdef it out as it does not
make any change for me (Intel pentim M (sse,sse2) please try on other
systems).
2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
Add minimal docs to make the remaining tcp elements show up.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversrc.c:
Add minimal docs to make the remaining tcp elements show up.
Fixes #564139.
2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/: Fix example to unref after emiting the push-buffer action.
Original commit message from CVS:
* examples/app/appsrc-ra.c: (feed_data):
* examples/app/appsrc-seekable.c: (feed_data):
* examples/app/appsrc-stream.c: (read_data):
* examples/app/appsrc-stream2.c: (feed_data):
Fix example to unref after emiting the push-buffer action.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
(gst_app_src_push_buffer_action):
Don't take the ref on the buffer in push-buffer action because it's too
awkward for bindings. Fixes #564482.
2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
win32/common/config.h: Update to CVS version.
Original commit message from CVS:
* win32/common/config.h:
Update to CVS version.
* win32/common/config.h.in:
Hardcode path to plugin install helper exe, just like we hardcode
the paths in core. Removes another source of VCS conflicts for
people hacking gst-plugins-base on systems with autotools.
2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
Original commit message from CVS:
* m4/Makefile.am:
And a couple more .m4 that don't exist anymore with gettext 0.17
2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
Original commit message from CVS:
* m4/Makefile.am:
inttypes.m4 hasn't been available since gettext-0.15, and since we now
require gettext >= 0.17 ... we can remove it from the list of files to
dist.
2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/: XRef to GstXOverlay.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
XRef to GstXOverlay.
2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
Free the factory array when finalizing.
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
Use a GstStaticPadTemplate since the src pad caps are fixed.
2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
(gst_vorbis_enc_init):
Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
pad templates.
2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add mapping for VP6 in avi/riff.
2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
Original commit message from CVS:
* gst/subparse/samiparse.c: (sami_context_push_state),
(sami_context_pop_state), (start_sami_element), (end_sami_element):
Some versions of libxml seem to be very picky as to strict formatting
of the input and never 'close' the final </body> tag.
In order to fix that bad behaviour, we trigger the flushing of
remaining data on both </body> and </sami>.
Fixes #557365
2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
Original commit message from CVS:
Patch by: Guillaume Emont <guillaume at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinders for MS Word files and OS X .DS_Store files to
prevent them to be recognized as MPEG files. Fixes bug #564098.
2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Add some more debug info.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain),
(gst_play_sink_reconfigure):
Add some more debug info.
Fix linking of just an encoded sink.
Handle failure to create a sink chain more gracefully than crashing.
2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
Pushing 10 buffers is enough to run the test.
2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Hook up the SKIP seek flag.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek), (stop_cb),
(skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
(main):
Hook up the SKIP seek flag.
2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb):
Error out with a missing-plugin error when the input-selector was not
found.
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Indentation.
2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_send_event), (gst_play_sink_change_state):
Use G_DEFINE_TYPE.
Try to set the selected sink to READY before using it. This will allow
for detection of incompatible formats sooner.
Don't cause a fatal error when conversion elements are missing but post
a missing-element message and a warning instead because things might
still link and run fine.
Simplyfy the construction of audio and video sink chains.
2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
(gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
init from glib.
2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
gst/: Include glib.h instead of a specific GLib header. Including single
Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.
2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
Original commit message from CVS:
2008-12-09 Julien Moutte <julien@fluendo.com>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Fix handling of odd chunks in riff metadata.
2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_before_transform), (volume_transform_ip):
Use new basetransform vmethod to reconfigure the dynamic properties and
any pending volume/mute changes. Fixes #563508.
2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
Original commit message from CVS:
* configure.ac:
First check for "theoraenc theoradec" and if that failed check
for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
deprecate the latter. Also linking on Windows fails with just "theora"
and the version check would fail for the release candidates.
Fixes bug #563718.
2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
Add basic docs to decodebin and link to decodebin from decodebin2.
2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
Original commit message from CVS:
* configure.ac:
Apparently AC_CONFIG_MACRO_DIR breaks when using more
than one macro directory, reverting last change.
2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
Original commit message from CVS:
* configure.ac:
Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
our M4 macros.
2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
sys/: Clear all flags on buffers returned from the image pool.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
Clear all flags on buffers returned from the image pool.
Fixes #563143
2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).
2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
Original commit message from CVS:
Patch by: Cygwin Ports maintainer
<yselkowitz at users dot sourceforge dot net>
* autogen.sh:
* configure.ac:
Require gettext 0.17 because older versions don't mix with libtool
2.2. At build time an older gettext version will still work.
Fixes bug #556091.
2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
* ChangeLog:
* gst/speexresample/Makefile.am:
fix build
Original commit message from CVS:
fix build
2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Update documentation of speexresample for the new element name.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-videorate.xml:
* gst/speexresample/gstspeexresample.c:
Update documentation of speexresample for the new element name.
2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
Original commit message from CVS:
* gst/speexresample/README:
Update README with the latest diff between the Speex resampler
and our copy.
2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (plugin_init):
Update the debug category from speex_resample to audioresample.
2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Remove audioresample files.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
* tests/check/elements/audioresample.c:
Remove audioresample files.
2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
Original commit message from CVS:
* docs/plugins/inspect/plugin-audioresample.xml:
Regenerated for library filename change.
2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst/speexresample/gstspeexresample.c: (plugin_init):
* gst/speexresample/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(GST_START_TEST), (test_pipeline):
Rename the moved speexresample to audioresample, integrate into the
build system and remove the old audioresample from the build system.
Fixes bug #558124, #385061, #346218, #116051.
2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!
2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
A successful gst_poll_wait() doesn't always mean successful connect() on
Windows. We should check errors by calling gst_poll_fd_has_error().
See #561924.
2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts with valgrind.
2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
Fix typo in the docs.
2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
If no stream was found before receiving EOS, post an error message.
Fixes #561924.
2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/: Parse segment events.
Original commit message from CVS:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: (gst_theora_enc_init),
(theora_buffer_from_packet), (theora_push_packet),
(theora_enc_sink_event), (theora_enc_is_discontinuous),
(theora_enc_chain):
Parse segment events.
Pass incomming buffer timestamps to outgoing buffers.
Use the running_time to construct the granulepos.
Fixes #562163.
2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Fix buffer-duration property.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Fix buffer-duration property.
2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Add notification of current stream. Add ability to configure buffer
sizes.
* gst/playback/gsturidecodebin.c:
Add ability to configure buffer sizes for streaming mode.
Bug #561734.
2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
If we're muxing a dirac stream, flush the page after every picture.
2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
Original commit message from CVS:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
Use gst_buffer_try_new_and_alloc() and fail properly if the
allocation failed. This prevents abort() if downstream elements
request an insane amount of memory.
2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
gst/volume/gstvolume.*: Cleanup volume, define and use default values.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_volume), (gst_volume_set_volume),
(gst_volume_get_volume), (gst_volume_set_mute),
(gst_volume_class_init), (gst_volume_init),
(volume_process_double), (volume_process_float),
(volume_process_int32), (volume_process_int32_clamp),
(volume_process_int24), (volume_process_int24_clamp),
(volume_process_int16), (volume_process_int16_clamp),
(volume_process_int8), (volume_process_int8_clamp), (volume_setup),
(volume_transform_ip), (volume_set_property),
(volume_get_property):
* gst/volume/gstvolume.h:
Cleanup volume, define and use default values.
Recalculate new volume and mute setup before processing. Fixes #561789.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add controller unit test. Patch by: Jonathan Matthew
Fix bogus test that messed with basetransform's internal state.
2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (GST_START_TEST):
Make the unit test a bit faster to prevent timeouts, especially
with valgrind.
2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
Original commit message from CVS:
* gst/videorate/gstvideorate.c:
Add jpeg and png image media types to the caps. Fixes #561436.
2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain):
Don't post an error when we can't configure the volume but post a
warning instead. Fixes #561780.
2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
Original commit message from CVS:
Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a zone plate pattern generator based on BBC R&D Report
1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
kx2=20 ky2=20 kt=1'.
2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
(gst_speex_resample_get_property):
Add a "filter-length" property that maps to the quality values
for compatibilty with audioresample.
2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix random fat-fingering making this not compile.
2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
If the top-level type of the stream is plain text, don't try to decode
it, matching behaviour of decodebin.
* gst/playback/gstplaysink.c:
If we fail to generate a text chain (e.g. due to missing optional
plugins), don't crash.
2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c:
Fix win32 build. Oops.
2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c:
Use WSAGetLastError() rather than errno/h_errno on win32.
2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Support WMA Lossless properly.
2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add "colorspec" property, specifying whether to generate BT.601
or BT.709 video. This only affects YCbCr values, not RGB, since
if you're generating a 709 test pattern, presumably you want
709 RGB primaries, not 601. Also add "smpte75" pattern, which
uses 75% colors instead of 100%, since this is often more useful
for testing (and also follows the SMPTE EG-1 guideline).
2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
Add a "sink-caps" property to decodebin like it's done for decodebin2.
Fixes #560380.
2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Guard against a NULL dereference I somehow encountered -
with a FLUSH_STOP arriving either before basetransform _start(),
or after _stop().
* gst/typefind/gsttypefindfunctions.c:
Make sure we never jump backwards when typefinding corrupt mov files.
2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
Fix random type causing a docs warning.
2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
Original commit message from CVS:
* sys/v4l/gstv4l.c:
Give it a minimal rank for autovideosrc.
2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
(plugin_init):
Improve typefinding of ISO JPEG2000 mime types.
2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
* sys/xvimage/xvimagesink.h:
Avoid typechecking when we do trivial casts.
Move error handling out of the main program flow.
Sneak in the display-region caps property, not completely correct yet.
Cache the width/height in buffer_alloc instead of parsing it from the
caps all the time.
2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (deactivate_group):
don't try to unlink the selector sinkpad when we don't have it yet. This
can happen if an error occured before the group was complete.
2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
(gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
(gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
(gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
(gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
(gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
(gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
(gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
(gst_rtp_buffer_get_payload_type),
(gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
(gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
(gst_rtp_buffer_set_timestamp),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
Avoid expensive type checks we already did as part of the
_validate() function that should be called first.
2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp):
Fix some cases where a newsegment event was not sent.
2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Catch state change errors and stop from the uridecodebin elements
instead of trying to continue in vain.
2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
gst/: Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_callback):
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
for the latency to expire, fixes #559567.
2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
Original commit message from CVS:
* gst/adder/gstadder.c:
Change author string after seeing output of gst-inspector.
2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Don't try to do crazy things when we only have a text pad without a
video pad. Fixes #559478.
2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsrc.*: Add is-live property.
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_push_buffer):
* gst-libs/gst/app/gstappsrc.h:
Add is-live property.
Add some more docs.
2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-media.c: Fix case where we don't have a range for the rates or channels as is the case with tr...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Fix case where we don't have a range for the rates or channels as is the
case with truespeech.
2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_init), (volume_setup),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume), (volume_get_property):
* gst/volume/gstvolume.h:
Keep negotiated state in a separate variable.
Protect the volume and mute properties with the object lock.
Protect modifying the transform with the transform lock.
2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
Only convert caps to string when debug is enabled.
2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/: Copy seqnum.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_init),
(gst_theora_dec_reset), (theora_dec_src_event),
(theora_dec_sink_event), (theora_handle_type_packet):
Copy seqnum.
Keep events in a pending list, like vorbisdec, instead of trying
to construct a segment event ourselves.
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
(vorbis_dec_src_event), (vorbis_dec_sink_event):
* ext/vorbis/vorbisdec.h:
Copy seqnum.
2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
(gst_ogg_demux_loop):
* ext/ogg/gstoggdemux.h:
Copy seqnums around to track playback segments and messages.
2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Don't install static libs for plugins. Fixes #550851 for -bad.
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
Original commit message from CVS:
Based on patch by: Matthias Kretz <kretz at kde dot org>
* ext/alsa/gstalsasink.c: (gst_alsasink_open),
(gst_alsasink_prepare), (gst_alsasink_unprepare),
(gst_alsasink_write):
Make all access non-blocking so that we can better handle unplugging
of usb devices. Fixes #559111
2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
Original commit message from CVS:
Patch by: Damien Lespiau <damien.lespiau gmail com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_write):
Make the next call to poll not depend on previous calls to poll with or
without reading from the active descriptor. Fixes #544293.
2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
Add TODO at the top of the file for enabling SSE/ARM specific
optimizations and choosing the fastest implementation at runtime.
Add g_assert_not_reached() at two places that should really never
be reached.
2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Fix format string and arguments.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_check_discont):
Fix format string and arguments.
* gst/speexresample/resample_sse.h:
Add missing file.
2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/: Add missing headers to Makefile.am.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
(_benchmark_int_int), (_benchmark_integer_resampling),
(plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add missing headers to Makefile.am.
Update copyright, years and my mail address.
Benchmark the integer resampling implementation against the
float implementation and use the faster one for 8/16 bit integer
input. On most recent systems the floating point version is faster.
2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
Original commit message from CVS:
Patch by: Nick Haddad <nick at haddads dot net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for other fourcc codes that are commonly used for
'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
Fixes #558553.
2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
The length for the buffer conversion function is the number of
audio frames, i.e. we need to multiply it by the number of channels
to get the number of values. Also spotted by the unit test after
running in valgrind.
2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_funcs),
(gst_speex_resample_transform_size),
(gst_speex_resample_convert_buffer),
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add support for int8, int24 and int32 input by converting internally
to/from int16 or double.
2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/arch.h:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(_gcd), (gst_speex_resample_transform_size),
(gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_wrapper.h:
* tests/check/elements/speexresample.c: (setup_speexresample),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance):
Add support for double samples as input and refactor the usage
of the different compilation flavors of the speex resampler.
2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Return the result of parent_class->event().
2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.c: Fix the docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Fix the docs.
2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_get_unit_size),
(gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
(gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
instead of GST_DEBUG, ...
2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
(gst_speex_resample_process):
Fixate to the nearest supported rate instead of the first one.
2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (audioresample_fixate_caps):
Fixate the rate to the nearest supported rate instead of
the first one. Fixes bug #549510.
2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_arm4.h:
* gst/speexresample/fixed_arm5e.h:
* gst/speexresample/fixed_bfin.h:
* gst/speexresample/fixed_debug.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/resample.c: (compute_func), (main), (sinc),
(cubic_coef), (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init_frac), (speex_resampler_process_native),
(speex_resampler_magic), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem):
* gst/speexresample/speex_resampler.h:
Update Speex resampler with latest version from Speex GIT.
2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
win32/common/libgstaudio.def: Add new symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add new symbols.
2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Attempt to make obfuscated code clearer.
2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Move float endianness conversion macros to core. Second part of bug ##555196.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/floatcast/floatcast.h:
Move float endianness conversion macros to core. Second part of
bug ##555196.
2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/: Don't mark as gtk-doc docs as they aren't public.
Original commit message from CVS:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
Don't mark as gtk-doc docs as they aren't public.
2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
* tests/icles/Makefile.am:
* tests/icles/test-colorkey.c:
Allow setting colorkey if possible. Implement property probe interface
for optional X features (autopaint-colorkey, double-buffer and
colorkey). Fixes #554533
2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Remove useless buffer size assignment. It already has this value.
2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Implement a separate activate functions to start monitoring the segments
or, in pull mode, pulling in data.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
(gst_base_audio_sink_activate_pull),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Implement pad and element convert query function.
Activate the ringbuffer.
Use the segment last_stop value as the offset to pull.
Use new basesink _do_preroll() method to preroll in the pulling thread.
Take appropriate locking in the pulling thread.
* gst-libs/gst/audio/gstringbuffer.h:
Update some docs.
2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
Improve MXF typefinding a bit by searching for a header partition
pack instead of just a general partition pack and checking more
bytes for valid values.
2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/icles/.cvsignore: update ignore file.
Original commit message from CVS:
* tests/icles/.cvsignore:
update ignore file.
* tests/icles/Makefile.am:
* tests/icles/test-box.c: (make_pipeline), (main):
Add another interactive command line experimentation suite for
dynamically boxing/cropping/saling an input video.
2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
Add methods to more accuratly control the pulling thread of a ringbuffer.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
(gst_ring_buffer_activate), (gst_ring_buffer_is_active):
* gst-libs/gst/audio/gstringbuffer.h:
Add methods to more accuratly control the pulling thread of a
ringbuffer.
Add format conversion helper code to the ringbuffer.
API: GstRingBuffer:gst_ring_buffer_activate()
API: GstRingBuffer:gst_ring_buffer_is_active()
API: GstRingBuffer:gst_ring_buffer_convert()
2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Signal thread startup earlier so that we can immediatly go into pull
mode when we have to and block on preroll.
2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read):
In pull mode we want the callback to prepull a buffer we can preroll on
even when we are not yet playing.
2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Don't install static libs for plugins. Fixes #550851 for base.
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gio/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/gdp/Makefile.am:
* gst/playback/Makefile.am:
* gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* sys/v4l/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
Don't install static libs for plugins. Fixes #550851 for base.
2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
Set the default blocksize to -1 because we will then use the configured
samplesperbuffer to create our output buffer.
2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add mappping for the KMVC (Karl Morton's Video) Codec.
2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Don't forget to advance the offset of what we're matching against, else
we end up in a forever loop.
2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Improve typefinding a bit. If we don't have a Unicode charset
try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
ext/theora/theoradec.c: Fix build on macosx.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_decode_buffer):
Fix build on macosx.
2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
Original commit message from CVS:
Based on patch by: Robin Stocker <robin at nibor dot org>
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_init),
(theora_dec_setcaps), (theora_handle_type_packet),
(theora_dec_decode_buffer), (theora_dec_change_state):
Parse input caps and make the PAR override the encoded PAR when
specified by a container. Fixes #555699.
2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add some more G_LIKELY
Fail when the setcaps function was not called.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
Propagate return value of setcaps.
2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
Original commit message from CVS:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_convert_to_utf8), (detect_encoding), (convert_encoding),
(get_next_line), (gst_sub_parse_data_format_autodetect),
(feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for UTF16/UTF32 subtitles as long as the first bytes of
the first buffer contain the BOM. This also adds support for other
encodings that allow NUL bytes via the encoding property.
Fixes bugs #552237 and #456788.
2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't drop the last byte of image tags if they're not an URI list.
Fixes bug #556066.
2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
For looking at the 4th byte we have to get 4 bytes of course
and not 3.
2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Improve FLAC-without-headers typefinding by looking at most of the
frame header and checking if invalid values are used. Should prevent
quite some false positives compared to the old version which only
check if the first 14 bits are set.
2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Don't assert on caps==NULL.
2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_start), (gst_audio_test_src_stop),
(gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Define the default property values in the usual place.
Implement start/stop to reset values correctly.
Calculate the sample size only once when we negotiate.
Rename some values to make more sense.
Keep track of our byte range.
Add support for pull based scheduling. Disabled for now until we have
the whole stack working.
Set the BUFFER_OFFSET correctly.
2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c:
(gst_vorbis_enc_buffer_check_discontinuous):
Fix discontinuity detection which was broken by last commit.
2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
Original commit message from CVS:
* configure.ac::
Require core CVS for ghostpad API additions used by decodebin2.
2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix debug statements (space between '%' and actual format).
2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
Remove bogus assert, the decodepad could have been created inside an
already existing group.
2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
changelog
Original commit message from CVS:
changelog
2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
Original commit message from CVS:
2008-10-08 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
target instead of setting it.
(gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
API for a decode pad. The bugfix is that we set the group in
activate(), not when the pad was created because it might be NULL
then.
(gst_decode_group_control_source_pad, gst_decode_group_expose):
Update to use the API.
2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
Original commit message from CVS:
2008-10-08 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
be a subclass of GstGhostPad.
(analyze_new_pad): So, when emitting the signals that determine
how we do autoplugging, already create the ghost pad and use it as
the pad in the signal arguments. This allows applications to make
a connection between the pad passed in e.g. autoplug-continue, and
the pad passed in new-decoded-pad.
(connect_pad, expose_pad): Update to receive the ghosted decode
pad in the args, retargetting it as necessary if we have to plug
the target pad through a multiqueue.
(gst_decode_group_control_source_pad): Adapt to receive an
already-ghosted pad that just needs activation, blocking, and
drain notification.
(sort_end_pads): Adapt for decode pads actually being pads.
(gst_decode_group_expose): Adapt for decode pads actually being
pads. Rewrite the decode pad names so they appear in order. Adds a
new error case if we couldn't set the name.
(gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
logic.
(gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
New API for the decode pad, needed because we shouldn't do these
things inside gst_decode_pad_new(), but after.
(gst_decode_pad_new): Change to actually make the real pad, and
delay the blocking/drainage bits.
2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
Original commit message from CVS:
Patch by: Daniel Drake <dsd at laptop dot org>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
Unref all buffers when clearing collectpads. Fixes bug #546955.
2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
Original commit message from CVS:
Based on a patch by: Klaas <klaas at rivercrew dot net>
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
(gst_vorbis_enc_buffer_check_discontinuous),
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Keep track of the upstream segments and use the running time on that
segment instead of the buffer timestamp everywhere. Fixes bug #525807.
2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_convert):
Prevent overflows with big buffer when calculating the size of
the intermediate buffer by using gst_util_uint64_scale() instead of
plain arithmetics. Fixes bug #552801.
2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
Original commit message from CVS:
Patch by: Pavel Zeldin <pzeldin at gmail dot com>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
(gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
(gst_clock_overlay_init), (gst_clock_overlay_set_property),
(gst_clock_overlay_get_property):
* ext/pango/gstclockoverlay.h:
API: Add ability to specify format for date/time display by
adding a "time-format" property.
Fixes bug #554879.
2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
Original commit message from CVS:
Patch by: Jan Gerber <j at oil21 dot org>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add FFV1 fourcc to support playback of FFMPEG lossless video
in AVI. Fixes bug #555319.
2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Implement skew clock slaving. Fixes #552559.
2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/: Fix include of config.h
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
* gst-libs/gst/audio/testchannels.c:
Fix include of config.h
2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
Original commit message from CVS:
Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
(print_media), (gst_sdp_message_dump):
Fix parsing of the c= field containing multicast addresses.
Fixes #552199.
Add the connection info to the session or streams.
Fix parsing of the bandwidth.
Add debugging for the connections and bandwidths for a media.
Add debugging for the bandwidth of the session.
2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_change_state):
Configure the next seqnum and timestamp in the state change so that they
can be queried soon after.
2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Improve debugging of the rtptime.
2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to development -> 0.10.21.1
Original commit message from CVS:
* configure.ac:
Back to development -> 0.10.21.1
2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
* ChangeLog:
ChangeLog surgery
Original commit message from CVS:
ChangeLog surgery
2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
(plugin_init):
Add typefinder for MXF.
2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
(plugin_init):
Add typefinder for MXF.
2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
Original commit message from CVS:
* tests/icles/Makefile.am:
Only build test-colorkey if GTK+ is available.
=== release 0.10.21 ===
2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/config.h:
Release 0.10.21
Original commit message from CVS:
Release 0.10.21
2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
Original commit message from CVS:
Update .po files
2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: 0.10.20.4 pre-release
Original commit message from CVS:
* configure.ac:
0.10.20.4 pre-release
2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
Original commit message from CVS:
Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
* ext/theora/theoraparse.c: (theora_parse_set_streamheader):
Set the BOS flag on the BOS packet. Fixes #553244.
2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspmessage.c:
(gst_rtsp_message_parse_request),
(gst_rtsp_message_parse_response):
Fix the g_return_val_if_fail() statements.
2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Fail to activate if there's insufficient data in the file to be usable,
preventing an assertion fail later. Fixes #552960
2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Commit stuff that should have gone in last week when I made the pre-releases:
Original commit message from CVS:
Commit stuff that should have gone in last week when I made the pre-releases:
2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:
0.10.20.2 pre-release
* po/LINGUAS:
* po/id.po:
* po/pt_BR.po:
New translations.
2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/: Recognise Kate subtitle streams (#550582).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
Recognise Kate subtitle streams (#550582).
2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
Remove trailing comma from enum list, which causes problems
with -pendantic (#550729).
2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
(gst_property_probe_get_properties),
(gst_property_probe_get_property),
(gst_property_probe_probe_property),
(gst_property_probe_probe_property_name),
(gst_property_probe_needs_probe),
(gst_property_probe_needs_probe_name),
(gst_property_probe_get_values),
(gst_property_probe_get_values_name),
(gst_property_probe_probe_and_get_values),
(gst_property_probe_probe_and_get_values_name):
More sanity checks for our second-favourite interface.
2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
Check for NULL pointer, in the hope that this fixes #532864.
2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
No really, the next release is 0.10.21 (fix Since: tags in docs).
2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
Disable a code path that is now called but causes a deadlock for some
reason and is unneeded.
2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
Add a "draw-border" property that can be set to false to disable
drawing borders.
* tests/icles/test-colorkey.c:
* tests/icles/Makefile.am:
Add new test application for the colorkey handling.
2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
This will also be fixed for upcoming gst-ffmpeg release so that once
this release of -base is out, it will work with the latest gst-ffmpeg
release.
2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add Truespeech mapping for RIFF formats (AVI/WAV).
Fixes #550656
2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Fixes #550638.
2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
Original commit message from CVS:
* configure.ac:
Disable subparse when xml is disabled. It woundn't work anyway. Fixes
test runs.
2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
po/POTFILES.in: Add some more files with strings for translation.
Original commit message from CVS:
* po/POTFILES.in:
Add some more files with strings for translation.
2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Use new geo location tags from core. Fixes #481169
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Use new geo location tags from core. Fixes #481169
2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Original commit message from CVS:
* tests/check/elements/audioresample.c: (setup_audioresample),
(fail_unless_perfect_stream), (test_perfect_stream_instance),
(test_discont_stream_instance):
Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Add debugging for coherence.
2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
Original commit message from CVS:
Patch by: Jonathan Matthew <notverysmart gmail com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for PDF documents (which is nice to have, since it's a
common format, but also helps prevent false positives). Fixes #549814.
2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
(no_more_pads_cb):
Fix nasty race where multiple decodebins could start pushing data before
we manage to configure the sinks, resulting in not-linked errors in
typical RTSP streaming cases.
2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Since we now call stop, we trigger this code path that causes a deadlock
is apparently not needed.
2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_stop):
Also allow the case where the ringbuffer was paused when we try to stop
it so that the basesrc stop function is still called.
2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
Original commit message from CVS:
Patch by: Mike Ruprecht <cmaiku at gmail dot com>
* sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
Reprobe devices again instead of taking a cached list as new
devices could've been plugged in. Fixes bug #549062.
2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
Original commit message from CVS:
Patch by: Alessandro Dessina <alessandro nnva org>
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain):
Don't add pads and activate them for skeleton streams. These are already
handled inside oggdemux. Fixes bug #537599.
2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
Reset variable so that query and convert fail after going back to
READY. Fixes #548898.
2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
If a buffer arrives with a timestamp before the timestamp+duration
of the previous buffer clip it instead of dropping it completely.
Slight improvement for the unfixable bug #548913.
2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
Take the current timestamp instead of timestamp+duration for the offset.
This offset will later be used for calculating the timestamp and
otherwise vorbisdec will interpolate timestamps wrong if upstream
only sends timestamps and no granulepos.
2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/seek.c: Don't crash when having no visualisations.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Don't crash when having no visualisations.
2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Fixes #548065.
2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
When cleaning up the caps fields also remove "depth" for the same
reason we remove "width".
2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
Add Lead H.264 here as well.
2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
Original commit message from CVS:
2008-08-14 Julien Moutte <julien@fluendo.com>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps): Add Lead H.264 variant.
2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
When not slaved to another clock also subtract the base_time from our
internal clock time to get the running time.
2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
Original commit message from CVS:
* ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
since it has no basis in libtheora.
2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.h:
Remove double "interface" from doc-string.
* gst-libs/gst/interfaces/xoverlay.h:
Document interface.
* gst-libs/gst/riff/riff.c:
Add basic doc blobs.
2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Don't try to build that example anymore.
2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
Original commit message from CVS:
* gst-libs/gst/audio/.cvsignore:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/make_filter:
Move audiofiltertemplate to gst-template.
2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
More docs and shuffling. What can we do with the hundreds of #defines.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiosrc.h:
More docs and shuffling. What can we do with the hundreds of #defines.
2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/: Reducing number of dundocumented symbols.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/interfaces/propertyprobe.h:
* gst-libs/gst/tag/gsttagdemux.h:
Reducing number of dundocumented symbols.
2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/audio.c: Fix doc comment syntax.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix doc comment syntax.
* gst-libs/gst/interfaces/propertyprobe.c:
Add more doc-comments and a FIXME: for the signal.
2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
(gst_ogg_mux_request_new_pad):
* ext/ogg/gstoggmux.h:
Don't pretend to support NEWSEGMENT events, instead override the
GstCollectPads event function to return FALSE on NEWSEGMENT events
and do the normal work for other events.
This prevents elements like flacenc to seek to the start and rewrite
some data which then results in a broken Ogg packet.
2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
* gst/playback/gstdecodebin.c: (plugin_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/gstqueue2.c: (plugin_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
* sys/v4l/gstv4l.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
Add audio/x-qdm for qtdemux.
2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/vorbis/vorbisdec.c: Do not leak old taglist.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
Do not leak old taglist.
2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/icles/test-scale.c: Include <stdlib.h> for atoi().
Original commit message from CVS:
* tests/icles/test-scale.c:
Include <stdlib.h> for atoi().
2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
Original commit message from CVS:
2008-08-04 Andy Wingo <wingo@pobox.com>
* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
documentation fix.
2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
Original commit message from CVS:
* gst/adder/gstadder.c:
Cleanup lots of empty lines that came from gst-indent going havoc
before I added the INDENT_ON/OFF marker some time agao.
2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Bump requirement to latest core and use new tag for riff formats.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/riff/riff-read.c:
Bump requirement to latest core and use new tag for riff formats.
Needed for #520694.
2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
Original commit message from CVS:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/codec-select.c: (make_encoder),
(make_pipeline), (do_switch), (my_bus_callback), (main):
Add example app that dynamically switches between 3 'encoders'.
2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Add some more comments.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
Add some more comments.
2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
(gst_video_test_src_create):
Discard buffers of the wrong size after renegotiation, this is perfectly
possible with things like capsfilter that could suggest caps changes
upstream without knowing the size of the buffer.
2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/icles/: Add dynamic rescaling tests for the new basetransform.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/test-scale.c: (make_pipeline), (main):
Add dynamic rescaling tests for the new basetransform.
2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Dist recently-added gstfastrandom.h.
2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Fix a "may be used uninitialized in this function" which weirdly only
appears on macosx (?).
2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Adding acid chunk for tempo and loop information.
2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
Original commit message from CVS:
* sys/xvimage/Makefile.am:
floor() needs linking to $(LIBM).
2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Aggregate short reads and add some comments and debug logging.
Fixes #537380
2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Fix property doc markup (its not a signal).
* sys/xvimage/xvimagesink.c:
Add since tag for new proeprties (also add sice tags fro the last two
other additions).
2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
Add autofill/colorkey properties. Fixes #538656.
2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Fix rounding errors when converting colorbalance values
between hardware and object property ranges. Partial
fix for #537889, however, there still seems to be a small
drift problem that could be totem's fault.
2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
This fixes a critical warning.
2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Allow muxing of CELT into Ogg streams.
2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (celt_type_find),
(plugin_init):
Add simple typefinder for the CELT codec (www.celt-codec.org).
2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
Original commit message from CVS:
Patch by: Jan Gerber <j at oil21 dot org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
Fix calculation of the start time from skeleton streams.
Fixes bug #530068.
2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither):
* gst/audioconvert/gstfastrandom.h:
Implement a linear congruential generator as pseudo random number
generator for the dither noise. This is about 2 times faster than
using GLib's mersenne twister. Also this uses only integer math for
generating integers while GLib internally uses floating point math.
2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
Original commit message from CVS:
* configure.ac:
Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
Original commit message from CVS:
Patch by: Damien Lespiau <damien.lespiau gmail com>
* gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
Use GST_STR_NULL to avoid crashes with libcs that don't
like NULL strings in printf args (such as the win32 one).
Fixes #544306.
2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
Oops - set the size of the image used for probing back to 1x1, for
consistency with ximagesink
2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/: it's not legal to ask the
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
Apparently on Solaris and OS/X (at least), it's not legal to ask the
X server to attach to a shared memory segment after we've deleted it,
with the result that MIT-SHM is disabled. Instead, remove it only after
X succeeds in attaching too.
2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
Add 'ticks', a 1/30 second sine wave pulse every second.
2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/video/video.c: Revert ABI change.
Original commit message from CVS:
* gst-libs/gst/video/video.c: Revert ABI change.
2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make it impossible to have NULL caps at the point where we set
framerate and other things. Also don't return immediately for "3ivd"
video and let framerate, etc be set. Might fix bug #542508.
2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
Video format can also be conveniently determined from (many)
non-fixed caps.
2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
First stab at integrating DVD subpicture overlay into
playbin. Successfully plugs and plays, but the queues need
shrinking - 3 seconds of video is too much buffering.
2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Remove now obsolete note in the docs.
2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
Original commit message from CVS:
* tests/examples/seek/Makefile.am:
Fix out of tree build by adding all required CFLAGS.
2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
And ref the pad before returning it again when linking to the queue
failed. Otherwise we will unref the pad twice later and things break.
2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
If linking the raw pad with a queue fails, try it without a queue
instead of failing completely. This should never happen.
2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
Original commit message from CVS:
Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
* gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
Add a queue after a demuxer if the demuxer outputs raw data. This was
done before only for non-raw data but is required in this case too.
Fixes bug #540215.
decodebin2 doesn't have this issue because all streams of a group
go through multiqueue.
2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* gst-libs/gst/sdp/gstsdpmessage.c:
Makes libgstsdp compile with mingw32 by defining the right WINVER so
that getaddrinfo() can be used. Fixes #541358.
2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_init),
(gst_video_test_src_set_property),
(gst_video_test_src_get_property), (gst_video_test_src_create):
* gst/videotestsrc/gstvideotestsrc.h:
Cleanups, use default property values as defines.
Add property to enable/disable peer buffer allocation.
2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Enable unit tests on PPC again as the bugs are now fixed.
2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
Fixes bug #540351.
2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
it on other formats. Also adjust the unit size only for that format
to not include the palette. Fixes bug #540497.
2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
Original commit message from CVS:
* gst/adder/gstadder.c:
Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ChangeLog: ChangeLog surgery.
Original commit message from CVS:
* ChangeLog:
ChangeLog surgery.
* tests/examples/seek/seek.c:
Move variable into ifdef too.
2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Include config.h and check if we have X. Fixes: #540334.
2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
Original commit message from CVS:
Patch by: Sam Morris <sam at robots dot org to uk>
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: Add "index" property to GstMixerTrack to differantiate between
multiple mixer tracks with the same label.
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set the "index" property of GstMixerTrack to the index given by ALSA.
Fixes bug #528299.
2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
Original commit message from CVS:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/seek.c:
Remove libgstvideo usage. Use gtk_get_option_group instead of
gtk_init().
2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/Makefile.am: Name the test registry format neutral.
Original commit message from CVS:
* tests/check/Makefile.am:
Name the test registry format neutral.
2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Do not double notify. Remove the unsued return value.
2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c:
Also consider "speaker" as a name for master volume. If that doesn't
help look for the first non-mono volume control that also has a
playback switch.
2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ChangeLog: Forgot to save the ChangeLog :/
Original commit message from CVS:
* ChangeLog:
Forgot to save the ChangeLog :/
2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/: Embedd the xwindow.
Original commit message from CVS:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/seek.c:
Embedd the xwindow.
2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
Original commit message from CVS:
* sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps):
* sys/ximage/ximagesink.h:
When the caps change, make sure to re-draw borders in
force-aspect-ratio=true mode.
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
Don't clear the border_draw flag until we actually draw the border.
* tests/check/Makefile.am:
Ignore alsasink/src during the states test too, so it doesn't fail
when running without access to the sound device.
2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Fix crasher when playing a parse-launch line the 2nd time.
2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c:
Properly ifdef tests to fix compilation.
2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
break long lines
Original commit message from CVS:
break long lines
2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
Add get-video-pad, get-audio-pad, get-text-pad action signals to
playbin2. This allows the user to get to the selector's sinkpads, and
thus inspect a range of things - caps, tags, etc.
2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Use a different constant for the convert-frame signal id.
Fixes #537009.
2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
gst/playback/: Fix a whole bunch of typos in comments and log statements.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
Fix a whole bunch of typos in comments and log statements.
2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Don't set colour balance values on the Xv port if the user hasn't
changed them (via properties or the interface). Avoids accumulating
rounding errors for the common case.
Partial fix for bug #537889.
2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Ensure decodebin2 emits 'drained' signal once, and only once, when all
pads are drained.
2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/README:
apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
Original commit message from CVS:
apparently it's an error to specify nc -l -p 3000 - though the short usage
does not make it very clear that you can drop the host arg with -l
2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
(gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
Report the encoder latency. Fixes #538232.
2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
(notify_source), (activate_group):
Implement the source property, emit notify when it changes in the
underlying uridecodebin.
2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb):
Free and clear the seek element list so that we don't use invalid
references when seeking after recreating a gst-launch line.
2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Bump verion back to devel -> 0.10.20.1
Original commit message from CVS:
* configure.ac:
Bump verion back to devel -> 0.10.20.1
2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't increase the size of non-string image buffers by one as this
might in theory confuse decoders. Still increase it by one for string
image buffers to append '\0'.
2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
Fix a buffer memleak and remove a confusing and wrong debug output.
Fixes bug #538663.
2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/appsink-src.c: Don't use a buffer after unreffing it.
Original commit message from CVS:
* examples/app/appsink-src.c: (on_new_buffer_from_source):
Don't use a buffer after unreffing it.
=== release 0.10.20 ===
2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* po/LINGUAS:
* win32/common/config.h:
Release 0.10.20
Original commit message from CVS:
Release 0.10.20
2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/it.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
Original commit message from CVS:
Update .po files
2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* examples/app/appsrc-ra.c:
* examples/app/appsrc-seekable.c:
* examples/app/appsrc-stream.c:
* examples/app/appsrc-stream2.c:
* ext/directfb/dfbvideosink.h:
* ext/metadata/gstbasemetadata.c:
* ext/metadata/gstbasemetadata.h:
* ext/metadata/metadata.c:
* ext/metadata/metadataexif.c:
* ext/theora/theoradec.h:
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/speedy.c:
* gst/deinterlace2/tvtime/speedy.h:
* gst/deinterlace2/tvtime/vfir.c:
Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
comments.
2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/app/gstappsrc.c:
gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
Original commit message from CVS:
2008-06-16 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Final round of doc updates.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.
2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
examples/app/.cvsignore: Ignore more.
Original commit message from CVS:
* examples/app/.cvsignore:
Ignore more.
2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: 0.10.19.3 pre-release
Original commit message from CVS:
* configure.ac:
0.10.19.3 pre-release
2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fix build on win32.
Patch By: David Schleef <ds@schleef.org>
Fixes: #536874
2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
Original commit message from CVS:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
(gst_gio_base_src_create):
* ext/gio/gstgiobasesrc.h:
Try to read the requested number of bytes, even if the first
read returns less than requested, until nothing is read anymore
or we have the requested amount of bytes. This fixes playback of
files via Samba as Samba only allows to read 64k at once.
Implement a caching algorithm that makes sure that we read at
least 4k of data every time. Some elements will try to read a few
bytes, then seek, read again a few bytes and so on and this is
painfully slow as every operation has to go over DBus if GVfs is
used as backend.
Fixes bug #536849 and #536848.
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
(gst_gio_src_check_get_range):
Override check_get_range() to blacklist http/https URIs
and whitelist file URIs. More to be added on demand.
2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
Original commit message from CVS:
* examples/app/appsrc-stream.c: (found_source), (main):
Use deep-notify until we can depend on a playbin2 with support for the
source property.
2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_set_property), (gst_app_src_get_property),
(gst_app_src_unlock), (gst_app_src_unlock_stop),
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: 0.10.19.2 pre-release
Original commit message from CVS:
* configure.ac:
0.10.19.2 pre-release
2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
win32/common/: Add new API functions to the dll exports
Original commit message from CVS:
* win32/common/libgstrtsp.def:
* win32/common/libgsttag.def:
Add new API functions to the dll exports
2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes #536521.
2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).
2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
Original commit message from CVS:
* tests/check/Makefile.am:
Do not try to run the check tests for subparse unless it has been
built.
2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
Do not try to run a test which requires vorbisenc unless we have
actually built it.
2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params),
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add a couple of missing argument guards.
Add a way of setting the DSCP for an RTSP connection.
Add an accessor method for the ip member of GstRTSPConnection as all
members are supposed to be private.
2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
Fixed accidental use of IPv4 options for all IPv6 addresses.
2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Document mixer track flags.
2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.
2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.
2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.
2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
Original commit message from CVS:
Based on patch by: John Millikin <jmillikin gmail com>
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
(gst_vorbis_tag_add_coverart):
Retrieve COVERART tags from vorbis comments (#512333)
2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Don't forget to add new enum value here too (should probably use
glib-mkenums here...).
2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
(gst_tag_image_data_to_image_buffer):
Add two utility functions to avoid code duplication (#512333):
API: add gst_tag_image_data_to_image_buffer()
API: add gst_tag_list_add_id3_image()
2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_check_channel_positions() to the exported symbols.
2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
API: Make gst_audio_check_channel_positions() public.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().
2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
Original commit message from CVS:
* sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
minrange and maxrange are scaled according to the frequency
multiplier.
2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
Original commit message from CVS:
* ext/pango/Makefile.am:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
(gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
Use gstvideo functions to calculate strides and plane offsets. Fixes
rendering issue ('ghost' images of the text on the chroma planes)
with widths or heights that are not multiples of 8 (#506659 and
probably also #485729).
* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
(main):
Test with odd height/width too.
2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.
2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ChangeLog:
ChangeLog surgery, mark API change
Original commit message from CVS:
ChangeLog surgery, mark API change
2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.
2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_clock_reset to the list of exported symbols.
2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).
2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.
2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
Add sane defaults for the 7 and 8 channel layouts as those are
undefined in the Vorbis spec. Use NONE channel layouts when decoding
more than 8 channels instead of erroring out. Fixes bug #535356.
2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
Add theoraparse to the docs and fix some docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/theora/theoraparse.c:
Add theoraparse to the docs and fix some docs.
2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
Fix EOS condition and track addition check, the track.end sector is
included in the track. Fixes #533265.
2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.
2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/examples/seek/seek.c: Initialise error to NULL as we should.
Original commit message from CVS:
* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
Initialise error to NULL as we should.
2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/adder/gstadder.c: Implement latency query.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.
2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
Original commit message from CVS:
* win32/vs6/libgstpbutils.dsp:
Add pbutils-enumtypes.c to sources (#518037).
2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).
2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/oggmux.c: Don't use deprecated function.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (test_pipeline):
Don't use deprecated function.
2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.
2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/video.c: More checks.
Original commit message from CVS:
* tests/check/libs/video.c:
More checks.
2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.
2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.
2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Add some documentation comments, and some new headers to be scanned.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.
2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.
2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.
2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
Original commit message from CVS:
* docs/Makefile.am:
Fix installing plugin documentation when gtk-doc is disabled.
2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
Original commit message from CVS:
* gst-libs/gst/rtsp/Makefile.am:
Distribute, don't install md5.h
2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Original commit message from CVS:
2008-05-21 Julien Moutte <julien@fluendo.com>
* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
Some debug and comment fixes.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.
2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/design/part-playbin2.txt: Add some leftover doc.
Original commit message from CVS:
* docs/design/part-playbin2.txt:
Add some leftover doc.
2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.
2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.
2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.
2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.h:
Make the GstRTSPTransport struct members public as there are no
setters/getters and it's supposed to be changed directly.
Fixes bug #533087.
2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
Original commit message from CVS:
* configure.ac:
Require core CVS for GstBaseSrc buffer caps setting magic.
2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.
2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.
2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Error out if we don't have the required version of core.
Original commit message from CVS:
* configure.ac:
Error out if we don't have the required version of core.
2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Better debug logging in port value handling. Merging separate port
value loops into one.
2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.
2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstrtsp.def:
Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
symbols.
2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.
2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.
2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
Original commit message from CVS:
Patch by: j^ <j at oil21 dot org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
(gst_ogg_pad_parse_skeleton_fisbone):
* ext/ogg/gstoggdemux.h:
Parse presentation time from skeleton streams and use it as offset
for the timestamps. Fixes bug #530068.
2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
Fix docs: type and missing word.
2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tunerchannel.c:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.c:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Document the GstTuner and GstColorBalance interfaces, and some
other random API functions that needed it. 70% symbol coverage, woo.
2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.
2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
Original commit message from CVS:
* win32/common/config.h.in:
Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
use the real thing than having "???" unconditionally.
2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
Original commit message from CVS:
* gst-libs/gst/app/.cvsignore:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp-marshal.list:
Add marshal.list, make it compile and add to cvsignore.
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
(gst_app_sink_stop):
Small cleanups.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_create), (gst_app_src_set_caps),
(gst_app_src_get_caps), (gst_app_src_set_size),
(gst_app_src_get_size), (gst_app_src_set_seekable),
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Beat appsrc in shape, add signals and actions.
Add some docs.
Add properties for caps, size, seekability and max-buffers.
Fix unlock/stop code.
2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_base_audio_src_[sg]et_slave_method() to the exported
symbols.
2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.*: Start some docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
It's SorensOn and not SorensEn.
2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Fix description of video/x-flash-video.
2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.
2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.
2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.
2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
* ChangeLog:
* gst/videotestsrc/videotestsrc.c:
Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes #526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: Cool kids don't divide by zero.
Original commit message from CVS:
* ext/theora/theoradec.c:
Cool kids don't divide by zero.
Treat PAR of x:0 as 1:1.
Fixes #530719.
2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.*: Add more docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix the docs about the seqnum compare function, it returns a difference.
2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_get_device_list): Don't return before freeing up
the allocated structures.
2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Remove obsolete streaminfo code and fix a leak. Fixes #529546
2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
Revert the event part, that should not go in.
2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
Don't leak GstPluginFeatures when filtering.
2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Add some logging for cases when grabbing the xv failed.
2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
packet. Should conform to what we currently think is the
final Ogg/Dirac muxing spec.
2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Fix typo that causes the overlay keying color to bright green
on a 16-bit display. Dark grey good. Bright green bad.
2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfsuri.c:
Add FIXME comment about using uri-list for source and sink.
2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
vaargs functions to gint. Otherwise the fractions will get 0 set
instead of the correct value on big endian systems. Fixes bug #529018.
2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnome_vfs_sink_uri_get_protocols):
* ext/gnomevfs/gstgnomevfssrc.c:
(gst_gnome_vfs_src_uri_get_protocols):
* ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
(gst_gnomevfs_get_supported_uris):
Get the list of supported URI schemes in a threadsafe way and use the
same list for the source and sink.
2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
Original commit message from CVS:
* ext/gio/gstgio.c: (_internal_get_supported_protocols),
(gst_gio_get_supported_protocols):
Don't generate a new supported protocols list on each call but cache
it. It's supposed to be static anyway, this way we only leak it once
per process.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_finalize),
(gst_gio_sink_set_property), (gst_gio_sink_get_property),
(gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_finalize),
(gst_gio_src_set_property), (gst_gio_src_get_property),
(gst_gio_src_start):
* ext/gio/gstgiosrc.h:
API: Add "file" properties where one can set a GFile as source/destination.
Add locking to the properties and use gst_element_class_set_details_simple()
instead of a static GstElementDetails struct.
2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Add "mpp" and "mp+" as possible extensions for MusePack files.
Add typefinding for MusePack StreamVersion 8 files and include the
stream version in the caps.
2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
(gst_rtp_payload_info_for_name):
Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
Original commit message from CVS:
* configure.ac:
Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
(NB: this only affects compilation of some of the examples).
Remove some configure.ac cruft that's not needed any longer.
2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Don't validate the payload if there isn't any.
Fixes #525915
2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
Use g_atomic_int_set() instead of gst_atomic_int_set().
2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Return NULL instead of a gchar * array with one NULL element if we
don't get any supported URI schemes from GIO.
2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Remove cpp style commented old code.
2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gstdecodebin2.c: Fix signal docs.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix signal docs.
2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
(gst_text_overlay_init):
Fix textoverlay unit test again by making the supposed default
value for the wait-text property the actual default value.
Also fix Since: tag for new property.
2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_new_caps),
(gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size),
(gst_video_format_convert):
Add guards to these functions to ensure sane input values.
* tests/check/libs/video.c:
Fix unit test not to create caps with width=0 and height=0.
2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/design/draft-keyframe-force.txt: Fix typo.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
Fix typo.
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_handle_src_query):
Set buffering mode in the messages.
Set buffering percent in the query.
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
(do_stream_buffering), (do_download_buffering), (msg_buffering):
Do some more fancy things based on the buffering method in use.
2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
Original commit message from CVS:
* tests/examples/seek/seek.c: (update_fill), (set_update_fill),
(play_cb), (pause_cb), (stop_cb), (msg_state_changed),
(msg_buffering), (main):
Add basic download reports to seek using the new buffering API.
2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
(gst_queue_src_checkgetrange_function):
Include extra buffering stats in the buffering message.
Implement BUFFERING query.
* gst/playback/gsturidecodebin.c: (do_async_start),
(do_async_done), (type_found), (setup_streaming), (setup_source),
(gst_uri_decode_bin_change_state):
Only add decodebin2 when the type is found in streaming mode.
Make uridecodebin async to PAUSED even when we don't have decodebin2
added yet.
2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Filter cdda from the supported URI schemes. We can't support
musicbrainz tags and everything else one expects from a cdda source
with GIO. Fixes bug #526794.
2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* sys/xvimage/xvimagesink.c:
Fix calculation of 'expected size' for YV12 buffers.
Original commit message from CVS:
2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_buffer_alloc):
Fix calculation of 'expected size' for YV12 buffers.
Be a little more verbose in the debug output for buffer-alloc'ed
buffers which turn out to have the wrong size.
2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
Fix calculation of 'expected size' for YV12 buffers.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_buffer_alloc):
Fix calculation of 'expected size' for YV12 buffers.
Be a little more verbose in the debug output for buffer-alloc'ed
buffers which turn out to have the wrong size.
2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
Merge other changes from 0.10.19 release branch.
Original commit message from CVS:
* NEWS:
* RELEASE:
* gst-plugins-base.doap:
Merge other changes from 0.10.19 release branch.
2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove some more fields.
2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
configure.ac: Actually build dlls when cross-compiling with mingw32.
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* configure.ac:
Actually build dlls when cross-compiling with mingw32.
Fixes bug #526247.
2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
Original commit message from CVS:
* configure.ac:
Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add statusbar.
Original commit message from CVS:
* tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
(msg_buffering), (connect_bus_signals), (main):
Add statusbar.
Add buffering support with feedback in the statusbar.
2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstoggmux.c: Fix sample pipeline description.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Fix sample pipeline description.
2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update introspection data.
* ext/ogg/gstoggmux.c:
Document oggmux.
* gst/playback/gstdecodebin2.c:
Don't use gtk-doc style comment start for private stuff, but make it
formatted like this for consistency.
2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
(gst_decode_bin_set_property), (gst_decode_bin_get_property),
(analyze_new_pad), (connect_pad), (expose_pad),
(gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
(gst_decode_group_expose), (gst_decode_group_free),
(do_async_start), (do_async_done), (gst_decode_bin_change_state):
Remove fakesink hack, we can now implement this more elegantly.
Added property to bypass typefinding.
Removed underrun callback and demuxer pad probe, we now use the srcpad
probe to expose groups.
API::sink-caps property
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Guard against multiple emissions of the no_more_pads signal, which
happens when we are dealing with chained oggs.
* gst/playback/gsturidecodebin.c: (remove_decoders),
(make_decoder), (type_found), (setup_streaming), (source_new_pad),
(setup_source):
For streams, use our own typefind element and plug our queue after it.
We will need this to determine the type of buffering to use for the
queue soon.
2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
Guard against over and underflows because of clock slaving.
When we are using our own clock, still compensate for any calibrations
that we might have done to our clock.
2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet),
(theora_dec_chain):
Don't try to do anything fancy with the return code from pushing an
event, it does not have enough information to turn it into a
GST_FLOW_ERROR.
2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Add small debug line.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
(gst_ogg_demux_chain_elem_pad):
Add small debug line.
Pass return code from the internal decoder instead of the too generic
GST_FLOW_ERROR.
2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
Original commit message from CVS:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/base64.c:
* gst-libs/gst/cdda/base64.h:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cddabasesrc_calculate_musicbrainz_discid):
Use GLib's base64 implementation instead of our own.
2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
(gst_ogg_demux_read_chain):
Refix oggdemux, we only have a problem if we failed to find a chain and
we are not EOF.
2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
ext/ogg/gstoggdemux.c: When we fail to find a BOS page and we and up with no chain, error out properly instead of seg...
Original commit message from CVS:
Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
(gst_ogg_demux_read_chain):
When we fail to find a BOS page and we and up with no chain, error out
properly instead of segfaulting. Fixes #525665.
2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
(gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
The new-pad-group sequence is add-pads, no-more-pads, add-pads,
no-more-pads...
2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_out_rates),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_set_property):
Update the estimated input data when we push out a buffer.
Add some debug info about the temp file.
Only forward src events when we are not using a temp file.
Don't block the duration query, we need to find something better.
Don't leak the temp filename.
2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: Require GLib 2.12 and liboil 0.3.14.
Original commit message from CVS:
* configure.ac:
Require GLib 2.12 and liboil 0.3.14.
* gst/volume/gstvolume.c: (volume_process_double):
Unconditionally use liboil 0.3.14 function.
2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
ms-gsm can have arbitrarty sample rates. See #481354.
2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
MP4S is generic MPEG-4, not a microsoft variant.
2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Check the body CRC (if set) when depayloading.
Fixes #522401.
2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Fix Since: version for new property.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
Fix Since: version for new property.
2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
Don't error when poll_wait returns EAGAIN.
2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_is_filled):
The queue is never filled when there are no buffers in the queue at all.
Fixes #523993.
2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Update some docs.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (free_group), (gst_play_bin_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
(gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_encoding), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb), (perform_eos), (autoplug_select_cb),
(activate_group), (deactivate_group), (setup_next_source),
(save_current_group), (gst_play_bin_change_state):
Update some docs.
Add new locks and conds to protect pipeline creation and group
switching.
Implement the sub-uri property.
Keep track of pending uridecodebin creation and configure the output
pipeline after all streams are configured.
Propagate subtitle encoding to the uridecodebins.
Implement getting the video/audio/visualisation elements.
Use input-selector for stream switching.
If we are asked to do visualisation, prefer to autoplug raw sinks
instead of sinks that accept encoded data.
2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
(gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
(gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
(gst_play_sink_set_volume), (gst_play_sink_get_volume),
(gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
(gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add methods to get audio/video/vis elements.
Add methods to set the font description for the overlay.
Remove properties, we're using this element with its methods only.
Add support for subtitles.
Rearrange the locking a bit to not use the object lock for protecting
the pipeline construction.
Try to use the volume and mute property on the sink when its available.
Implement the mute option with volume when the sink does not have a mute
property.
Only add volume element when the sink has no volume property.
Only do visualisations with raw audio pads.
2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
(gst_text_overlay_init), (gst_text_overlay_set_property),
(gst_text_overlay_get_property), (gst_text_overlay_src_event),
(gst_text_overlay_text_event), (gst_text_overlay_video_event),
(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
(gst_text_overlay_change_state):
* ext/pango/gsttextoverlay.h:
Add property to configure waiting for text on the textpad or not, with
the default behaviour being the old one (always wait for text before
rendering the video). This default behaviour is usually not the best one
because the text stream can very sparse and could require queueing a lot
of video.
Fix the flushing and EOS handing so that we don't mix up their meaning.
2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_factories),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
(gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (no_more_pads_full),
(new_decoded_pad_cb), (gen_source_element), (remove_decoders),
(proxy_autoplug_factories_signal), (make_decoder),
(source_new_pad), (setup_source):
Add a readonly source property and notify.
Add new lock for protecting the construction of the pipeline.
Keep track of the decodebins we plugged.
Correctly proxy the autoplug signal so that it actually continues.
Proxy subtitle-encoding to the decodebins.
2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
Original commit message from CVS:
* tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (main):
Rearrange some buttons in playbin2 and make some other boxes insensitive
when needed.
Add language codes to subtitle selection boxes when we gind the right
tags for the streams.
2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding),
(gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
(deactivate_free_recursive):
Protect caps property with the object lock.
Protect encoding property with the object lock.
Keep list of elements we added that have the subtitle-encoding property.
Distribute the subtitle-encoding to all of the elements when it
changes.
2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
Small debug improvement.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix bug in determining the sample start/stop position, we want to base
this decision on the fact that we are going forwards or backwards, not
slower or faster. This fixes some ugly resync warnings when playing at
very slow speeds.
2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Correctly set the supported URI schemes and don't leave
some schemes in the middle or at the start at NULL.
2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c:
Make test compile without unused function/variable warnings on PPC.
2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Filter http and https protocols. GIO/GVfs handles them but it's
impossible to implement iradio/icecast with it. Better use
souphttpsrc or something else for this.
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
If getting the file informations by a query fails try it with the
seek-to-end trick too.
2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_base_init), (gst_volume_class_init),
(volume_process_double), (volume_process_float),
(volume_transform_ip), (plugin_init):
memset buffers to zero if we get a GAP buffer. We usually see a
buffer as one unit so let's handle it as one and don't care about
volume changes while processing one buffer.
Also clean up some stuff a bit.
2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_create_silence_buffer),
(gst_audio_convert_transform):
Make audioconvert GAP-aware by outputting silence buffers when the
input has the GAP flag set. This is up to 8x faster.
Based on a patch by Stefan Kost. Fixes bug #517813.
2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_double):
Use oil_scalarmultiply_f64_ns() for double processing when it's
available at compile time.
2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
Original commit message from CVS:
* configure.ac:
Fix lrint/lrintf checks to actually work. These functions are
in libm on Linux at least so try to link to it.
2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to development - 0.10.18.1
Original commit message from CVS:
* configure.ac:
Back to development - 0.10.18.1
=== release 0.10.18 ===
2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* po/LINGUAS:
* win32/common/config.h:
Release 0.10.18
Original commit message from CVS:
Release 0.10.18
2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/hu.po:
* po/it.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
Original commit message from CVS:
Update .po files
2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
0.10.17.4 pre-release
Original commit message from CVS:
* configure.ac:
* win32/common/config.h:
0.10.17.4 pre-release
2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
Use GST_STR_NULL when trying to print strings that could be NULL because
this might crash on some platforms. See #520808.
2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(read_line), (gst_rtsp_connection_read_internal):
Generic Windows fixes that makes libgstrtsp work on Windows when
coupled with the new GstPoll API. See #520808.
2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
Original commit message from CVS:
Patch by: Milosz Derezynski <internalerror at gmail dot com>
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
If seeking to a new position succeeds don't simply return from
create() without creating a buffer. Do this only in the case
seeking to the new position fails. Fixes bug #523054.
2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
(gst_video_format_from_rgba32_masks):
Fix gst_video_format_parse_caps() for RGB caps with alpha channel
(#522635).
* tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
Add unit test for the RGB caps parsing and creation, checking for
internal consistency of the new API and consistency of the API with
the old GST_VIDEO_CAPS_* defines.
2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: Oops, revert last change
because -base is in freeze.
2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
Original commit message from CVS:
Patch by: William M. Brack
* gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Revert change that caused regression until a real fix is found.
Fixes #522203.
2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h:
Rename recently added buffer types to make more sense.
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
(gst_alsasink_write):
Adapt for above API changes.
Fixes bug #520523.
2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
Original commit message from CVS:
* win32/common/libgstnetbuffer.def:
Add new symbol gst_netaddress_equal. Fixes bug #521743.
2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
0.10.17.3 pre-release
Original commit message from CVS:
* configure.ac:
* win32/common/config.h:
0.10.17.3 pre-release
2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes #520300.
2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
Add trivial function to compare GstNetAddress. See #520626.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Add trivial function to compare GstNetAddress. See #520626.
API: GstNetBuffer::gst_netaddress_equal
2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Update mode property docs, it's deprecated now.
2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/: Remove GstPollMode from gstpoll constructor.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create):
* gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
Remove GstPollMode from gstpoll constructor.
2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
0.10.17.2 pre-release
Original commit message from CVS:
* configure.ac:
* win32/common/config.h:
0.10.17.2 pre-release
2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
Original commit message from CVS:
* gst/Makefile.am:
GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
them twice
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Add new API to the defs
2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
Original commit message from CVS:
Patch by: Mersad Jelacic <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/mixer.c: Add a few simple checks for the new message types.
Original commit message from CVS:
* tests/check/libs/mixer.c:
Add a few simple checks for the new message types.
2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
(gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
(gst_mixer_message_get_type),
(gst_mixer_message_parse_option_changed),
(gst_mixer_message_parse_options_list_changed):
* gst-libs/gst/interfaces/mixer.h: (GstMixerType),
(GST_MIXER_MESSAGE_OPTION_CHANGED),
(GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
(GST_MIXER_MESSAGE_MIXER_CHANGED):
API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
(gst_mixer_options_get_values):
* gst-libs/gst/interfaces/mixeroptions.h:
(GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
(_GstMixerOptions), (_GstMixerOptionsClass):
API: add GstMixerOptions::get_values vfunc (#519906)
2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
Original commit message from CVS:
* configure.ac:
Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
plug-ins are included/excluded. (#498222)
2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for IMelody files, using audio/x-imelody.
See bug #519516.
2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst/playback/gstplaybin2.c:
Make the function signature of the _get_*_tags() functions match
the signature of the vfuncs they implement, ie. return a
GstTagList rather than a GstStructure, which is more correct,
even if one is typedef'ed to the other (#518940).
2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Don't include unix headers unconditionally (fixes #518037).
2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
Original commit message from CVS:
* tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
(fourcc_list_struct), (fourcc_list), (fourcc_get_size),
(paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
(paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
(paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
(paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
(gst_video_format_is_packed), (video_format_is_packed):
Add unit test that makes sure that the strides, offsets and
sizes returned for the various YUV formats by the new video API
match the old reference implementation in videotestsrc.
2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
(gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
(gst_video_format_is_rgb), (gst_video_format_is_yuv),
(gst_video_format_has_alpha), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size):
* gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
(GST_VIDEO_FORMAT_Y42B):
API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
YV12 is I420 with swapped components 1 and 2, so the offset of
component 1 for I420 should be the offset for component 2 for YV12
and vice versa.
2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
Original commit message from CVS:
* sys/v4l/gstv4lelement.c:
Add missing semicolon to fix indentation.
2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
Original commit message from CVS:
2008-02-29 Julien Moutte <julien@fluendo.com>
* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
(gst_mixer_message_parse_mute_toggled),
(gst_mixer_message_parse_record_toggled),
(gst_mixer_message_parse_volume_changed),
(gst_mixer_message_parse_option_changed):
De-cruft and fix message type assertions (NULL is not a really
valid mixer message type string).
2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_vis_src_negotiate):
When negotiating, actually start from a format that we can support
instead of from the too generic template.
2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Enable vis setting.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
Enable vis setting.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gen_vis_chain):
Implement vis switching while playing.
2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/riff/riff-media.c: Add Dirac mapping
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: Add Dirac mapping
2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/tcp/Makefile.am:
* gst/tcp/fdsetstress.c:
* gst/tcp/gstfdset.c:
* gst/tcp/gstfdset.h:
Removed fdset and stress test, they are now known as GstPoll in
core.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
(gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_queue_buffer),
(gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
(gst_multi_fd_sink_stop):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
(gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
(gst_tcp_gdp_read_caps):
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
(gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
(gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
(gst_tcp_client_src_create), (gst_tcp_client_src_start),
(gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
(gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
(gst_tcp_server_src_create), (gst_tcp_server_src_start),
(gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
* gst/tcp/gsttcpserversrc.h:
Port to GstPoll. See #505417.
2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
* ChangeLog:
Patch Changelog a bit to give credit and refer to the relevant bug.
Original commit message from CVS:
Patch Changelog a bit to give credit and refer to the
relevant bug.
2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_poll),
(gst_rtsp_connection_flush):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Use GstPoll for the rtsp connection.
2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
Original commit message from CVS:
* tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
(init_visualization_features), (vis_combo_cb), (shot_cb), (main):
Add combo box for visualisations, populate it with a factory list
of all visualisation plugins, configure vis plugin instance in
playbin2.
2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
Original commit message from CVS:
* tests/check/libs/rtp.c: (GST_START_TEST):
Add check for RTP buffer defaults, padding and marker bit API.
2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
Original commit message from CVS:
* gst-libs/gst/cdda/sha1.c: (sha_transform):
Use memcpy() instead of upcasting a byte array to long *. This
fixes an unaligned memory access, resulting in SIGBUS on IA64.
This should be ported to GCheckSum once we can use GLib 2.16.
Partially fixes bug #500833.
2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
Push tag event after the newsegment event. Log the pointer of
the buffer we're actually going to push rather than the buffer
we're feeding to _make_metadata_writable().
2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Comment smoke typefinder for now. The smokedec plugin needs one
frame per buffer but we have no parser yet, thus it simply crashes
in most situations.
2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for the smoke video codec. Copied from the jpeg plugin.
2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mid_type_find),
(plugin_init):
Add midi typefinder, copied from the timidity plugin.
2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
Original commit message from CVS:
* tests/check/gst-plugins-base.supp:
Add a suppression for a cached value in GIO that wasn't moved
while moving gio from -bad to -base.
2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* configure.ac:
Don't hardcode -Wall and -Werror for configure checks, this fails
with non-GCC compilers. Fixes bug #517991.
2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnome_vfs_sink_handle_event):
Return FALSE when seeking for a new segment fails instead
of silently ignoring the failure and appending every buffer
that comes for the new segment.
2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (find_property),
(gst_play_sink_find_property), (gen_video_chain),
(gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
Recursively search the sink element for a last-frame property so that we
can also find the property in autovideosink and friends that don't
always proxy the internal sink properties.
2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
Fix confusing terminology in docs and code: structure fields are
'fields' and not 'properties'.
2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions), (add_list_to_struct):
Give more useful warning messages if one of the channel
layout enums passed to us is invalid and if the "channels"
field in the caps has a GType we don't expect.
2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Fix typo in docs blurb.
2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
Original commit message from CVS:
2008-02-19 Julien Moutte <julien@fluendo.com>
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
typefind lookup to fix typefinding on HD clips.
2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
Original commit message from CVS:
* gst/playback/gstscreenshot.c:
* gst/playback/gstscreenshot.h:
Fix up copyright (I rewrote the GStreamer-0.10 code for
this from scratch back in the days).
2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Add screenshot conversion code from totem.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
(create_element), (gst_play_frame_conv_convert):
* gst/playback/gstscreenshot.h:
Add screenshot conversion code from totem.
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
(gst_play_bin_class_init), (gst_play_bin_convert_frame),
(gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
Implement frame property to get a color-unconverted snapshot.
Implement convert-frame action signal to get a converted snapshot image.
Configure connection speed in uridecodebin.
Document some more properties.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_get_last_frame):
* gst/playback/gstplaysink.h:
Use last-buffer property of the video sink to get a video snapshot.
* tests/examples/seek/seek.c: (shot_cb), (main):
Add snapshot button for playbin2 and use the frame property to save the
frame as a png in the current directory.
2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
Original commit message from CVS:
Patch by: Josep Torra Valles <josep at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
(plugin_init):
Add typefinding support for h264 elementary streams.
Fixes bug #517420.
2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
configure.ac: Require CVS of core for new API in collectpads.
Original commit message from CVS:
* configure.ac:
Require CVS of core for new API in collectpads.
* gst/adder/gstadder.c:
Use new API to make adder sparse stream aware.
2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb):
Get the object data correct so that we can remove our channels
correctly.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Add option to disable async behaviour in the sinks when possible. This
makes it possible to avoid an audio queue when dealing with
visualisations.
Add option to add a queue for the audio path.
* tests/examples/seek/seek.c: (clear_streams), (update_streams),
(main):
Disable the vis checkbox to match the defaults of playbin2.
Only get the stream info when we need to.
2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/: Don't use async operations as they require a running main loop.
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
(gst_gio_base_src_set_stream):
* ext/gio/gstgiosink.c: (gst_gio_sink_start):
* ext/gio/gstgiosrc.c: (gst_gio_src_start):
Don't use async operations as they require a running main loop.
This makes us block again when closing streams and unable
to mount the enclosing volume of an URI if it isn't yet.
2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Move tee in front of the audio and vis pipelines.
Add queue for audio for now.
Add visualisation support.
* tests/examples/seek/seek.c: (main):
Visualisation is by default disabled.
2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/: Improve debugging a bit.
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (close_stream_cb):
* ext/gio/gstgiobasesrc.c: (close_stream_cb):
Improve debugging a bit.
* ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Try to mount the enclosing volume of a GFile if it isn't mounted
yet. This requires us to wait for an async operation to finish, done
with an nested GMainLoop. Authentication is not supported yet, will
come later.
2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Add mute property.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_audio_chain):
* gst/playback/gstplaysink.h:
Add mute property.
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Make sure we forward the event only once.
* tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
Add and implement the mute button for playbin2.
2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
ext/alsa/gstalsasink.c: Add some more debug info.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
Add some more debug info.
Make sure we never return a negative delay. Fixes #516246.
2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
Revert patch that makes the sink hold the object lock when
calling snd_pcm_delay(), since it breaks playback for me.
2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Add some seek flags when changing rate.
Original commit message from CVS:
2008-02-12 Julien Moutte <julien@fluendo.com>
* tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
some seek flags when changing rate.
2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
Fix potential leaks.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
Fix leak when there is no function configured.
2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
Original commit message from CVS:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
(gst_v4lsrc_buffer_finalize):
Correctly chain up the finalize method.
2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
Original commit message from CVS:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
Add documentation and example code for giostreamsink/giostreamsrc.
* tests/check/pipelines/gio.c: (GST_START_TEST):
Ask the GMemoryOutputStream for the data instead of assuming that
the pointer to the data stayed the same. It could've been realloc'ed.
2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
Original commit message from CVS:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
Make the documentation of giosink/giosrc complete, large parts
are based on the gnomevfssink/gnomevfssrc docs.
2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
docs/plugins/: Add the GIO documentation again and while at that run make update.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
Add the GIO documentation again and while at that run make update.
2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/: Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling against libasound >= 1.0.16, since it's be...
Original commit message from CVS:
* ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
* ext/alsa/gstalsasink.c: (set_swparams):
* ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
against libasound >= 1.0.16, since it's been deprecated in
0.10.16, and alignment is always 1 then, apparently. (#512899)
2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
* gst/playback/gstplaysink.c: (gen_audio_chain):
Handle case where we can't create the volume element a bit
better (#514307).
2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/: Add support for https protocol. Fixes #510229.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
Add support for https protocol. Fixes #510229.
2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
Original commit message from CVS:
2008-02-11 Julien Moutte <julien@fluendo.com>
Patch by: Alan Peevers <peeves@pacbell.net>
* ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
lock when calling alsa methods.
2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Bump rank of jpeg and png typefinders, which will return maximum
probability in the most common cases (thus short-circuiting more
expensive typefinders like the mp3 one for these two quite common
image types).
2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
Original commit message from CVS:
* ext/theora/theoraparse.c:
Fix long description of the theora parser to be more verbose than just
the type name.
2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
Original commit message from CVS:
Patch by: Branko Čibej <brane at xbc dot nu>
* sys/xvimage/xvimagesink.c:
Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
Fixes bug #515654.
2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Set is_dynamic as True if there are elements with both request
and sometimes src pad templates instead of breaking out when it
finds the first pad template that is a src.
2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb), (clear_streams),
(update_streams), (video_combo_cb), (audio_combo_cb),
(text_combo_cb), (volume_spinbutton_changed_cb), (main):
Add some stream switching and volume gui for playbin2.
2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
Added marshal for streamselector Tags.
* gst/playback/gstplaybasebin.c: (set_active_source):
Streamselector now selects pads based on the pad object instead of its
name.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (get_group), (get_tags),
(gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
(gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
Remove option to mute streams with the current-a/v/t property, we have
this functionality in the flags.
Add signals to notify when the number of A/V/T channels changed.
Add action signals to get tags for the A/V/T streams.
Implement setting the current A/V/T stream.
Rearrange some things to simplify stream selection.
Implement volume.
* gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
(gst_play_sink_get_volume), (gst_play_sink_set_property),
(gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
(activate_vis), (gst_play_sink_reconfigure):
* gst/playback/gstplaysink.h:
Add and implement volume setting methods.
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_finalize),
(gst_stream_selector_set_property),
(gst_stream_selector_get_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_request_new_pad):
* gst/playback/gststreamselector.h:
Add pad properties for tags and status of pads.
Keep tags on pads.
Make active pad selection based on pad object instead of name.
2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
Original commit message from CVS:
* configure.ac:
Revert last change as we now check in gtk-doc.m4 for sed.
2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Find and subst SED when building the docs.
Original commit message from CVS:
* configure.ac:
Find and subst SED when building the docs.
2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
Original commit message from CVS:
2008-02-08 Julien Moutte <julien@fluendo.com>
* tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
(main): Make sure bus signals are reconnected when pressing STOP
and then PLAY again for a parse launch pipeline. Fix a ref leak
on the bus.
* win32/common/config.h: Updated.
2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
Original commit message from CVS:
* configure.ac:
Make DISABLE_DEPRECATED defined *only* during CVS, not during
pre-releases or releases.
2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
Original commit message from CVS:
* configure.ac:
* ext/gio/Makefile.am:
Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
reporting
2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
Original commit message from CVS:
* docs/plugins/Makefile.am:
Add the headers which need scanning for the GIO plugin. The rest of
the docs still need migrating.
2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Add gio in a few more places.
Original commit message from CVS:
* ext/Makefile.am:
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
Add gio in a few more places.
2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Move gio plugin from -bad and mark as experimental.
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* tests/check/Makefile.am:
Move gio plugin from -bad and mark as experimental.
2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Comment out a couple of other things which break the build when
GST_DISABLE_DEPRECATED isn't on but -Werror is.
2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Fix pbutils header.
2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
commit spec file update which includes all the split .pc files
Original commit message from CVS:
commit spec file update which includes all the split .pc files
2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix compiler warning.
2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Clear the addrinfo struct using memset. Fixes #514937.
2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstfdset.h: Remove unused field to same some memory.
Original commit message from CVS:
* gst/tcp/gstfdset.h:
Remove unused field to same some memory.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Mark action signals as such.
2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
Original commit message from CVS:
* ext/theora/theoradec.c: (_theora_granule_frame),
(_inc_granulepos):
Increment granulepos for new-bitstream versions appropriately.
Fixes #514623.
2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek),
(rate_spinbutton_changed_cb), (update_streams), (main):
Remove obsolete stream_time reset after flushing seek, core does that
automatically now.
Improve accuracy of speed spinbutton.
Only do playbin2 stuff when we actually use it.
2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
Original commit message from CVS:
* tests/check/Makefile.am:
Revert previous change of the test environment's GST_PLUGIN_PATH.
The problem is not with the plugins, but with element factories
and only occurs if elements are split out from existing plugins
or if plugins change name (see #512740).
2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
Original commit message from CVS:
* tests/check/Makefile.am:
Fix the tests environment's GST_PLUGIN_PATH: we want the directory
with the core's plugins first and our local build directories last,
since we might be building against an installed core, and that
core's plugin directory may contain older or other versions of
our own -base plugins, but we really do want to test our local
ones (if there are multiple plugins or element factories with the
same name, those inspected last will trump those read in earlier).
Fixes #512740 for the most part.
2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Use gmtime_r if available as gmtime is not MT-safe.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Use gmtime_r if available as gmtime is not MT-safe.
Fixes bug #511810.
2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(get_group), (get_n_pads), (gst_play_bin_get_property),
(pad_added_cb), (no_more_pads_cb), (perform_eos),
(autoplug_select_cb), (deactivate_group):
Remove stream-info, we going for something easier.
Refactor getting the current group.
Implement getting the number of audio/video/text streams.
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_get_property),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Add property for number of pads.
* tests/examples/seek/seek.c: (set_scale), (update_flag),
(vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (msg_async_done),
(msg_state_changed), (main):
Block slider callback when updating the slider position.
Add gui elements for controlling playbin2.
Add callback for async_done that updates position/duration.
2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/plugins/: First round of plugin docs cleansups.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
First round of plugin docs cleansups.
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Regenerate.
* ext/ogg/Makefile.am:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
Add header for oggmux. the c-file needs a doc blob still.
2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
Add gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes #511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (close_stream_cb),
(gst_gio_base_sink_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesrc.c: (close_stream_cb),
(gst_gio_base_src_stop), (gst_gio_base_src_create),
(gst_gio_base_src_set_stream):
Use async variants of the close stream functions to prevent blocking
for a long time there and add some more sanity checks for a correct
stream.
2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.17 ===
2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/config.h:
Release 0.10.17
Original commit message from CVS:
Release 0.10.17
2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Also remove the conditional registration of the signals
that disappeared with the ABI change in 0.10.14
2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Revert patch to gstrtspconnection.c for brown paper bag
release of -base. Re-opens: #511825
2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
Fixes: #513018
2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
Original commit message from CVS:
* win32/common/libgstpbutils.def:
Export the two new _get_type() functions which are needed
by the python bindings.
2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
Initialize the GstRingerBuffer class to get it's debug category
initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
category and otherwise we get some g_critical(). Fixes bug #512334.
2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.16 ===
2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/config.h:
Release 0.10.16
Original commit message from CVS:
Release 0.10.16
2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* common:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
Original commit message from CVS:
Update .po files
2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_extension_data):
Fix typos and wrong extension check. Fixes #511274.
2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
Original commit message from CVS:
* po/sk.po:
Oops - add new sk.po mentioned in the LINGUAS I just committed
2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
po/LINGUAS: Add ca translation to the disted list.
Original commit message from CVS:
* po/LINGUAS:
Add ca translation to the disted list.
* win32/vs6/libgstsdp.dsp:
Convert line endings to CRLF
2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
Original commit message from CVS:
* win32/MANIFEST:
Add win32/vs6/libgstrtsp.dsp to MANIFEST
2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Update for API changes in GIO and require GIO 2.15.2 for this.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Update for API changes in GIO and require GIO 2.15.2 for this.
2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
win32/common/: Add new API declarations
Original commit message from CVS:
* win32/common/libgstsdp.def:
* win32/common/libgstvideo.def:
Add new API declarations
2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraparse.h:
* ext/theora/theoradec.c:
* ext/theora/theoraparse.c:
Take a 2nd stab at handling libtheora granulepos changes in the decoder
and parser by inspecting the bitstream version of the incoming data.
2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Provide one pkg-config file for every gst-plugins-base library.
Original commit message from CVS:
* configure.ac:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
* pkgconfig/gstreamer-audio.pc.in:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
* pkgconfig/gstreamer-cdda.pc.in:
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-fft.pc.in:
* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
* pkgconfig/gstreamer-floatcast.pc.in:
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
* pkgconfig/gstreamer-interfaces.pc.in:
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
* pkgconfig/gstreamer-netbuffer.pc.in:
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-pbutils.pc.in:
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
* pkgconfig/gstreamer-riff.pc.in:
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtp.pc.in:
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp.pc.in:
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
* pkgconfig/gstreamer-sdp.pc.in:
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
* pkgconfig/gstreamer-tag.pc.in:
* pkgconfig/gstreamer-video-uninstalled.pc.in:
* pkgconfig/gstreamer-video.pc.in:
Provide one pkg-config file for every gst-plugins-base library.
This makes linking to those libraries much more intuitive and
provides standard pkg-config behaviour for them. Fixes bug #499697.
2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c:
Fix valgrind error on 4tap scaling method.
2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Include Winsock2.h for VS6 and use a different way initialize
hints structure so it can build with VS6.
* win32/MANIFEST:
* win32/vs6/libgstsdp.dsp:
* win32/common/libgstsdp.def:
Add new files for libgstsdp.
* win32/vs6/grammar.dsp:
Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstdecodebin2.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstvolume.dsp:
Add new dependencies to the link list.
2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
win32/common/: Update/Add generated files in the win32 build directory.
Original commit message from CVS:
2008-01-13 Julien Moutte <julien@fluendo.com>
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
(gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
(gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
(gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
(gst_rtsp_header_field_get_type),
(gst_rtsp_status_code_get_type):
* win32/common/interfaces-enumtypes.c:
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
(gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
(gst_mixer_track_flags_get_type),
(gst_tuner_channel_flags_get_type):
* win32/common/multichannel-enumtypes.c:
(gst_audio_channel_position_get_type):
* win32/common/pbutils-enumtypes.c:
(gst_install_plugins_return_get_type):
* win32/common/pbutils-enumtypes.h: Update/Add generated files
in the win32 build directory.
2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
Add test to see if hstrerror is available or if we need libresolv
(Solaris) for it, then use it in libgstrtsp.
2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/tag/Makefile.am: Fix include path order
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
Fix include path order
2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/pbutils/.gitignore:
Ignore more and make buildbot happy
Original commit message from CVS:
Ignore more and make buildbot happy
2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_context_copy),
(gst_install_plugins_context_get_type):
* gst-libs/gst/pbutils/install-plugins.h:
Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
for bindings.
2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
Original commit message from CVS:
* ext/theora/theoradec.c: (gst_theora_dec_class_init),
(_theora_granule_frame), (_theora_granule_start_time),
(theora_dec_sink_convert), (theora_dec_decode_buffer):
Adapt for post-alpha meaning of granulepos, when we
have a newer version of libtheora.
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
(theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
(theora_enc_is_discontinuous), (theora_enc_chain):
Likewise.
* tests/check/Makefile.am:
Link libtheora into theoraenc test so we can check which version of
libtheora we're testing against.
* tests/check/pipelines/theoraenc.c: (check_libtheora),
(check_buffer_granulepos),
(check_buffer_granulepos_from_starttime), (GST_START_TEST),
(theoraenc_suite):
Adapt tests to check the values that are now defined for theora; make
the tests backwards-adapt the passed values if we're running against an
old libtheora.
Fixes #497964
2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
Original commit message from CVS:
* autogen.sh:
Add -Wno-portability to the automake parameters to stop warnings
about GNU make extensions being used. We require GNU make in almost
every Makefile anyway.
* configure.ac:
Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
at the same time is required for per target flags.
2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gsttagdemux.c: Post an error message if we can't pull as many bytes as we need for the tag. This mak...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
Post an error message if we can't pull as many bytes as we need
for the tag. This makes sure the user gets to see a proper error
message if a file with a partial ID3 tag is fed to decodebin, and
not a 'no ID3 tag demuxer' error, which would be confusing
(see #508138).
2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description strings for ID3, APE, and ICY tags.
2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Make sure we error out correctly if we can't activate one of
the elements we've added. Fixes #508138.
2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
(check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
Use snd_mixer_selem_set_{playback|capture}_volume_all() if
the volume is the same for all channels. This works around
some problem in alsa that leaves us with inconsistent state
for some reason (#486840).
2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
Original commit message from CVS:
Patch by: Jerone Young <jerone at gmail com>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
If there's no mixer track by the name of 'Master' or 'Front',
check if there's one called 'PCM' before trying the generic
fallback logic (fixes #506928, where we pick 'Mic' as master
track for the AD1984 card in a Thinkpad T61/X61 laptop).
2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplay-enum.*: Add enums for configuration flags.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type), (register_gst_play_flags),
(gst_play_flags_get_type):
* gst/playback/gstplay-enum.h:
Add enums for configuration flags.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (gst_play_bin_set_property),
(gst_play_bin_get_property), (no_more_pads_cb),
(autoplug_select_cb), (gst_play_bin_change_state):
Merge mode with flags.
Add more property getters/setters, defaults and docs.
Add properties to get number of audio/video/text streams.
Create sink object in _init so that we can always rely on it being
there.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gen_video_chain), (gen_audio_chain), (gen_vis_chain),
(activate_vis), (gst_play_sink_reconfigure),
(gst_play_sink_set_flags), (gst_play_sink_get_flags),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Use flags to configure the sink pipelines.
Add tee before audio pipeline so that we can use it for visualisations.
Start working on integrating visualisations.
Remove mode, we can do everything with the flags now.
Add method to configue the sink pipeline.
2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
Original commit message from CVS:
* configure.ac:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
* tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
Update to GMemoryInputStream API changes in GLib SVN and require
gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
We can also report the duration for every GSeekable, not only
GFileInputStream and GMemoryInputStream.
2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
(check_buffer_timestamp), (check_buffer_duration):
Turn these functions into macros so we can see right away
where the failure occured.
2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
Original commit message from CVS:
2008-01-05 Julien Moutte <julien@fluendo.com>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
debugging information to understand how X calculates the stride
for XvImages.
2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
Original commit message from CVS:
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_choose_func),
(gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
(volume_setup):
* gst/volume/gstvolume.h:
Use GstAudioFilter as base class for the volume element instead of
plain GstBaseTransform.
2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
Don't set element details for the abstract GstAudioFilter class.
2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
Implement get_unit_size() vmethod of GstBaseTransform.
2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/pbutils.h:
Use glib-enum generator to have a proper enum GType for
GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c:
Reenable theoraenc test, which fails on the buildbot but
not locally.
2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
docs/: Add *-undeclared.txt to fix buildbot.
Original commit message from CVS:
* docs/libs/.cvsignore:
* docs/plugins/.cvsignore:
Add *-undeclared.txt to fix buildbot.
2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
Original commit message from CVS:
* tests/check/Makefile.am:
Second attempt at disabling theoraenc test long enough to
get buildbot to compile -base.
2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c:
Disable theoraenc test long enough to get the buildbot to
compile a recent -base.
2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb):
Make sure we reset the slider value to 0.0 without racing against a
possible g_idle that sets it to something else.
2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
sys/ximage/ximagesink.c: fix typo
Original commit message from CVS:
* sys/ximage/ximagesink.c:
fix typo
2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
Add Location header so that we can start implementing redirects.
See #506025.
2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
gst/subparse/gstssaparse.c: combine if's
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
combine if's
2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
gst/subparse/gstssaparse.c: remove duplicate log message
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
remove duplicate log message
2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
Original commit message from CVS:
* configure.ac:
* ext/gio/gstgio.c:
* ext/gio/gstgio.h:
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
* ext/gio/gstgiobasesrc.h:
* ext/gio/gstgiosink.c: (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.h:
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.h:
* tests/check/pipelines/gio.c:
Update to latest API changes in GLib/GIO and require at least
gio-2.0 2.15.0 for this.
* ext/gio/Makefile.am:
Add GST_PLUGIN_LDFLAGS to LDFLAGS.
2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_chain):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x and don't
abort() in any case but properly report the error.
2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin2.c: Code cleanups.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
(autoplug_select_cb), (activate_group), (deactivate_group),
(setup_next_source), (save_current_group),
(gst_play_bin_change_state):
Code cleanups.
Remove next-uri, we can use the uri property just fine.
Fix some crasher.
Unref uridecodebin when switching.
Fix going to READY.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(gen_video_chain), (gen_text_element), (gen_audio_chain),
(gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_set_flags),
(gst_play_sink_get_flags), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add some locking to make things threadsafe.
* gst/playback/test7.c: (about_to_finish_cb):
Fix test.
2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
(gst_video_scale_transform):
Don't claim to be able to handle/transform caps that can't really
be handled by the currently selected scaling method (here: RGB or
packed YUV with 4-tap method). Also add locking to method property.
* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
(test_basetransform_based):
Some test pipelines for the above (not entirely valgrind clean yet
apparently).
2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
Original commit message from CVS:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add additional RGBA and RGB-24 video formats.
2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_primary_decoder):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
(cddabasesrc_suite):
Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
deprecated in the future (see #498924).
2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gststreamselector.c: Don't leak event.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event):
Don't leak event.
2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Use GST_ROUND_UP_2 macro
2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/.cvsignore: Ignore more.
Original commit message from CVS:
* gst/playback/.cvsignore:
Ignore more.
2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* gst/playback/gstplaybasebin.c: (set_subtitles_visible),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(setup_sinks), (playbin_set_subtitles_visible):
Make switching off of subtitles work. To avoid all kind of
problems with unlinking of the subtitle input, we just keep
the subtitle inputs linked as they are and tell textoverlay
not to render them. Fixes #373011.
Other subtitle switching issues (esp. when there are both
external and in-stream subtitles) remain. They'll be solved
in playbin2.
2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gststreamselector.c: Init the pad segment too.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_init):
Init the pad segment too.
2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
(gst_audio_sink_create_ringbuffer):
Improve debug output.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_delay):
Prevent some functions from doing things and failing when the
ringbuffer is not yet acquired.
2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
Original commit message from CVS:
* gst-libs/gst/interfaces/interfaces.h:
Also remove interfaces.h from CVS as it is not needed anymore.
2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
interfaces.h is not used anymore so remove it from the build
process.
2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
Add a "blink" pattern. Turn on the pain. Apologies. It's useful
for testing vertical refresh synchronization.
2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
Add new GstVideFormat enum and write a bunch of helper functions based around it.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add new GstVideFormat enum and write a bunch of helper functions
based around it.
2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
Makefile.am: Use new common/win32.mak.
Original commit message from CVS:
* Makefile.am:
Use new common/win32.mak.
2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.
2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Yet another fix for broken software that produce files with an empty
blockalign field. Instead of completely failing, make a second attempt
at guessing the width/depth by looking at strf->size.
2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
(gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_spawn_child), (gst_install_plugins_supported):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail),
(gst_missing_encoder_installer_detail_new):
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Turn a few g_assert_not_reached() into g_return_val_if_reached() to
avoid compiler warnings (#503930).
2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
for jpeg video streams.
Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
for the above modification.
2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
(gst_x_overlay_handle_events):
More guards (we don't want klass to end up being NULL).
2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
Original commit message from CVS:
* configure.ac:
* gst/volume/gstvolume.c: (gst_volume_init):
Use new gst_base_transform_set_gap_aware() function as volume
correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
for this.
2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
Original commit message from CVS:
* tests/examples/seek/seek.c: (msg_segment_done), (main):
Don't go to READY on EOS as this avoids testing of seeking and
restarting after EOS, use the stop button when you want to READY.
Don't try to do a flushing seek in segment-done, it does not make
sense to use this for gapless playback and is not needed.
2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Use separate timers for input and output rates.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
(reset_rate_timer), (update_in_rates), (update_out_rates),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_chain), (gst_queue_loop):
Use separate timers for input and output rates.
Pause measuring the output rate when we block for more data.
See #503262.
2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
* gst/speexresample/Makefile.am:
update spec file and add two missing files for disting
Original commit message from CVS:
update spec file and add two missing files for disting
2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_chain):
Pause the timer to measure the input rate when we block because the
queue is filled. See #503262.
2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_free):
Close control sockets. Fixes #503440.
2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
Expose the right pad in the right place with the right element.
2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description for 'private' dts caps (who come up with that name?).
2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
Makefile.am: Add check-exports target and run it with 'make check'.
Original commit message from CVS:
* Makefile.am:
Add check-exports target and run it with 'make check'.
* configure.ac:
Be stricter about what we export in our libraries: change regexp so that
we only export _gst_foo(), but not __gst_foo().
* gst-libs/gst/cdda/base64.h: (rfc822_binary):
* gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
Change internal functions to __gst_foo so they dont' get exported.
* win32/common/libgstaudio.def:
Add missing symbols.
2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
* ChangeLog:
ChangeLog: remove conflict markers
Original commit message from CVS:
ChangeLog: remove conflict markers
2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
Original commit message from CVS:
* ext/gnomevfs/Makefile.am:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
Use gst_tag_freeform_string_to_utf8() here, which also takes
into account any character sets specified by the user via
environment variables.
2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audioconvert/Makefile.am: Also link to libm.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Also link to libm.
2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
No need for floating point operations here. avoids having to link
against the math library too.
2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats),
(format_info_get_desc):
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
(GST_START_TEST):
Add one or two missing formats. Generate ADPCM description
dynamically depending on layout/format.
2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
Original commit message from CVS:
* configure.ac:
Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes #502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplay-enum.*: Add missing files.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type):
* gst/playback/gstplay-enum.h:
Add missing files.
2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
Original commit message from CVS:
* gst/playback/Makefile.am:
Group decodebin2 and uridecodebin into the same plugin so that they
can share the GEnumType.
* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
Add signal to sort factories instead of the more awkward autoplug-select
signal.
Modify autoplug_select so that we can try, skip or expose the
autopluggin of an element on a pad.
* gst/playback/gstfactorylists.c: (compare_ranks),
(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
(element_filter), (gst_factory_list_get_elements),
(gst_factory_list_debug), (gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Simplify the API, allow getting elements based on mask.
* gst/playback/gstplay-marshal.list:
Add some more marshallers.
* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
(autoplug_select_cb), (activate_group):
Add support for managing non-raw sinks by providing a custom element and
sink list to decodebin2.
Try to plug non-raw sinks when decodebin2 using autoplug-select of
decodebin2.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_set_mode), (gst_play_sink_request_pad):
* gst/playback/gstplaysink.h:
Add support for raw and non-raw sinks.
Add support to force sinks selected by playbin2.
Don't plug raw converters for non-raw sinks.
* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
(plugin_init):
Use right accumulators.
Proxy new signal.
2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Use runnning time as the base time instead of the timestamp.
Spotted by Saur on IRC.
2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_read_chain):
If we find a new serial number but it does not contain a BOS page, make
sure we initialize the chain to NULL because else we will try to scan it
and crash. Fixes #500763
2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Refactor some common code to filter factories and check caps compat.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
(get_feature_array), (decoders_filter), (sinks_filter),
(gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
(gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Refactor some common code to filter factories and check caps compat.
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad),
(find_compatibles):
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_finalize),
(autoplug_factories_cb), (activate_group):
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(proxy_drained_signal):
Add some more debug info and use factor filtering code.
2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
configure.ac: Add QuickTime Wrapper plug-in.
Original commit message from CVS:
2007-11-26 Julien Moutte <julien@fluendo.com>
* configure.ac: Add QuickTime Wrapper plug-in.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
build on Mac OS X Leopard. Incorrect printf format arguments.
* sys/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/qtwrapper/audiodecoders.c:
(qtwrapper_audio_decoder_base_init),
(qtwrapper_audio_decoder_class_init),
(qtwrapper_audio_decoder_init),
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
(make_samr_magic_cookie), (open_decoder),
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
(qtwrapper_audio_decoder_chain),
(qtwrapper_audio_decoder_sink_event),
(qtwrapper_audio_decoders_register):
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
(fourcc_to_caps):
* sys/qtwrapper/codecmapping.h:
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
(image_description_for_mp4v), (image_description_from_stsd_buffer),
(image_description_from_codec_data):
* sys/qtwrapper/imagedescription.h:
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
(get_output_info_from_component), (dump_avcc_atom),
(dump_image_description), (dump_codec_decompress_params),
(addSInt32ToDictionary), (dump_cvpixel_buffer),
(DestroyAudioBufferList), (AllocateAudioBufferList):
* sys/qtwrapper/qtutils.h:
* sys/qtwrapper/qtwrapper.c: (plugin_init):
* sys/qtwrapper/qtwrapper.h:
* sys/qtwrapper/videodecoders.c:
(qtwrapper_video_decoder_base_init),
(qtwrapper_video_decoder_class_init),
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
(fill_image_description), (new_image_description), (close_decoder),
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
(decompressCb), (qtwrapper_video_decoder_chain),
(qtwrapper_video_decoder_sink_event),
(qtwrapper_video_decoders_register): Initial import of QuickTime
wrapper jointly developped by Songbird authors (Pioneers of the
Inevitable) and Fluendo.
2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/: Add GAP-flag support.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Add GAP-flag support.
2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
Original commit message from CVS:
2007-11-24 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Increase the range of the
rate selector as I would like to test QOS behavior at higher
forward and reverse playback speed like say 64x.
2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioresample/gstaudioresample.c: Implement latency query.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_query), (audioresample_query_type),
(gst_audioresample_set_property):
Implement latency query.
2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes #498767.
2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/Makefile.am: Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
Add missing file.
2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
Original commit message from CVS:
Patch by: Joe Peterson <lavajoe at gentoo dot org>
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix compilation on FreeBSD (Gentoo). Fixes #498228.
2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add speexresample to the docs and while at that do a make update.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/speexresample/gstspeexresample.h:
Add speexresample to the docs and while at that do a make update.
2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add resample element based on the Speex resampling algorithm.
Original commit message from CVS:
* configure.ac:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
(gst_speex_resample_init), (gst_speex_resample_start),
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
(gst_speex_resample_transform_caps),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_event), (gst_speex_resample_check_discont),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_set_property),
(gst_speex_resample_get_property), (plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
(resampler_basic_direct_single), (resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init), (speex_resampler_init_frac),
(speex_resampler_destroy), (speex_resampler_process_native),
(speex_resampler_process_float), (speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate), (speex_resampler_get_rate),
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
(speex_resampler_set_quality), (speex_resampler_get_quality),
(speex_resampler_set_input_stride),
(speex_resampler_get_input_stride),
(speex_resampler_set_output_stride),
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem), (speex_resampler_strerror):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add resample element based on the Speex resampling algorithm.
2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
Original commit message from CVS:
* tests/check/libs/fft.c: (GST_START_TEST):
Fix scaling to really have dB instead of something else.
2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: There's a nice macro to check
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): There's a nice macro to
check
GTK version, use it.
2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Try to support stable version of GTK.
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Try to support stable version
of GTK.
2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/: Fix the build + little README update.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/test7.c:
Fix the build + little README update.
2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add playbin2 seek pipeline.
Original commit message from CVS:
* tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
Add playbin2 seek pipeline.
2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Add playbin2.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
(eos_cb), (about_to_finish_cb), (main):
Add playbin2.
Added gapless playback example.
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
* gst/playback/gstqueue2.c:
* gst/playback/test.c:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(pad_removed_cb):
* gst/playback/gststreaminfo.h:
Change email.
* gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
(gst_play_bin_class_init), (init_group), (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (gst_play_bin_handle_message),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
(drained_cb), (unlink_group), (activate_group),
(setup_next_source), (gst_play_bin_change_state),
(gst_play_bin2_plugin_init):
Added raw first version of playbin2. Does chained oggs and gapless
playback fine. No support for raw sinks yet. No visualisations or
subtitles yet.
* gst/playback/gstplaysink.c: (gst_play_sink_get_type),
(gst_play_sink_class_init), (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(post_missing_element_message), (free_chain), (add_chain),
(activate_chain), (gen_video_chain), (gen_text_element),
(gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_send_event), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Added Element that abstracts the sinks and their pipelines for playbin2.
2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
(gst_selector_pad_class_init), (gst_selector_pad_init),
(gst_selector_pad_finalize), (gst_selector_pad_reset),
(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_stream_selector_get_type),
(gst_stream_selector_base_init), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_set_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_is_active_sinkpad),
(gst_stream_selector_activate_sinkpad),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Improve streamselector, make it select and unselect the current pad more
intelligently.
Subclass GstPad for the sinkpads of the selector.
Handle segments more correctly.
Fix caps negotiation.
Implement release_pad.
2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_group_check_if_drained), (source_pad_event_probe),
(remove_fakesink):
Add drained signal fired when decodebin finishes decoding the data.
Remove deprecated STATE_DIRTY message.
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
(analyse_source), (proxy_drained_signal), (make_decoder),
(source_new_pad), (value_list_append_structure_list),
(handle_redirect_message), (handle_message):
Proxy the new drained signal.
Handle pad removed from decodebin.
Handle redirect messages by sorting multiple redirections based on the
connection speed.
2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix leaking headers. Fixes #496761.
2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
sys/: Don't leak the PAR on errors. Fixes #496731.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_change_state):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
Don't leak the PAR on errors. Fixes #496731.
2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
(gst_tag_from_id3_user_tag):
Add mapping for audio cd discid tags, so we can extract
them from tags as well (see #347848). Also compare identifiers
in ID3v2 TXXX frames in a case-insensitive way to increase
compatibility when reading tags (discid vs. DiscID vs. DiscId).
2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-plugins-base.doap: Oops, fix the release name.
Original commit message from CVS:
* gst-plugins-base.doap:
Oops, fix the release name.
2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-plugins-base.doap: Add 0.10.15 release
Original commit message from CVS:
* gst-plugins-base.doap:
Add 0.10.15 release
2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.15 ===
2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: releasing 0.10.15, "No need to argue"
Original commit message from CVS:
=== release 0.10.15 ===
2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:
releasing 0.10.15, "No need to argue"
2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
Original commit message from CVS:
Update .po files
2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
win32/vs6/libgstfft.dsp: Convert line endings to DOS.
Original commit message from CVS:
* win32/vs6/libgstfft.dsp:
Convert line endings to DOS.
2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
Original commit message from CVS:
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstfft.dsp:
* win32/MANIFEST:
Add a project file for fft plugin and remove socket
based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Convert line endings back to DOS.
Fixes #496724
2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
win32/vs6/: Convert line endings back to DOS
Original commit message from CVS:
* win32/vs6/libgstinterfaces.dsp:
* win32/vs6/libgstrtsp.dsp:
Convert line endings back to DOS
2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
Original commit message from CVS:
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.h:
Don't include malloc.h which doesn't exist on Mac OSX.
Instead, pull in glib.h and use g_malloc/g_free for
consistency. Fixes: #496548
2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Dont leak ghostpad. Fixes #475451.
2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
Update some more docs and comments.
Original commit message from CVS:
* docs/design/design-decodebin.txt:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Update some more docs and comments.
2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Require GIO >= 0.1.2 and adjust unit test for an API change.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Require GIO >= 0.1.2 and adjust unit test for an API change.
2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
Original commit message from CVS:
* ext/gio/gstgio.h:
Add macro to check if a stream supports seeking.
* ext/gio/Makefile.am:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
* ext/gio/gstgiobasesrc.h:
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
base classes that only require a GInputStream or GOutputStream to
work.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Use the newly created base classes here.
* ext/gio/gstgio.c: (plugin_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
(gst_gio_stream_sink_get_property):
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
(gst_gio_stream_src_get_property):
* ext/gio/gstgiostreamsrc.h:
Implement GstGioStreamSink and GstGioStreamSrc that have a property
to set the GInputStream/GOutputStream that should be used.
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
(gio_testsuite), (main):
Add unit test for giostreamsrc and giostreamsink.
2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
Original commit message from CVS:
* ext/gio/gstgio.c: (plugin_init):
Remove nowadays unnecessary workaround for a crash.
* ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
(gst_gio_sink_start), (gst_gio_sink_stop),
(gst_gio_sink_unlock_stop):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
(gst_gio_src_stop), (gst_gio_src_unlock_stop):
* ext/gio/gstgiosrc.h:
Make the finalize function safer, clean up everything that could stay
around.
Reset the cancellable instead of creating a new one after cancelling
some operation.
Don't store the GFile in the element, it's only necessary for creating
the streams.
2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
Original commit message from CVS:
Patch by: Sebastien Moutte <sebastien moutte net>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
Fix some C99-isms and and a missing function that some versions of
MSVC don't like too much (#494346).
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Update vs6 projects files (#494346).
2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
win32/common/: More missing symbols to export (fixes #493986).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* win32/common/libgstaudio.def:
* win32/common/libgstcdda.def:
* win32/common/libgstinterfaces.def:
* win32/common/libgstnetbuffer.def:
* win32/common/libgstpbutils.def:
* win32/common/libgstrtp.def:
* win32/common/libgstrtsp.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
More missing symbols to export (fixes #493986).
2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/fft/gstffts32.h:
* tests/check/libs/fft.c: (GST_START_TEST):
Remove the magnitude and phase calculation functions as these have
very special use cases and can't even be used for the spectrum
element. Also adjust the docs to mention some properties of the used
FFT implemention, i.e. how the values are scaled. Fixes #492098.
2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
(finish_source):
Avoid crash when there are external subtitles (fixes #491722).
2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
'Could not open resource for writing' is not an acceptable
error message when we can't open the audio device (see #492334),
even less so when we're trying to open it to record something.
2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
win32/common/libgstrtp.def: Add some more missing symbols (#492813).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* win32/common/libgstrtp.def:
Add some more missing symbols (#492813).
2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/audioconvert.c: (verify_convert):
Add check to make sure that the out caps have a channel layout
set on them where they should have one.
2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
Include our own _stdint.h instead of sys/types.h, makes MingW happy
(#492306).
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
Use _pipe directly, GLib doesn't have a pipe() macro any longer
(it disappeared in GLib 2.14.0) (#492306).
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix includes and LIBS for win32/Mingw (#492306).
* tests/examples/dynamic/addstream.c (pause_play_stream):
Use more portable g_usleep() instead of sleep() (#492306).
2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value. Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes #430677.
2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
Original commit message from CVS:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
Original commit message from CVS:
* tests/check/libs/audio.c: (init_value_to_channel_layout),
(test_channel_layout_value_intersect), (audio_suite):
Add simple unit test to make sure GstValue intersection
of channel layouts works the way I think it does.
2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Fix the docs according to what gtk-doc complained about.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/icles/stress-playbin.c: Fix the build.
Original commit message from CVS:
* tests/icles/stress-playbin.c:
Fix the build.
2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Post nice/more useful error message if we don't have a decoder for
the primary type.
2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
Be a bit more useful, unblock the pads after we fired the no-more-pads
signal so that we can use the signal to inspect and connect all pads
without having to keep extra state outside of decodebin.
2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_continue),
(gst_uri_decode_bin_class_init), (no_more_pads_full):
Implement default signal handler so that we return TRUE when nothing is
connected.
2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
* ChangeLog:
Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
Original commit message from CVS:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
Original commit message from CVS:
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
instead.
2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update spec file
Original commit message from CVS:
update spec file
2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
(gst_decode_bin_get_property), (analyze_new_pad):
Move subtitle encoding property to decodebin2 so that it can set the
property value on all elements that it autoplugs and that require it.
Make caps refcounting more consistent in get/set.
* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(make_decoder):
Proxy properties and relevant signals from the internal decodebin.
Make properties MT safe.
2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
Original commit message from CVS:
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
* gst-libs/gst/tag/tags.c:
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
(gst_tag_to_vorbis_comments):
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
just mapping everything I found in the wild) (#414539).
2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
Original commit message from CVS:
Inspired by patch of: René Stadler <mail at renestadler dot de>
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
(find_compatibles):
* gst/playback/gstplay-marshal.list:
Remove the autoplug-sort signal and replace it with a binding friendly
autoplug-select signal.
Add an autoplug-factories signal that can be used to generate a list of
factories to try to autoplug.
Add the GstPad to the autoplugging signal args as it might be needed to
make a good factory selection.
Fix up the marshallers for this. Fixes #407282.
2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't abort with an assertion if we receive a seek event with
a start type of NONE (see launchpad bug #155878).
2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state), (gst_ximagesink_reset):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
(gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_change_state), (gst_xvimagesink_reset):
Make sure that before we clean up the X resources, we shutdown and join
the event thread.
Also make sure the event thread does not shut down immediatly after
startup because the running variable is not yet correctly set.
Fixes #378770.
2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad), (type_found):
Make the window for a race in typefind and shutting down smaller until
we figure out the right locking here. Avoids #485753 usually.
* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
Remove unneeded lock causing a race in typefind and shutting down.
Fixes #485753.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Also remove sinks when going to NULL because we might not complete the
state change to PAUSED, causing the PAUSED->READY state change not to
happen.
2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/icles/: Does what it says on the tin.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/stress-playbin.c:
Does what it says on the tin.
2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
Fix queue negotiation. See #486758.
2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Actual code change to go along with:
Original commit message from CVS:
Actual code change to go along with:
2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
(gst_xvimagesink_xwindow_new),
(gst_xvimagesink_update_colorbalance),
(gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
Fix handling of some of the X atoms. If the last parameter is True,
XInternAtom won't create the atom if it doesn't exist, and therefore
might return None. This causes X errors on Xv implementations that
don't provide the colour balance attributes.
2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
Remove stray character from the changelog.
Original commit message from CVS:
Remove stray character from the changelog.
2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
I'm too lazy to comment this
Original commit message from CVS:
*** empty log message ***
2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
Extract vorbis comment LICENSE tags correctly.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Extract vorbis comment LICENSE tags correctly.
2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
Original commit message from CVS:
Patch by: Jason Kivlighn <jkivlighn gmail com>
* gst-libs/gst/tag/gstid3tag.c:
* tests/check/libs/tag.c:
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't error out when a buggy downstream element doesn't
handle the newsegment event we send properly (especially
not without posting a meaningful error message on the
bus). See bug #471370 and launchpad bug #136264.
2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Fix silly search-replace oversight.
2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix caps memleak. Fixes #484989.
2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
Fix debug output.
2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
to avoid compiler warnings
2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (type_found),
(gst_decode_bin_change_state):
Don't disconnect the have_type signal because we never reconnect it
later on. Instead keep a variable to see if we already detected a type.
2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(type_found):
Unlink the signal handler when we found the type, we're not going to do
anything sensible with more type_found signals anyway.
2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Use GIO function to get a list of supported URI schemes instead of
hard coding something.
2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't leak caps.
2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.c:
* gst-libs/gst/tag/gsttagdemux.h:
API: add GstTagDemux base class for simple tag demuxers.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add GstTagDemux to docs.
2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_payload_subbuffer):
Fix bug introduced with last commit which inverted the logic and
caused all buffers to be dropped. Fixes #483620.
Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Replace g_return_if_val (as it could be disabled), with regular return
and warning.
2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
Original commit message from CVS:
* tests/check/pipelines/simple-launch-lines.c:
Print message name and not just number.
2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/snapshot/snapshot.c: Fix the build.
Original commit message from CVS:
* tests/examples/snapshot/snapshot.c:
Fix the build.
2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/gstgiosink.c: Update to API changes in GIO.
Original commit message from CVS:
* ext/gio/gstgiosink.c: (gst_gio_sink_start):
Update to API changes in GIO.
2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.h:
Add RFC 3556 bandwidth modifiers.
2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
Update documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtppayloads.c:
Update documentation.
2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/: Added new file and header to deal with payload info.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
(gst_rtp_payload_info_for_name):
* gst-libs/gst/rtp/gstrtppayloads.h:
Added new file and header to deal with payload info.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Payload specific stuff is move to new headers.
Implement _default_clock rate using the new payload function.
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
(gst_sdp_parse_line):
* gst-libs/gst/sdp/gstsdpmessage.h:
Add some more comments.
2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(sdp_check_header), (sdp_type_find), (plugin_init):
Add typefind function for application/sdp.
Remove some old dirac typefind code that was ifdeffed out.
2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/common/libgstaudio.def: Add new exported functions.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add new exported functions.
* win32/vs6/grammar.dsp:
Add autogeneration and copy of some autegenerated files from win32/common
for rtsp library.
* win32/vs6/libgstaudioconvert.dsp:
Add gstaudioquantize.c to the build.
* win32/vs6/libgstinterfaces.dsp:
Add videoorientation.c to the build.
* win32/vs6/libgstriff.dsp:
Add libgsttag to the link libraries list.
* win32/vs6/libgstvolume.dsp:
Add liboil to the link.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstrtsp.dsp:
* win32/common/libgstrtsp.def:
Add files to build libgstrtsp library.
2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
Original commit message from CVS:
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_set_property), (gst_gio_sink_render):
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_set_property):
Some minor cleanup and allow setting the location only when the
element is not playing or paused.
2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
Original commit message from CVS:
* tests/examples/snapshot/snapshot.c: (main):
Print error when pipeline failed to construct.
2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Add mappings for the new GST_TAG_COMPOSER for vorbis comments
and ID3v2 tags.
2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Don't include config.h in an installed public header, this
might break compilation of applications that don't have such
a header and doesn't necessarily do what it's supposed to do
anyway (ie. check for the lrint/lrintf defines) (#442065).
Add docs for the various macros and document how this header
has to be used (link against libm, etc.); add a few FIXMEs;
include math.h for non-c99 code path. Based on patch by
Jan Schmidt.
2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
Original commit message from CVS:
* configure.ac:
Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
of duplicating these macros in configure.ac.
2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/: Updated translations to 0.10.14
Original commit message from CVS:
* po/hu.po:
* po/sv.po:
* po/uk.po:
Updated translations to 0.10.14
2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* po/LINGUAS:
add languages
Original commit message from CVS:
add languages
2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/pl.po: Added Polish translation.
Original commit message from CVS:
translated by: Jakub Bogusz <qboosh@pld-linux.org>
* po/pl.po:
Added Polish translation.
2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/fi.po: Added Finnish translation.
Original commit message from CVS:
translated by: Ilkka Tuohela <hile@iki.fi>
* po/fi.po:
Added Finnish translation.
2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/es.po: Added Spanish translation.
Original commit message from CVS:
translated by: Jorge González González <aloriel@gmail.com>
* po/es.po:
Added Spanish translation.
2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/da.po: Added Danish translation.
Original commit message from CVS:
translated by: Mogens Jaeger <mogens@jaeger.tf>
* po/da.po:
Added Danish translation.
2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/zh_CN.po: Added Chinese (simplified) translation.
Original commit message from CVS:
translated by: Funda Wang <fundawang@linux.net.cn>
* po/zh_CN.po:
Added Chinese (simplified) translation.
2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/bg.po: Added Bulgarian translation.
Original commit message from CVS:
translated by: Alexander Shopov <ash@contact.bg>
* po/bg.po:
Added Bulgarian translation.
2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
Update hierarchy.
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.h:
Mark private fields of the instance structs private.
2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dfbvideosink.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
Add the GIO plugin to the docs and do a make update
while doing that.
* ext/gio/gstgiosrc.c: (gst_gio_src_start):
Fix a small memleak.
2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* configure.ac:
* ext/Makefile.am:
* ext/gio/Makefile.am:
* ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
(gst_gio_get_supported_protocols),
(gst_gio_uri_handler_get_type_sink),
(gst_gio_uri_handler_get_type_src),
(gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
(gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
(gst_gio_uri_handler_do_init), (plugin_init):
* ext/gio/gstgio.h:
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_set_property),
(gst_gio_sink_get_property), (gst_gio_sink_start),
(gst_gio_sink_stop), (gst_gio_sink_unlock),
(gst_gio_sink_unlock_stop), (gst_gio_sink_event),
(gst_gio_sink_render), (gst_gio_sink_query):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_set_property),
(gst_gio_src_get_property), (gst_gio_src_start),
(gst_gio_src_stop), (gst_gio_src_get_size),
(gst_gio_src_is_seekable), (gst_gio_src_unlock),
(gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
(gst_gio_src_create):
* ext/gio/gstgiosrc.h:
Add a GIO/GVFS plugin with source and sink elements. This will
only be enabled when --enable-experimental is given to configure
for now as the GIO API is not stable yet. Fixes #476916.
2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_push_one):
Fix compilation wrt printf arguments.
2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
Original commit message from CVS:
* examples/app/appsrc_ex.c: (main):
Fix compilation after changing the name of a method.
2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
Add simple snapshot example program using appsink.
Original commit message from CVS:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/snapshot/.cvsignore:
* tests/examples/snapshot/Makefile.am:
* tests/examples/snapshot/snapshot.c: (main):
Add simple snapshot example program using appsink.
2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_set_property), (gst_app_sink_get_property),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_event), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add properties, signals and actions to access the element even without
linking to the library.
Fix some method names and signatures.
2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/generic/states.c: Improved state change unit test.
Original commit message from CVS:
* tests/check/generic/states.c:
Improved state change unit test.
2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Ignore registries in any format.
Original commit message from CVS:
* docs/plugins/.cvsignore:
* tests/check/.cvsignore:
Ignore registries in any format.
2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp):
Only copy timestamp on outgoing packets if the depayloader did not set
one.
Also copy duration on outgoing packets.
2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix compilation because of missing %d in printf.
When fixating caps, fixate what we can and throw away all remaining
unfixed caps, subclasses should do something smart if they need to.
2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Improve debug logs a bit and be more verbose if things go wrong.
2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Fix a bunch of compile warnings shown with Forte.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Also fix #476514 for queue2.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_flush), (gst_queue_locked_enqueue),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_push_one), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_src_activate_pull):
Also fix #476514 for queue2.
2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Remove code to deal with RTP to GST time conversion, we now just copy
the GST timestamp we receive to the outgoing buffers.
Handle segment and flushes correctly.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
When we have no valid input timestamp, use the previous rtp timestamp on
the outgoing RTP packet instead of the RTP base time.
2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
Original commit message from CVS:
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
Change alsa alloca's to malloc to fix warnings on gcc-4.2.
2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps), (gst_basertppayload_push):
Add some debug info when negotiating caps.
2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
A buffer with an empty payload is also a valid buffer.
2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
(gst_basertppayload_change_state):
Make sure we start our RTP timestamp from the random base RTP
timestamp even if the buffer timestamp starts from some random value.
2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
Original commit message from CVS:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/dynamic/.cvsignore:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/addstream.c: (create_stream),
(pause_play_stream), (message_received), (eos_message_received),
(perform_step), (main):
Add simple exmple app to demonstrate starting and pausing live and
non-live bins in a PLAYING pipeline.
2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
Original commit message from CVS:
2007-09-14 Julien MOUTTE <julien@moutte.net>
* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
typefind for QCP files (RFC #3625)
2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init):
Disable pull mode scheduling, we're not ready for it yet and it subtly
breaks a lot of things.
2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
Original commit message from CVS:
* tests/check/elements/libvisual.c:
Test all libvisual plugins, not just the first one; this reproduces
bug #450336 quite easily. Looks like a problem with the 'jess'
visualisation.
2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/libvisual.c:
Add basic libvisual test case in an attempt to reproduce bug #450336.
Doesn't reproduce that bug, but some other crasher instead (invalid
free), at least with make elements/libvisual.forever and the bumscope
plugin on x86-64/gutsy. Leaving test disabled for now.
2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
gst/: Printf format fixes (#476128).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/app/gstappsink.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvparse.c:
* gst/interleave/deinterleave.c:
* gst/switch/gstswitch.c:
Printf format fixes (#476128).
2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
(read_body), (gst_rtsp_connection_receive):
Make sure we can not cancel in the middle of receiving a message.
Fixes #475731.
2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
Original commit message from CVS:
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c:
Increase upper limit for audio queue a bit; fixes preroll problem
with playbin and decodebin2 when playing a quicktime trailer with
multichannel audio via http (#464666).
2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Allow othe clocks than the internal clock to be used for the pipeline.
Add property to disable clock provide.
API: GstBaseAudioSrc::provide-clock
2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Don't leak request pads. Fixes #475395.
2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_class_init):
Correctly chain up finalize with the parent class to prevent
memory leaks. Fixes #474880.
2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
Original commit message from CVS:
Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update spec file to include latest RTSP libraries and headers and more
Original commit message from CVS:
update spec file to include latest RTSP libraries and headers and more
2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
win32/: Add rtsp enumtypes (#474384) and update others.
Original commit message from CVS:
* win32/MANIFEST:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/gstrtsp-enumtypes.h:
* win32/common/interfaces-enumtypes.c:
* win32/common/interfaces-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
Add rtsp enumtypes (#474384) and update others.
2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
configure.ac: Fix configure check for HAVE_LIBXML_HTML.
Original commit message from CVS:
* configure.ac:
Fix configure check for HAVE_LIBXML_HTML.
2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
Original commit message from CVS:
* tests/check/libs/.cvsignore:
Ignore more, in case the build bots work again one day.
2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
Add libgstfft, a FFT library based on Kiss FFT which is
Original commit message from CVS:
Reviewed by: Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
* gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
* gst-libs/gst/fft/gstfft.h:
* gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
(gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
(gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
(gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
(gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
(gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
(gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
(gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
(gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
* gst-libs/gst/fft/gstffts32.h:
* gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
(kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
(kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
(kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
(kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
* gst-libs/gst/fft/kiss_fft_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
(kiss_fftr_f32), (kiss_fftri_f32):
* gst-libs/gst/fft/kiss_fftr_f32.h:
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
(kiss_fftr_f64), (kiss_fftri_f64):
* gst-libs/gst/fft/kiss_fftr_f64.h:
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
(kiss_fftr_s16), (kiss_fftri_s16):
* gst-libs/gst/fft/kiss_fftr_s16.h:
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
(kiss_fftr_s32), (kiss_fftri_s32):
* gst-libs/gst/fft/kiss_fftr_s32.h:
* gst-libs/gst/fft/kiss_version:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Add libgstfft, a FFT library based on Kiss FFT which is
BSD licensed. Supported sample formats are int16, int32,
float and double. For those formats a real FFT and IFFT
can be done, different windowing functions can be applied
and functions for extracting the magnitude and phase exist.
Fixes #468619.
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Integrate libgstfft into the docs.
* tests/check/Makefile.am:
* tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
Add unit tests for libgstfft, currently only testing the FFT.
Unit tests for IFFT will follow soon.
2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(is_multicast_address), (gst_sdp_message_as_text),
(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
(gst_sdp_media_init), (gst_sdp_media_uninit),
(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Separate INIT_ARRAY() and related macros into two versions, one for
structures and one for pointers (e.g., INIT_ARRAY() and
INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
lists of emails and phone numbers.
Add missing const as appropriate.
Change all gint to guint since they all actually represent unsigned
values.
Do not use time as a variable name as it shadows the global time().
Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
Actually implement gst_sdp_message_add_time().
Make gst_sdp_message_add_time() take repeat times as an argument.
Store repeat times in GstSDPTime as a GArray rather than as gchar**.
Corrected the definition of gst_sdp_media_get_bandwidth() (was
misspelled as badwidth).
gst-indented and a little clean up. Fixes #471067.
2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Need to include stdlib.h for abs() here too.
2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gststreaminfo.c: Fix build.
Original commit message from CVS:
* gst/playback/gststreaminfo.c:
Fix build.
2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
Original commit message from CVS:
* gst/playback/gststreaminfo.c:
Clean up some half-disabled code and comment.
2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
Original commit message from CVS:
2007-09-03 Johan Dahlin <jdahlin@async.com.br>
* gst/typefind/gsttypefindfunctions.c (plugin_init):
Add an audio/x-nsf typefind function for the nsfdec element.
2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Included "myth://" on stream_uris list for enable buffering to mythtv files
2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
Fix parsing of RB blocks.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix parsing of RB blocks.
Fix docs.
Added helper functions to convert to/from UNIX and NTP time.
API: gst_rtcp_ntp_to_unix()
API: gst_rtcp_unix_to_ntp()
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_get_header_len),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix some more docs.
Implement handling of packets with extensions.
Fix padding check in _validate().
Added function to get extension data.
API: gst_rtp_buffer_get_header_len()
API: gst_rtp_buffer_get_extension_data()
2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_set_gst_timestamp):
Add some more docs for the queue-delay property and fix a typo in a
comment.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Fix typo.
2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
When skew slaving, try to hover around the middle of a segment so that
we at most drift by half a segment.
If we are aligning in the oposite direction of the clock skew, we don't
have to resync.
2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Be less silly with the segment start, just apply the clock-base to the
timestamp.
2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Deprecate the queue handling thread thing and remove the code.
Use new method to calculate the extended timestamp.
2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_copy_entry):
Use g_strndup which does exactly what we want.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add helper function to compare seqnums.
Add helper function to calculate extended timestamps.
API: gst_rtp_buffer_compare_seqnum()
API: gst_rtp_buffer_ext_timestamp()
2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_get_entry),
(gst_rtcp_packet_sdes_copy_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix and document SDES item data function.
Add new function that makes a proper copy of SDES item data.
API: gst_rtcp_packet_sdes_copy_entry()
2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
The tcp and subparse plugins are under gst, but not totaly free of
dependencies. Handle selection inconfigure.ac, so that they show up
on the final list of what is build and what is not. Maybe they should
better be moved to ext.
2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
Check if libxml provides HTML parser which subparse needs.
Original commit message from CVS:
Patch by: Daniel Díaz <yosoy@danieldiaz.org>
* configure.ac:
* gst/Makefile.am:
Check if libxml provides HTML parser which subparse needs.
Fixes #451970.
2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
Original commit message from CVS:
* ext/alsa/gstalsa.c:
Fix typo and compilation on big endian systems.
2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
Convert SSA newline codes into actual newline characters (#470766).
2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
API: also add gst_install_plugins_supported() while we're at it (see #470456).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* tests/check/libs/pbutils.c:
API: also add gst_install_plugins_supported() while we're at it
(see #470456).
2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.h:
* tests/check/libs/pbutils.c:
API: add gst_missing_*_installer_detail_new() convenience API so
that applications that know exactly what they're missing can request
installer detail strings for those items directly instead of having
to first create a dummy missing-plugin message and then get the
installer detail string from that. Fixes #470456.
2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
We need to set up delayed-linking whenever the caps are non-fixed,
not just when there are multiple types - use gst_pad_is_fixed()
to test.
2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail):
Add missing separator in PID fallback case.
2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
Original commit message from CVS:
Based on a patch by: Davyd <davyd at madeley dot id dot au>
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_int32),
(volume_process_int32_clamp), (volume_process_int24),
(volume_process_int24_clamp), (volume_process_int16),
(volume_process_int16_clamp), (volume_process_int8),
(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
* gst/volume/gstvolume.h:
Add support for int32, int24 and int8 to the volume element.
Fixes #445529.
2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/examples/Makefile.am: Fix even more.
Original commit message from CVS:
* tests/examples/Makefile.am:
Fix even more.
2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
* gst-libs/gst/Makefile.am:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
* sys/v4l/v4lsrc_calls.c:
* tests/examples/Makefile.am:
* win32/common/config.h:
Revert unwanted commit. many thanks to moap. I want a fix for
https://thomas.apestaart.org/moap/trac/ticket/239
2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
* ChangeLog:
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst/typefind/gsttypefindfunctions.c:
* gst/volume/gstvolume.c:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
* sys/v4l/v4lsrc_calls.c:
* tests/examples/Makefile.am:
* win32/common/config.h:
Original commit message from CVS: reviewed by: <delete if not using a buddy> patch by: <delete if not someone else's patch> * configure.ac: * docs/libs/Makefile.am: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * ext/gnomevfs/gstgnomevfssrc.c: * ext/gnomevfs/gstgnomevfssrc.h: * gst-libs/gst/Makefile.am: * gst-libs/gst/audio/gstaudiofilter.h: * gst/typefind/gsttypefindfunctions.c: * gst/volume/gstvolume.c: * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: * pkgconfig/gstreamer-plugins-base.pc.in: * sys/v4l/v4lsrc_calls.c: * tests/examples/Makefile.am: * win32/common/config.h:
2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/audio.c: Clarify the docs a little.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Clarify the docs a little.
2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
Original commit message from CVS:
* gst/volume/gstvolume.c:
Enable liboil for float and add more details about problems with
int16.
2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
When calculating the first timestamp of the buffers, don't go below 0
and clip the samples because the offset was on the eos page.
Fixes #466717.
2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
(gst_ogg_demux_collect_chain_info):
Also submit the eos page when trying to find the first timestamp.
See #466717.
2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes #467667.
2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Small cleanups.
On shutdown, don't read the control socket yet.
Set timeout value correctly in all cases.
Add function to check if the server accepts reads or writes.
API: gst_rtsp_connection_poll()
* gst-libs/gst/rtsp/gstrtspdefs.h:
Fix compilation with -pedantic.
Add enum for _poll.
2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Override the preroll vmethod instead of overriding the render method
twice.
2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_getcaps):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add getcaps vfunc to basertppayload. See #465146.
2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
Only post buffering messages when we are a stream.
2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/: Small docs fix and addition.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.c:
Small docs fix and addition.
2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.c: Don't use new API.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
Don't use new API.
2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/app/gstappsink.*: Make love to appsink.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
(gst_app_sink_class_init), (gst_app_sink_dispose),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_get_caps),
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Make love to appsink.
Make it support pulling of the preroll buffer.
Add docs and debug statements.
Fix some races wrt to EOS handling and stopping.
Implement getcaps.
Implement FLUSHING.
API: gst_app_sink_pull_preroll()
2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/icles/: Add a dumb little test for textoverlay alignments.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/test-textoverlay.c:
Add a dumb little test for textoverlay alignments.
2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
Original commit message from CVS:
Patch by: Dan Williams <dcbw redhat com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
API: add "line-alignment" property (#459334). Add gtk-doc blurb for
"silent" property so there's a Since tag in the API reference.
2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
fix ... by: lines
Original commit message from CVS:
fix ... by: lines
2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
* gst-libs/gst/rtp/gstbasertppayload.h:
Improve caps negotiation so that downstream elements can confiure
certain RTP properties by fixing them on the caps. See #465146.
Add docs.
2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Mark as deprecated some macros which were presumably meant to be
private API and accidentally exposed in the public header file.
Also actually _init() lock (only works at the moment because the
struct is zeroed out when created and the initial values in the
mutex struct are zeroes too). (#459585)
2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/Makefile.am: Remove cruft and do some cleanups.
Original commit message from CVS:
* docs/libs/Makefile.am:
Remove cruft and do some cleanups.
* docs/libs/gst-plugins-base-libs-docs.sgml:
Prepare for comming gtkdoc features (rebase against online docs).
2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
gst/audiorate/gstaudiorate.c: Debug output fixes.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/: Printf format fixes (#465028).
Original commit message from CVS:
* gst/playback/gstqueue2.c:
* gst/videorate/gstvideorate.c:
Printf format fixes (#465028).
2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
gst/playback/gstplaybasebin.c: Fixes: #465015
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_remove_probe), (queue_threshold_reached):
Patch by: Josep Torra Valles <josep@fluendo.com>
Fixes: #465015
Make sure we remove the check_queues buffer probe from the
correct queue to avoid racily going back to "buffering 99%" when
buffering is actually complete.
Also, fix the spelling of Josep's surname in the ChangeLog.
2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/ogg/gstoggmux.c: Do not leak oggmux instance.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Do not leak oggmux instance.
* ext/vorbis/vorbisenc.c:
Also log values.
2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/: Updated translations.
Original commit message from CVS:
* po/hu.po:
* po/it.po:
* po/nl.po:
* po/uk.po:
* po/vi.po:
Updated translations.
2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
Original commit message from CVS:
patch by: Yang Hong <hongyang@redflag-linux.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
Add 'silent' property to GstTimeOverlay. Fixes #462979
2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
Add connection-speed property. Fixes #464690.
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* docs/plugins/gst-plugins-base-plugins.args:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (gen_source_element):
Add connection-speed property. Fixes #464690.
2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
Fix compilation on windows. Fixes #464320.
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
Fix compilation on windows. Fixes #464320.
2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (queue_threshold_reached),
(gen_source_element), (setup_substreams),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(gst_play_bin_handle_redirect_message):
Move connection-speed property from playbin to playbasebin so that we
can also configure it in source elements that have the connection-speed
property. Fixes #464028.
Add some debug info here and there.
2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Properly respond to conversion queries. Fixes #464079.
2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add float/double and int32 support to audiotestsrc. Fixes #460422.
Also set the default volume to the default value specified in the
GParamSpec.
2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/audioconvert/gstaudioquantize.c:
Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
Add rdt manager for rdt transport.
Fix parsing of RDT transport.
2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.14 ===
2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/config.h:
Release 0.10.14
Original commit message from CVS:
Release 0.10.14
2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/de.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
Original commit message from CVS:
* tests/check/libs/audio.c: (GST_START_TEST):
Fix the test to reflect the behaviour of gst_audio_clip_buffer.
2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
When clipping a buffer with no timestamp, assume it is
within the segment without warnings.
Fixes: #460978
2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
Fire the signal on the object, not the interface.
2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ber. Don't include the full path, idiot.
2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ignore generated files.
2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
* gst-libs/gst/interfaces/rtspextension.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp.h:
* gst-libs/gst/rtsp/gstrtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/rtsp/gstrtspextension.h:
* gst-libs/gst/rtsp/rtsp-marshal.list:
Move the rtspextension.h interface into gstrtspextension.h
as part of libgstrtsp instead of libgstinterfaces, because it's
only for use within plugins, not applications.
Add stuff to do the enum & marshal generation needed in libgstrtsp now.
Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
is abstract.
2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/interfaces/: Fix marshaller for the send signal.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_iface_init),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Fix marshaller for the send signal.
Add URL to stream selection interface method.
2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
Pull in our dependencies from -base before those from outside.
2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
API: gst_rtsp_base64_decode_ip()
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
* gst-libs/gst/rtsp/gstrtspbase64.h:
API: gst_rtsp_base64_decode_ip()
Added function to decode Base64 in-place.
2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/libs/.cvsignore: Ignore the mixer test binary.
Original commit message from CVS:
* tests/check/libs/.cvsignore:
Ignore the mixer test binary.
2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
Gratuitous comment change to trigger a rebuild on the buildbots.
2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_get_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val):
* gst-libs/gst/sdp/gstsdpmessage.h:
Constify args where we can.
2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Move interface for RTSP extensions from -good to here.
Added helper methods to invoke interface methods.
2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
Fix some more RTSP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
(gst_rtsp_message_init_response),
(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_get_body), (dump_key_value):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
Fix some more RTSP docs.
Add some missing methods for dealing with messages.
2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
Added beginnings of RTSP documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (add_auth_header),
(gst_rtsp_connection_write), (gst_rtsp_connection_send),
(read_body), (gst_rtsp_connection_receive),
(gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtspurl.h:
Added beginnings of RTSP documentation.
2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
Document the SDP library.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_media_new),
(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_media_get_media), (gst_sdp_media_set_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_add_format),
(gst_sdp_media_get_information), (gst_sdp_media_set_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_set_key), (gst_sdp_media_get_key),
(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
(print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Document the SDP library.
Add some of the missing SDPMedia methods.
2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
Move SDP and RTSP from helper objects in -good to a reusable library.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
(parse_response_status), (parse_request_line), (parse_line),
(gst_rtsp_connection_read), (read_body),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
(gst_rtsp_strresult), (gst_rtsp_method_as_text),
(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
(gst_rtsp_find_method):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_new), (gst_rtsp_message_init),
(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
(gst_rtsp_message_free), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
(gst_rtsp_message_dump):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse), (gst_rtsp_range_free):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
(range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
(gst_rtsp_transport_free):
* gst-libs/gst/rtsp/gstrtsptransport.h:
* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
(gst_rtsp_url_free), (gst_rtsp_url_set_port),
(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
* gst-libs/gst/rtsp/gstrtspurl.h:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
(gst_sdp_attribute_init), (gst_sdp_message_new),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
(gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
(read_string), (read_string_del), (gst_sdp_parse_line),
(gst_sdp_message_parse_buffer), (print_media),
(gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
Move SDP and RTSP from helper objects in -good to a reusable library.
Use a proper gst_ namespace.
2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
(vorbis_dec_flush_decode):
Use the new buffer clipping function from gstaudio here.
2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Cleanup the docs.
2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
Original commit message from CVS:
Patch by: Dan Williams <dcbw at redhat dot com>
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_get_streaminfo_value_array):
Don't return NULL when querying the stream info value array but instead
return an empty array. Fixes #459204.
2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gsturidecodebin.c: Init debug category before using it.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
Init debug category before using it.
2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Add padding vars in place of the signal pointers
when building with DISABLE_DEPRECATED so that the
interface structure doesn't change size.
2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
Fixes: #152864
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixertrack.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.c:
* gst-libs/gst/interfaces/mixertrack.h:
* tests/check/Makefile.am:
* tests/check/libs/mixer.c:
Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
Fixes: #152864
Add support for notifying mixer changes on the message bus, and
implement it in alsamixer.
API: gst_mixer_get_mixer_flags
API: gst_mixer_message_parse_mute_toggled
API: gst_mixer_message_parse_record_toggled
API: gst_mixer_message_parse_volume_changed
API: gst_mixer_message_parse_option_changed
API: GstMixerMessageType
API: GstMixerFlags
2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
xcontext->im_format is only for testing XShm support (as the header
file comments document). Use xvimage->im_format for everything else.
Avoids spurious warnings on buffer allocation before setcaps.
2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/: We should use $(LIBM).
Original commit message from CVS:
* tests/examples/volume/Makefile.am:
* tests/icles/Makefile.am:
We should use $(LIBM).
2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/icles/Makefile.am: This needs -lm.
Original commit message from CVS:
* tests/icles/Makefile.am:
This needs -lm.
2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Add stdlib include (free, atoi, exit).
Original commit message from CVS:
* examples/app/appsrc_ex.c:
* examples/switch/switcher.c:
* ext/neon/gstneonhttpsrc.c:
* ext/timidity/gstwildmidi.c:
* ext/x264/gstx264enc.c:
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/spectrum/demo-audiotest.c:
* gst/spectrum/demo-osssrc.c:
* sys/dvb/gstdvbsrc.c:
Add stdlib include (free, atoi, exit).
2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property):
Don't break ABI, restore previous ranges. Keep the default random
selection of timestamp and seqnum offset but as soon as the app sets a
specific value, use that one.
2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess dot net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
Add option to turn off double-buffering for debugging purposes.
Fixes #437169.
2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
Original commit message from CVS:
Patch by: Jorn Baayen <jorn at openedhand dot com>
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
(gst_ximagesink_set_property), (gst_ximagesink_get_property),
(gst_ximagesink_init), (gst_ximagesink_class_init):
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
add 'handle-expose' property. Useful for video widgets which may want to
be in control of Expose behaviour. Fixes #380625
2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Fix ranges of rtp payloader properties so that the full range can be
used in addition to -1 (random).
Fix wrong seqnum reporting in caps.
Fixes #420326.
2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videorate/gstvideorate.c: Use boilerplate.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_init),
(gst_video_rate_query):
Use boilerplate.
Add latency query, might not be perfect yet but already works a lot
better. Fixes #442557.
2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_setcaps):
* sys/xvimage/xvimagesink.h:
After a caps change, redraw our borders to avoid garbage left there
when the image format changes to a smaller size, like 16:9 -> 4:3
Also, hold the flow_lock a bit longer in the set_caps while we're
fiddling with the xcontext.
2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* tests/Makefile.am:
Remove bogus check for libcheck, since we check for
gstreamer-check and it pulls in the required info from there, and we
weren't actually _using_ the information for libcheck ourselves
anyway.
2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
Fix the r_mask test for RGBA32 on little-endian.
Fix a stupid typo that would have obviously broken
compilation on big-endian, if anyone was testing.
2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
(paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add alpha to the color struct.
Use a default alpha value of 255 instead of 128.
2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (no_more_pads_full),
(setup_source):
Clear the dynamic pads counter when starting a new uri. This makes
reusing playbin work again.
Fixes #454264.
2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
configure.ac: Use pkg-config to locate check.
Original commit message from CVS:
* configure.ac:
Use pkg-config to locate check.
2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
Fix 'make check' build against core CVS.
Original commit message from CVS:
* configure.ac:
* tests/check/elements/volume.c: (GST_START_TEST):
Fix 'make check' build against core CVS.
2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/: Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Fix location of includes in the docs.
2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
Original commit message from CVS:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
(img_get_alpha_info):
Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
of the existing BGRA32 and RGBA32 formats with the alpha at the other
end of the word. Partially fixes #451908
2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/: Simplify --extra-dir as gtkdoc scans recursively.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Simplify --extra-dir as gtkdoc scans recursively.
2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
(gst_adder_request_new_pad):
Make getcaps more robust by not using the proxycaps function. This makes
sure that we don't end up recursively calling getcaps upstream.
See #316248.
2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Include math.h to fix compilation.
2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
format, as produced by some dc1394 cameras like the iSight.
See http://www.fourcc.org/yuv.php#IYU1
2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
Use other metrics as well when estimating the buffer level.
2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Small debug improvement.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
Small debug improvement.
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
(plugin_init):
Tweak the rate estimation period.
When calculating the buffer filledness in rate estimation mode, don't
mix it with other metrics.
2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
When creating the groups, allow for a 5 second, unlimited buffers
preroll phase after which we expose the group.
When the group is exposed, use a small number of buffers up to a 2
second limit. Also disconnect the overrun signal from multiqueue when we
exposed the group because it is not needed anymore.
2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
Add description for Windows Media RTP caps.
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove RTP fields that don't define the format from caps.
2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Skip empty buffers, but not empty header buffers. That way the original
vorbisdec unit test still passes (#451145); also, take into account
that those empty packets might carry a granulepos.
* tests/check/Makefile.am:
* tests/check/elements/vorbisdec.c:
(_create_codebook_header_buffer), (_create_audio_buffer),
(GST_START_TEST), (vorbisdec_suite):
Add unit test that sends an empty packet.
2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Don't error out on 0-sized packets, just emit a warning because this is
not a fatal error. Fixes #451145.
2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/plugins/: Update docs with caps info.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update docs with caps info.
2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
po/POTFILES.in: Add more files with translatable strings (#450875).
Original commit message from CVS:
* po/POTFILES.in:
Add more files with translatable strings (#450875).
2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
The chain should be freed if we error out here, else it will leak.
* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
(cleanup_decodebin):
Don't forget to *properly* remove the signals, else it will leak.
2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
MAINTAINERS: Updating all the maintainers files
Original commit message from CVS:
* MAINTAINERS:
Updating all the maintainers files
2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
Original commit message from CVS:
* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
(main):
Destroy and recreate parse-launch based pipeline after stop to be able
to play again. Reorder some code and add more comments.
2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
When handling a delayed-caps notification case, mark
the group as dynamic so that the nbdynamic count is
incremented and decremented correctly. Fixes: #449156
Patch by: Wim Taymans <wim@fluendo.com>
2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* win32/common/config.h:
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-06-19 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Change minimum rate back to 1000 to allow low-sample-rate wav files
to play back.
2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/vi.po: Update translations.
Original commit message from CVS:
* po/vi.po:
Update translations.
2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
gst/playback/gstqueue2.c: Fix compile error from ignored return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Fix compile error from ignored return value.
2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
Update tmpbuf for all neccesary rows, not just one, as is required
when downscaling.
Fixes #402076.
2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
(eos_buffer_probe):
Add a test that ensures we set DELTA_UNIT on all non-header,
non-video buffers, if we have a video stream.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_process_best_pad):
Move setting delta_pad to earlier, where we inspect all pads, so
that leading audio pages don't get DELTA_UNIT unset if they come
before the first DELTA_UNIT from video pages. Fixes the newly-added
test. Fixes #385527.
2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
fails on the p5-ppc64 build bot and the failure looks like it is due
to the same issue as #348114, ie. a compiler bug.
2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstqueue2.c: Fix build on MacOSX.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Fix build on MacOSX.
2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
Fix compilation on mingw. Fixes #446972.
2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
Original commit message from CVS:
Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_enqueue):
Fix a division by zero when the max percent is <= 0. Fixes #446572.
also update the buffering status when receiving events. Fixes #446551.
2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_peer_query),
(gst_queue_handle_src_query):
Wait for preroll before attempting to forward a duration query upstream.
Fixes #445505.
2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Use G_GINT64_CONSTANT macro for int64 constant.
* win32/common/libgstinterfaces.def:
* win32/common/libgsttag.def:
Add new exported functions.
2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
The BOS page of the first Dirac video stream needs to come before
the BOS page of any Vorbis streams or other audio streams, just like
it is with Theora.
2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Fix compilation.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_range):
Fix compilation.
2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_init),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
(gst_queue_src_activate_pull):
Add pull based scheduling and fix some deadlocks. Fixes #444523.
Does not yet completely work because duration queries upstream won't
block yet.
2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
Some more fseeko checks.
Original commit message from CVS:
* configure.ac:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Some more fseeko checks.
2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
configure.ac: check for large file support.
Original commit message from CVS:
* configure.ac:
check for large file support.
2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
Original commit message from CVS:
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
* gst/subparse/gstsubparse.c: (parse_subrip),
(subviewer_unescape_newlines), (parse_subviewer),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a unit test for both SubViewer formats.
2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
Don't overflow intermediate values when seeking to large time values
in audiotestsrc.
2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Include stdio to define fseeko.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
Include stdio to define fseeko.
2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
(gst_v4lsrc_query):
Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
our own implementation.
2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Handle timestamp wraparound.
2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (no_more_pads_full),
(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
(gst_uri_decode_bin_change_state):
Make sure we name srcpads uniquely even when using different internal
decodebins.
Signal no-more-pads when no more dynamic elements exist.
Remove pads on cleanup.
2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_class_init),
(gst_queue_init), (gst_queue_finalize),
(gst_queue_write_buffer_to_file), (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_is_empty), (gst_queue_is_filled),
(gst_queue_change_state), (gst_queue_set_temp_location),
(gst_queue_set_property):
Add support for filebased buffering. Fixes #441264.
2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
(caps_notify_group_cb), (gst_decode_group_new),
(gst_decode_group_free):
Add support for delayed caps fixation when autoplugging.
Optimize cases where a multiqueue is not needed/wanted, like right after
anything that is not a demuxer.
2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
consideratly speedup ogg chain detection by not trying to find a base
timestamp for skeleton streams.
2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_remove_flush),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Add support for remuve_flush.
2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
Add draft design for forcing keyframes in encoders and implement in theoraenc.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
* ext/theora/theoraenc.c: (theora_enc_sink_event),
(theora_enc_chain):
Add draft design for forcing keyframes in encoders and implement in
theoraenc.
2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.13 ===
2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/config.h:
* win32/vs6/grammar.dsp:
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstadder.dsp:
* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstaudioconvert.dsp:
* win32/vs6/libgstaudiorate.dsp:
* win32/vs6/libgstaudioresample.dsp:
* win32/vs6/libgstaudioscale.dsp:
* win32/vs6/libgstaudiotestsrc.dsp:
* win32/vs6/libgstcdda.dsp:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstdecodebin2.dsp:
* win32/vs6/libgstdirectsound.dsp:
* win32/vs6/libgstffmpegcolorspace.dsp:
* win32/vs6/libgstgdp.dsp:
* win32/vs6/libgstinterfaces.dsp:
* win32/vs6/libgstnetbuffer.dsp:
* win32/vs6/libgstogg.dsp:
* win32/vs6/libgstpbutils.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstriff.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstsinesrc.dsp:
* win32/vs6/libgstsubparse.dsp:
* win32/vs6/libgsttag.dsp:
* win32/vs6/libgsttheora.dsp:
* win32/vs6/libgsttypefindfunctions.dsp:
* win32/vs6/libgstutils.dsp:
* win32/vs6/libgstvideo.dsp:
* win32/vs6/libgstvideorate.dsp:
* win32/vs6/libgstvideoscale.dsp:
* win32/vs6/libgstvideotestsrc.dsp:
* win32/vs6/libgstvolume.dsp:
* win32/vs6/libgstvorbis.dsp:
Release 0.10.13 "What's going on?"
Original commit message from CVS:
Release 0.10.13 "What's going on?"
2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/de.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
In riff, the depth is stored in the size field but it just means that
the least significant bits are cleared. We can therefore just play
the sample as if it had a depth == width. Fixes: #440997
Patch by: Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Define inline when needed on win32 builds. Fixes: #441295
2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun),
(no_more_pads_full):
Stop buffering when the group is commited because the queues filled up.
Fixes #442024.
2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
(gst_alsa_mixer_free), (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_interface_supported),
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_get_property),
(gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
(gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h:
Revert commits towards #152864 made so far. We'll pick it up again
after the 0.10.13 release.
2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes #417992.
2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Don't add channels and rate fields to the template caps for
audio/x-dts, as wavparse might not always be able to set them,
which would then lead to 'caps are not a real subset of the
template caps' warnings.
2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
Handle unknown or invalid pads without crashing, as might occur if
a media file like an mp3 is specified as a subtitle file.
Fixes: #410039
2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
(setup_sinks):
Block the subtitle bin output queue before ghosting it and linking,
then unblock after. This avoids spurious not-linked errors caused
by the queue starting up (because it gets linked when it is ghosted).
Fixes: #350299
2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
file. Avoids flukes where the input gets typefound to some valid but
useless type.
2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
Add unit test for gnomevfssink seeking and position reporting for
file:// URIs.
2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
ext/gnomevfs/gstgnomevfssink.*: see #412648.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
* ext/gnomevfs/gstgnomevfssink.h:
Fix position reporting, especially after a seek (from upstream),
see #412648.
2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
Repair umlaut.
2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Specify the full valid range for MP3 samplerates. Fixes a regression
caused by extra header checks since the last release.
2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
sys/: Fix a locking-order bug I introduced with my changes the other day.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Fix a locking-order bug I introduced with my changes the other day.
Patch by Mike Smith.
2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_data_packet):
Don't look inside 0-length packets (which indicate duplicated
frames)
2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
Small cleanups.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes #415001.
2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update spec
Original commit message from CVS:
update spec
2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
* ChangeLog:
* gst-libs/gst/interfaces/mixer.h:
gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
set_option, get_option, _gst_reserved):
Revert reordering functions (keep ABI).
2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/: When we create our own window, indicate that we handle the
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_show_frame):
When we create our own window, indicate that we handle the
WM_DELETE client message from the window manager, so that it won't
kill our window (and our app) along with it. Handle ClientMessage,
post an error on the bus, and close the window. Further buffers
arriving will result in a FlowError because the window has been
destroyed.
Fixes: #393975
Clean up the X event handling loop and make them the same for
both xvimagesink and ximagesink while I'm at it.
2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
Make decodebin2 autoplug depayloaders too.
* gst/playback/gsturidecodebin.c: (source_new_pad):
Set the newly created decoder in a usable state when autoplugging a
dynamic source such as RTSP.
2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Ignore video-codec tag for audio streams and ignore audio-codec tags
for video streams. Should make codec name collection a bit more
robust against sloppy demuxers that send tag events containing both
tags down each pad.
2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_rates):
Tweak the buffering thresholds a little.
Update the buffer size with the previously calculate rate instead of
only when we calculate a new rate so that we get smoother buffering
updates.
* gst/playback/Makefile.am:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(source_no_more_pads), (new_decoded_pad), (array_has_value),
(gen_source_element), (has_all_raw_caps), (analyse_source),
(remove_decoders), (make_decoder), (remove_source),
(source_new_pad), (setup_source), (decoder_query_init),
(decoder_query_duration_fold), (decoder_query_duration_done),
(decoder_query_position_fold), (decoder_query_position_done),
(decoder_query_latency_fold), (decoder_query_latency_done),
(decoder_query_seeking_fold), (decoder_query_seeking_done),
(decoder_query_generic_fold), (gst_uri_decode_bin_query),
(gst_uri_decode_bin_change_state), (plugin_init):
New element that intergrates a source, optional buffering element and
decodebin.
2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
Original commit message from CVS:
* configure.ac:
Bump libtheora requirement to 1.0alpha5 for the pixformat check
(also has a .pc file, so we don't need the fallback check any
longer). Fixes #438840.
2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstqueue2.c: fix build.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
(apply_segment), (apply_buffer), (update_buffering),
(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_filled),
(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
(plugin_init):
fix build.
2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
(gst_queue_getcaps), (gst_queue_bufferalloc),
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
(apply_buffer), (update_buffering), (reset_rate_timer),
(update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_empty),
(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
(gst_queue_loop), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_change_state),
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
On our way to playbin2 this is the new network queue that does buffering
all by itself using high and low watermarks. It can also measure up and
downstream bandwidth to optimally size the queue.
2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
Original commit message from CVS:
* docs/Makefile.am: Install docs even when --disable-gtk-doc
is disabled. This matches the behavior of gtk+. Fixes #349099.
2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
Some more chained streaming ogg timestamp fixes.
2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Add some FIXMEs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_handle_page):
Add some FIXMEs.
Fix chain start/stop segment handling based on patch by
<ahalda at cs dot mcgill dot ca> see #320984.
2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
configure.ac: We don't require a C++ compiler. So don't require one.
Original commit message from CVS:
* configure.ac:
We don't require a C++ compiler. So don't require one.
2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
* ChangeLog:
* ext/alsa/gstalsamixer.c:
ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track):
Apply some of the cleanup Tim suggested in #152864 afterwards.
2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
_GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
gst_alsa_mixer_element_interface_supported,
gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
gst_alsa_mixer_element_set_property,
gst_alsa_mixer_element_get_property,
gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
volume_changed, option_changed, _gst_reserved):
Implement notification for alsamixer. Fixes #152864
2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add support for video/x-raw-bayer.
2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Add some sanity checking for the XVImage size returned by X.
Related to #377400.
2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_collect_chain_info):
If there is a stream in a chain without any data packets, ignore the
stream in the total length calculations. Might be related to #436820.
2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Consolidate and re-work our mpeg system stream detection to probe
more packets and produce a higher confidence result. Fixes a
regression caused by lowering the typefind probability last year
- related to bug #397810. Remove the redundant MPEG-1 specific
typefind function, as the new one detects both MPEG-1 & MPEG-2
happily.
Also cleanup the MPEG elementary and MPEG-TS detection functions a
little.
Tested against my media test directory, with some improvements and
no regressions.
2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
(queue_out_of_data):
Connect to the new queue "pushing" signal instead of the broken
"running" one.
2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet):
gst_pad_push_event() does not return a GstFlowReturn!
2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/: Some small cosmetic changes.
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (stop_cb), (main):
* tests/examples/seek/seek.c: (do_seek):
Some small cosmetic changes.
2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
* ChangeLog:
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
Original commit message from CVS:
* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
gst_adder_change_state):
* gst/adder/gstadder.h (bps, offset, collect_event, segment,
segment_pending, segment_position, segment_rate):
Handle playback-rate on adder.
2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_dec_sink_event), (theora_handle_comment_packet),
(theora_handle_type_packet), (theora_dec_change_state):
Don't push events (newsegment, tags) before initialising the
decoder.
This is neccesary for seeking to work correctly in gnonlin.
2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/: gst/audiotestsrc/gstaudiotestsrc.c
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_create_white_noise):
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
volume_sink_template, volume_src_template, gst_volume_init,
volume_process_double, volume_process_int16,
volume_process_int16_clamp):
Doc fixes and formatting.
2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Minimal check for volume's GstController usability; also another
test for #422295.
2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix it so that it (a) makes sense and (b) doesn't break
everything cdda-related including the unit test.
2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix build when disabling asserts.
2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
When XShm is not available, we might get row strides that are not
rounded up to multiples of four; this is bad, because virtually
every RGB-processing element in GStreamer assumes rowstrides are
rounded up to multiples of four, so let's allocate at least enough
memory to avoid crashes in this case. The image will still be
displayed distorted though if this happens, so that still needs
fixing (maybe by allocating a bigger image with an 'even' width
and then clipping it appropriately when rendering - something for
Xlib aficionados in any case).
2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If a buffer doesn't have a timestamp, assume it's contiguous with
the previous buffer, and synthesise timestamps appropriately.
2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
Original commit message from CVS:
* tests/check/elements/videorate.c: (GST_START_TEST):
Set buffer timestamp to a valid value in order to test the buffer
really does stay in videorate.
2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
There is no sensible way to handle incoming buffers which don't have a
valid timestamp. We therefore discard them and wait for the next one.
2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Better error message for text files.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
* gst/playback/gstdecodebin2.c: (plugin_init):
Better error message for text files.
2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
Fix offset bug in generation RR packets.
2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
Original commit message from CVS:
2007-04-27 Julien MOUTTE <julien@moutte.net>
* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_push_forward), (theora_handle_data_packet),
(theora_dec_decode_buffer): Calculate buffer duration correctly
to generate a perfect stream (#433888).
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont): Glib provides ABS.
2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix RB block parsing and writing.
Add support for constructing BYE packets.
2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
When posting a warning message because samples were dropped, post something more intelligible than he default error m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
(gst_base_audio_src_create):
* po/POTFILES.in:
When posting a warning message because samples were dropped, post
something more intelligible than he default error message for clock
errors which is just confusing in this context (#432984).
2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
(read_packet_header), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
(gst_rtcp_packet_sdes_get_item_count),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_entry),
(gst_rtcp_packet_sdes_next_entry),
(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement code to write SR, RR and SDES packets.
2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
Original commit message from CVS:
Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
* sys/ximage/ximagesink.c:
Fix build if XShm is not available (#432362).
2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
pointers to random memory which are passed to g_free() when
audio_convert_prepare_context() is called the first time.
2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
Original commit message from CVS:
Patch by: Dan Williams <dcbw redhat com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
Don't leak incoming buffer if gst_pad_push() returns a
non-OK flow. Fixes #432755.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Unit test for the above by Yours Truly.
2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_sink_event), (gst_adder_collected):
Fix non-flushing segmented seeks, Fixes #340060 for me
2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery: add API keyword
Original commit message from CVS:
ChangeLog surgery: add API keyword
2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_dispose):
Chain up to parent class in dispose function; get rid of
unnecessary 'diposed' flag in private structure (#415001).
2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
Some minor docs fixes and additions; also add missing 'Since' bits.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
Some minor docs fixes and additions; also add missing 'Since' bits.
2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
Original commit message from CVS:
Patch by: Zeeshan Ali <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).
2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.
2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
Original commit message from CVS:
* tests/check/elements/audioresample.c:
Add unit test for audioresample shutdown crasher (#420106).
2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/subparse/: Use GST_DISABLE_XML here
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
Use GST_DISABLE_XML here
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_buffer_alloc),
(gst_xvimagesink_navigation_send_event):
* sys/xvimage/xvimagesink.h:
Include stdlib.h when using atoi.
* tests/check/elements/playbin.c: (playbin_suite):
Use GST_DISABLE_REGISTRY here
2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
Original commit message from CVS:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
(theora_enc_sink_event), (theora_enc_change_state):
Track initialisation state; don't try to use encoder state if we're
not initialised (it'll segfault).
2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/pipelines/.cvsignore: Fix build.
Original commit message from CVS:
* tests/check/pipelines/.cvsignore:
Fix build.
2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.
2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Allow random depths between 1 and 32 instead of only multiplies of 8.
2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set the maximum number of channels for PCM and float in the correct
place to have it also used when creating the template caps.
2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Correctly support 4, 6 and 8 channels with normal PCM and float
wav files.
Fix the depth and signedness calculation in extensible wav files and
also handle 1, 2, 4, 6, 8 channels here when a file without channel
mask is found.
Add support for float, alaw and mulaw in extensible wav files.
This allows correct playback of all but 5 files from
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
(gst_riff_create_audio_template_caps):
Add voxware and float formats to the template caps.
2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
Use the correct format strings for integer formats.
2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-plugins-base.doap:
fix release date
Original commit message from CVS:
fix release date
2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-plugins-base.doap:
fix release date
Original commit message from CVS:
fix release date
2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
Original commit message from CVS:
* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
Don't use pad_alloc_buffer_and_set_caps to create a small header
packet, or, worse, to create a big temporary video buffer using the
src pad.
2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, buffer_probe_cb, GST_START_TEST):
Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
add debug
Original commit message from CVS:
add debug
2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* tests/check/pipelines/streamheader.c:
tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
streamheader_suite):
Add another test set up for failure
2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
* gst/gdp/gstgdpdepay.c:
debug changes
Original commit message from CVS:
debug changes
2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
GST_START_TEST, streamheader_suite, main):
Add a test for the streamheader bug Wim fixed.
2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/theora/theoradec.c: Fix misleading comment.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix misleading comment.
2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
More sanity checks for the header fields.
2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Try encodings from all environment variables, not just those in the
first environment variable that is set.
2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videorate/gstvideorate.c: Add some debug.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_chain):
Add some debug.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Added check for videorate changing caps handling. Closes #421834.
2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
Use scale functions to avoid overflow when calculating duration of
vorbis buffers.
2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
API: add gst_tag_freeform_string_to_utf8() (#405072).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
API: add gst_tag_freeform_string_to_utf8() (#405072).
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
Use gst_tag_freeform_string_to_utf8() here.
2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
log tweaking
Original commit message from CVS:
log tweaking
2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event):
Make sure we set the IN_CAPS flag correctly.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Get the IN_CAPS flag before we call functions that mess with the flags.
2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* gst/gdp/gstgdppay.c:
gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Only stamp buffers with offset/offset_end right before they get
pushed. This ensures offset continuity, which was not the case
before as shown by
gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
adding debugging
Original commit message from CVS:
adding debugging
2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
* common:
* gst-plugins-base.spec.in:
update spec file for RTP changes
Original commit message from CVS:
update spec file for RTP changes
2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink),
(gst_play_bin_change_state):
Activate sync in playbin, we are ready to handle it for live streams.
2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
Original commit message from CVS:
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream), (playbin_suite):
Add small test for stream-info-value-array code paths.
2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstplaybasebin.c: (add_stream):
Fix leak in add_stream(), when g_value_set_object() increases the
refcount of streaminfo object. Fixes #426250.
2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a test pattern called "circular", which has concentric
rings with varying radial frequency. The main purpose of this
pattern is to test fidelity loss in a filter or scaler element.
Notably, this pattern is scale invariant, and is optimally viewed
with a width (and height) of 400.
2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
(deactivate_free_recursive):
Decodebin2 doesn't unref pads it obtains in some occasions:
- multiqueue src pads, when either connecting further or exposing
- sink pads of new autoplugged elements
- peer pads when recursively freeing elements
Fixes #425455.
2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add audio/x-raw-float support, now that audioconvert support
non-native endianness floats.
2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
gstreamer-plugins-base.pc doesn't exist, it's
gstreamer-plugins-base-0.10.pc.
2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
with some minor changes
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes #339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_base_init),
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add Private structure.
Bring element code to 2007.
Parse clock-base caps param and use it when generating the
newsegment.
Reset variables before going to PAUSED.
Fix some docs.
2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
Add RTCP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_get_adapter):
Add RTCP docs.
Fix some more docs.
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
(gst_rtcp_packet_sr_get_sender_info),
(gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
(gst_rtcp_packet_sdes_get_chunk_count),
(gst_rtcp_packet_sdes_first_chunk),
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
(gst_rtcp_packet_bye_get_ssrc_count),
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_get_reason_len),
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Add new helper object for parsing and creating RTCP messages.
2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
PCM samples with width=8 must be always unsigned, no matter what
depth they have.
2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
Original commit message from CVS:
2007-03-29 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
perfect offsets also, not just timestamps.
* tests/check/elements/videorate.c (test_more): Test that given
any incoming offsets, that videorate produces perfect offsets.
2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add some more RIFF formats.
2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix fixed payload names and docs.
Added method to get the default clock rates of fixed payload types.
API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
Original commit message from CVS:
* tests/check/pipelines/.cvsignore:
Add new vorbisdec test to cvsignore.
2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes #420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* tests/check/libs/tag.c: (GST_START_TEST):
Make sure we parse floating-point numbers in vorbis comments
correctly with either '.' or ',' as separator, no matter what
the current locale is. Add unit test for this too.
2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/pipelines/vorbisdec.c:
commit new file
Original commit message from CVS:
commit new file
2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
When writing out floating-point numbers to vorbis comment tags, always
use the same character as separator no matter what the current locale is
(fixes #423051).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit tests for replaygain tags in vorbis comments (closes #423055).
2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
Original commit message from CVS:
* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
vorbis_handle_data_packet):
Correctly set DURATION to generate a timestamp-continuous stream.
One bug left at the end; see
ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
* tests/check/Makefile.am:
* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
Add a test to check this. Without the above patch this test fails.
2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update spec file
Original commit message from CVS:
update spec file
2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_reset), (gst_video_rate_chain):
If videorate changes caps, we can no longer use the old buffer
(which may have a different size, incompatible with our caps).
So don't do that; just duplicate the new frame more times.
2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
Remove playbin's override of the set_clock vmethod. It's irrelevant
after Wim's commit on the 19th.
2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Don't cache file sizes. Fixes #341078.
2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink):
Use GST_PTR_FORMAT to log caps.
2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (handle_start_font):
Special-case some more colour names that pango doesn't handle by
default. Fixes #420578.
2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
If we get a zero-sized input buffer, don't pass it to libvorbis, as
that marks EOS internally. After that, libvorbis will buffer all
input data, and encode none of it, eventually leading to memory
exhaustion.
2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink):
Don't post STATE_DIRTY anymore.
* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Remove stream_time reset in seek handling, core does that now.
Disable clocking for live pipelines by forcing a NULL clock to the
complete pipeline, core is too smart now for our previous hack.
We can always autoplug in PAUSED now.
2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
Original commit message from CVS:
* REQUIREMENTS: Update this file, change the formatting to make
it more consistent, plus more machine readable.
2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont), (audioresample_transform):
Don't trigger discontinuities for very small imperfections; a filter
flush will sound bad, and many plugins have rounding errors leading
to these.
2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Add min-ptime property to RTP base audio payloader. Patch by
olivier.crete@collabora.co.uk.
Fixes #415001
Indentation/whitespace/documentation fixes.
2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
Original commit message from CVS:
2007-03-14 Julien MOUTTE <julien@moutte.net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_transform_size), (audioresample_do_output),
(audioresample_transform), (audioresample_pushthrough): Handle
discontinuous streams.
* gst/audioresample/gstaudioresample.h:
* tests/check/elements/audioresample.c:
(test_discont_stream_instance), (GST_START_TEST),
(audioresample_suite): Add a test for discontinuous streams.
* win32/common/config.h: Updated.
2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
po/: Update translations from translation project.
Original commit message from CVS:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update translations from translation project.
2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/gdp/gstgdpdepay.c:
add buffer logging
Original commit message from CVS:
add buffer logging
2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
Original commit message from CVS:
* gst/audioresample/debug.h:
* gst/audioresample/resample.c: (resample_init):
Since I really am not interested in a debug line for each sample
being processed, move the library's debugging to its own category,
libaudioresample
2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/audioresample/gstaudioresample.c:
add debugging and reformat docs
Original commit message from CVS:
add debugging and reformat docs
2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet):
Since the plugin doesn't support anything other than 4:2:0 right
now, post an error and fail if we get something else. Won't matter
until libtheora supports the other pixel formats, but hopefully
that'll be soon...
2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
* ChangeLog:
I'm too lazy to comment this
Original commit message from CVS:
Mention Patch by: Alex Lancaster in a recent commit.
2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
Original commit message from CVS:
* examples/app/.cvsignore:
The buildbot demands .cvsignore files, and I comply.
2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
Add appsrc/appsink example.
Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/app/Makefile.am:
* examples/app/appsrc_ex.c:
Add appsrc/appsink example.
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst/app/gstapp.c:
Add appsink.
2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Also accept partial dates with only year and month,
like 1999-12-00 (fixes #410396 even more).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit test for the above.
2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
Original commit message from CVS:
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add unit test for MPL2 subtitle format (#413799).
2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
gst/subparse/: Add support for MPL2 subtitle format (#413799).
Original commit message from CVS:
Patch by: Kamil Pawlowski <kamilpe gmail com>
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
* gst/subparse/mpl2parse.h:
Add support for MPL2 subtitle format (#413799).
2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: We require core CVS for the new buffer metadata copy functions.
Original commit message from CVS:
* configure.ac:
We require core CVS for the new buffer metadata copy functions.
2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Fixes #414496.
2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/libvisual/visual.c: Improve adapter usage and comments.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
Improve adapter usage and comments.
2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
Use new metadata copy function.
Original commit message from CVS:
* ext/pango/gsttextrender.c: (gst_text_render_chain):
* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
Use new metadata copy function.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
Basetransform copied the metadata for us.
2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
(gst_text_overlay_video_event):
Some more logging. Only accept newsegment events in TIME format and
send a WARNING message if they are not in TIME format.
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
(gst_sub_parse_chain), (gst_sub_parse_sink_event):
* gst/subparse/gstsubparse.h:
No need to allocate GstSegment structure dynamically, just put it
into the instance structure; ignore newsegment events in BYTE
format and in particular don't let it overwrite our saved TIME
segment from the last seek.
2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
Replace AC3 typefinder with one that isn't terrible, and actually
works usefully.
2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
fix error category and translatable string
2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
pkgconfig/: Fix up utils => pbutils here too.
Original commit message from CVS:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Fix up utils => pbutils here too.
2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer):
Break out of loop in chain function as soon as possible if we get
a non-OK flow return.
2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
Original commit message from CVS:
* tests/check/elements/alsa.c: (GST_START_TEST):
Unref the mixer if the state change fails too (if the
alsa devices are inaccessible, for example)
2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
Original commit message from CVS:
* tests/check/Makefile.am:
Don't test libvisual elements in the states check, because libvisual
seems to leak internally.
Re-enable the alsa and states tests now that there's new suppressions
in gst.supp.
* tests/check/elements/alsa.c: (GST_START_TEST):
Don't leak the alsamixer we instantiated.
2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state), (gst_ximagesink_reset),
(gst_ximagesink_finalize):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
(gst_xvimagesink_reset), (gst_xvimagesink_finalize):
Move some cleanup stuff from the state change handler into a _reset()
function that can be called from _finalize(). This ensures that things
get freed even if (for some reason) the NULL->READY state transition
fails in the parent class.
Even if a parent state change fails, process our downward state change
logic instead of bailing out early.
Free the correct xcontext pointer in ximagesink's xcontext_clear.
2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/alsa/gstalsasink.c: Extra log line.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
Extra log line.
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
Use pango_font_description_set_family_static instead of
pango_font_description_set_family to save a string copy (it was
leaking due to the strdup anyway)
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
Chain up in finalize.
2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: add "untranslated-label" property which should be set by
implementations at construct time (#414645).
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set "untranslated-label" when constructing mixer track objects.
* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
Unit test to check the above.
2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Fix confusing debug message.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
Fix confusing debug message.
2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-plugins-base.doap: update doap file with new version
Original commit message from CVS:
* gst-plugins-base.doap:
update doap file with new version
2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
update docs
Original commit message from CVS:
update docs
2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.12 ===
2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/config.h:
Release 0.10.12
Original commit message from CVS:
Release 0.10.12
2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* common:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/de.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Bump version to 0.10.11.4 pre-release
Original commit message from CVS:
* configure.ac:
Bump version to 0.10.11.4 pre-release
2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
Fix regression that made GStreamer skip the first samples of audio.
Fixes #414684.
2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Bump version to 0.10.11.3 pre-release
Original commit message from CVS:
* configure.ac:
Bump version to 0.10.11.3 pre-release
2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
Original commit message from CVS:
* po/POTFILES.in:
Update paths for the rename from utils to pbutils to fix the build.
2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
Change directory to install headers in from gst/utils to gst/pbutils
as well.
2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/libs/.gitignore:
moap ignore
Original commit message from CVS:
moap ignore
2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* win32/common/config.h:
* win32/common/libgstutils.def:
update defs
Original commit message from CVS:
update defs
2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
rename utils to pbutils
Original commit message from CVS:
* configure.ac:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/descriptions.c:
(gst_pb_utils_get_source_description),
(gst_pb_utils_get_sink_description),
(gst_pb_utils_get_decoder_description),
(gst_pb_utils_get_encoder_description),
(gst_pb_utils_get_element_description),
(gst_pb_utils_add_codec_description_to_tag_list),
(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
* gst-libs/gst/pbutils/descriptions.h:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_source_message_new),
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new),
(gst_missing_plugin_message_get_description):
* gst-libs/gst/pbutils/missing-plugins.h:
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/pbutils/pbutils.h:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/descriptions.h:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/install-plugins.h:
* gst-libs/gst/utils/missing-plugins.c:
* gst-libs/gst/utils/missing-plugins.h:
* gst-plugins-base.spec.in:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element):
* gst/playback/gstplaybin.c: (plugin_init):
* tests/check/Makefile.am:
* tests/check/libs/pbutils.c: (GST_START_TEST),
(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
* tests/check/libs/utils.c:
rename utils to pbutils
2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/app/Makefile.am: Install the headers.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Install the headers.
2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Hacking to address issues in 413418.
2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
Move the app library to gst-libs/gst/app (duh!)
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* ext/Makefile.am:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Move the app library to gst-libs/gst/app (duh!)
2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Add documentation for decodebin2 that indicates that the API is still unstable.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/inspect/plugin-decodebin2.xml:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
Add documentation for decodebin2 that indicates that the API
is still unstable.
2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
Original commit message from CVS:
* configure.ac:
Update to 0.10.11.2 (0.10.12 pre-release)
2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
base time is irrelevant here.
2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Improve debugging.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Improve latency and clock slaving calculations.
Improve slave clock calibration.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
When we are asked to render N sample to 0 bytes, return N.
2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.*: Remove unused dispose function.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
(gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Remove unused dispose function.
Rename lock to not interfere with alsasrc lock.
* ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
(gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
(gst_alsasrc_read), (gst_alsasrc_reset):
* ext/alsa/gstalsasrc.h:
Implement finalize function.
Use lock to protect alsa access.
Implement _reset.
Fine tune sw params.
2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* configure.ac:
typo
Original commit message from CVS:
typo
2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
configure.ac: Convert to new AG_GST style.
Original commit message from CVS:
* configure.ac:
Convert to new AG_GST style.
2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
Original commit message from CVS:
Patch by: Ed Catmur <ed at catmur dot co dot uk>
* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
Fix race condition when rapidly switching visualisations in playbin.
Fixes #401029.
2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
Original commit message from CVS:
* tests/check/Makefile.am:
Include local stuff before system installed things in LDFLAGS and
CFLAGS.
2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Improve debugging.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
Improve debugging.
2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/v4l/: Fix duration and timestamping, taking latency into account.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
(gst_v4lsrc_fixate), (gst_v4lsrc_query):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
Fix duration and timestamping, taking latency into account.
Implement latency query.
2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_new):
Fix clock name.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_query):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create):
Improve latency query code.
Use proper clock names.
2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/generic/states.c:
plug test leak
Original commit message from CVS:
plug test leak
2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/generic/states.c: Copy the states.c test from core again
Original commit message from CVS:
* tests/check/generic/states.c: (GST_START_TEST):
Copy the states.c test from core again
* tests/check/Makefile.am:
ignore cdio and cdparanoiasrc
2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index), (check_default),
(audio_convert_prepare_context), (audio_convert_convert):
Also make valgrind happy and avoid copying data in some cases.
2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/generic/states.c:
use a macro
Original commit message from CVS:
use a macro
2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes #339837 even more.
2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
Original commit message from CVS:
2007-02-27 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (do_seek), (set_update_scale),
(msg_segment_done): Fix various seeking bugs (Slider was not
updating when doing a non flushing seek, Reverse playback
on segment seek was wrong).
2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
Add a new plugin/library to make it easy for apps to shove data into a pipeline.
Original commit message from CVS:
* configure.ac:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Add a new plugin/library to make it easy for apps to shove
data into a pipeline.
2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_seek):
When we stop scrubbing, don't leave the pipeline PLAYING when we
requested a PAUSED state.
2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse date strings in vorbis comments that have an invalid (zero)
month or day (#410396).
* tests/check/libs/tag.c: (GST_START_TEST):
Test case for the above.
2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/alsa/Makefile.am:
* gst/audiotestsrc/Makefile.am:
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Improve docs: point out that the application needs to assist playbin
with buffering.
2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
* tests/check/libs/utils.c: (missing_msg_check_getters):
Change GStreamer marker prefix in detail string from 'gstreamer.net'
to just 'gstreamer'. Document the caps string component of the
decoder/encoder detail a bit better, since not everyone will be
familiar with the GStreamer media type/caps system (but they better
enjoy nested itemized lists).
2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
Original commit message from CVS:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
Fix copying of GstNetBuffer (would crash before, or at least lead to
invalid memory access, #410772), for now by copying the GstBuffer copy
code from the core over here so we can copy the GstBuffer fields on a
provided buffer instance (of type GstNetBuffer in this case). Would be
better to fix this with some support by the core though (and in the long
run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
* tests/check/Makefile.am:
Enable unit test for GstNetBuffer.
2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* gst-libs/gst/audio/gstbaseaudiosink.c:
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-22 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
* gst/audioconvert/gstchannelmix.h:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add float as an intermediate format, as well as float mixing. Enable
test that was failing before. Fixes #339837
2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek):
Undo the previous commit: -1 as a stop time implies that the stop
time is the end of file, clearing any previously configured segment.
2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek):
Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/volume/gstvolume.c: Unbreak volume, value remains gint.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps):
Unbreak volume, value remains gint.
2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_double), (volume_process_float),
(volume_process_int16), (volume_process_int16_clamp),
(volume_set_caps), (volume_transform_ip), (volume_update_volume):
* gst/volume/gstvolume.h:
Extend float audio support (double) and some int->uint cleanups.
2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
(sort_end_pads), (gst_decode_group_expose),
(gst_decode_group_hide):
Don't free groups from the streaming threads. Just put them aside and
free them in dispose.
2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_element),
(pad_added_group_cb), (gst_decode_group_check_if_blocked),
(sort_end_pads), (gst_decode_group_expose):
Handle dynamic pads within groups.
Sort pads before exposing them in order to make playbin happy.
There still is a race with the multiqueue filling up. This should be
solved separately.
Fixes #398721
2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
Original commit message from CVS:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
Some more docs (and descriptions for two subtitle formats).
2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/audio.c: Fix documentation.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix documentation.
2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
Original commit message from CVS:
Patch by: Yves Lefebvre <ivanohe abacom com>
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
Don't leak caps. Fixes #408278.
2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
More docs coverage and some ChangeLog surgery (add missing names)
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.h:
* ext/ogg/gstoggdemux.h:
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/videoorientation.h:
* gst/adder/gstadder.h:
More docs coverage and some ChangeLog surgery (add missing names)
2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/: Small constifications.
Original commit message from CVS:
* sys/ximage/ximagesink.c:
(gst_ximagesink_calculate_pixel_aspect_ratio):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_calculate_pixel_aspect_ratio):
Small constifications.
2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
Ignore errors in reset, these are not fatal. They also grab the element
lock which is already taking when this function is called. Fixes
#405451.
2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
add header file for easy codec install
Original commit message from CVS:
add header file for easy codec install
2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
Original commit message from CVS:
* configure.ac:
Remove 'tests/examples/xerror/Makefile' from output files again.
2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Also crossref against gst-plugins-base-libs.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
Also crossref against gst-plugins-base-libs.
2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
* gst-libs/gst/audio/audio.h:
Source formatting.
* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
Add own debug category.
2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
(#403597).
2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
When we have external subtitles and wait for the subtitle decodebin
to get up and running, we set up a (sync) bus handler for the
subtitle decodebin, so we can stop waiting when it posts an error
message. However, we should do that before we set the subtitle
decodebin's state to playing, otherwise things are racy and we might
miss error messages posted before we had a chance to set up the bus.
This should finally fix totem hanging on .txt pseudo-subtitle files.
2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Use gst_gdouble_to_guint64 for conversions.
* win32/common/config.h.in:
Add a define for GST_INSTALL_PLUGINS_HELPER
* win32/common/libgstaudio.def:
* win32/common/libgstcdda.def:
* win32/common/libgstnetbuffer.def:
* win32/common/libgstrtp.def:
* win32/common/libgutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstnetbuffer.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstvorbis.dsp:
* win32/vs6/libgstcdda.dsp:
* win32/vs6/libgstgdp.dsp:
* win32/vs6/libgstutils.dsp:
Update and add new project files.
2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
(subrip_remove_unhandled_tags), (parse_subrip):
For SubRip (.srt) subtitles, ignore all markup tags we don't
handle (like font tags, for example).
* tests/check/elements/subparse.c:
Add test for this.
2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery
Original commit message from CVS:
ChangeLog surgery
2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (add_fakesink),
(gst_decode_bin_change_state):
Don't error out if there is no fakesink in the READY to NULL state
change, since when decodebin is re-used, we're only adding the
fakesink element in READY to PAUSED.
* tests/check/elements/decodebin.c:
(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
(decodebin_suite):
Minimal unit test to make sure we can use the same decodebin
instance twice (at least with audiotestsrc input).
2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
Try to get devic-name from device string first, and from handle only
as fallback (seems to yield better results and is more robust
against buggy probing code on the application side).
2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
Original commit message from CVS:
Based on patch by: Julien Puydt <julien.puydt at laposte net>
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
(gst_alsa_find_device_name):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
Improve device-name detection a bit, especially in the case where
the device is not actually open (#405020, #405024). Move common code
into gstalsa.c instead of duplicating it.
2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
Original commit message from CVS:
2007-02-06 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_get_xv_support),
(gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_interface_supported),
(gst_xvimagesink_probe_get_properties),
(gst_xvimagesink_probe_probe_property),
(gst_xvimagesink_probe_needs_probe),
(gst_xvimagesink_probe_get_values),
(gst_xvimagesink_property_probe_interface_init),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init),
(gst_xvimagesink_get_type):
* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
for XVAdaptors so that one can choose the adaptor to use with
gstreamer-properties.
2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Also mention that a conversion from double to float is suboptimal still.
2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
Clear our formats structure and free the caps contained in it when
shutting down.
2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* gst-libs/gst/audio/gstbaseaudiosink.c:
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-05 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
A width and height of 1 makes us crash, so increase minimum size to
2x2 pixels until someone feels like fixing this (#404512).
2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
Add small test to make sure request pads are cleaned up properly
even if oggmux never changes state out of NULL.
2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
Original commit message from CVS:
* tests/check/libs/utils.c: (GST_START_TEST):
Fix unit test. Turns out things work much better when you
NULL-terminate string arrays. Should make p5 build bot happy again.
2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init),
(gst_audio_filter_template_class_init),
(gst_audio_filter_template_init),
(gst_audio_filter_template_set_property),
(gst_audio_filter_template_get_property),
(gst_audio_filter_template_setup),
(gst_audio_filter_template_filter),
(gst_audio_filter_template_filter_inplace), (plugin_init):
Oops, forgot to commit fixed-up example.
2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes #403963 (and eventually also #403572).
Also document all of this a bit.
2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_spawn_child):
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Lowering log level to see why things fail on the p5 build bot;
fix some typos in unit test messages.
2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
Original commit message from CVS:
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Don't hard-code temp directory for test helper; use GLib functions
to write out file and do error checking etc.
2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
Original commit message from CVS:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_context_set_xid),
(gst_install_plugins_context_new),
(gst_install_plugins_context_free),
(gst_install_plugins_get_helper),
(gst_install_plugins_spawn_child),
(gst_install_plugins_return_from_status),
(gst_install_plugins_installer_exited),
(gst_install_plugins_async), (gst_install_plugins_sync),
(gst_install_plugins_return_get_name),
(gst_install_plugins_installation_in_progress):
* gst-libs/gst/utils/install-plugins.h:
API: add API for applications to initiate installation of missing
plugins, ie. gst_install_plugins_async() primarily.
Based on libgimme-codec by Ryan Lortie.
* configure.ac:
Add --with-install-plugins-helper configure option so distros can specify
the path of the helper script or program to call when plugin installation
is requested (distros: please do any argument munging in this helper
script instead of patching GStreamer to pass arguments differently
to another program directly).
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Build and document new API.
* tests/check/libs/utils.c: (result_cb),
(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
(libgstbaseutils_suite):
Some simple checks for the new API.
2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (test_float_conversion):
Add small test for 32bit float <=> 64bit float conversion (works
only one way so far, 32=>64 produces structured noise).
2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
po/: Add German translation (#352069).
Original commit message from CVS:
Patch by: Holger Wansing <linux wansing-online de>
* po/LINGUAS:
* po/de.po:
Add German translation (#352069).
2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
Original commit message from CVS:
reviewed by: Wim Taymans <wim@fluendo.com>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
Use newly added GstCollectPads API to free the allocated resources in
the GstOggPad structures (#402393).
2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_vis_element):
Add audioresample+audioconvert in front of the visualisation
element, so that elements like libvisual 0.4 that don't support all
samplerates can work.
Fixes: #402505
2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
Take some locks and make a copy of the streaminfo value array we
maintain while holding the lock, so that the application can
retrieve the stream-info as a value array in a thread-safe way.
2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Don't fail on 0 sized buffers. Fixes #396835.
2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Detect BBCD as video/x-dirac, so we can play raw dirac
streams.
2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
Original commit message from CVS:
* ext/theora/theoraenc.c: (theora_enc_chain):
Check return value of theora_encode_header(), or we might try to
allocate a random number of bytes. theora_encode_header() can fail
if libtheora has been compiled with encoding support disabled.
Fixes #398110.
2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/gst/.cvsignore: Do as buildbot says.
Original commit message from CVS:
* tests/check/gst/.cvsignore:
Do as buildbot says.
2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_src_setcaps):
Fix strides in libvisual. Gst uses X strides.
Inspired by: <ed at catmur dot co dot uk> and
<tim at centricular dot net>
Fixes #401118.
2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
(gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
(gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
(gst_ogg_demux_perform_seek),
(gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
(gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
(gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
(gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
* ext/ogg/gstoggdemux.h:
Properly propagate streaming errors when we are scanning the file for
chains so that we don't crash when shut down. Might fix some crashers
when quickly switching oggs in RB such as #332503 and #378436.
2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
error code as well.
2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (remove_source):
Don't try to disconnect a signal from a finalized object.
2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
Cast lock macro parameters to make sure we're actually accessing the
lock member at the right class level. Free list itself in _dispose()
as well and NULL it in case dispose gets called multiple times.
2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_bin_dispose),(gst_decode_bin_finalize):
Free GstDecodeGroups no longer used.
(gst_decode_group_expose):
Don't unlock too many times !
(deactivate_free_recursive):
Free iterator once we're done with it.
Fix for recursively deactivating elements (stop at ghostpads).
2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (handoff):
Fix up caps on the frame buffer before we save it and potentially
make it accessible to other threads via g_object_get; also use
gst_buffer_replace() instead of gst_mini_object_replace().
2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Make getting the current frame thread-safe.
2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
(gst_decode_group_new), (gst_decode_group_free):
Set queues to bigger sizes to cope with HD contents.
Fix some mutex freeing and add comment about MT safe methods.
2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
(gst_text_overlay_text_event):
Don't unnecessarily ref (and then leak) upstream events if the text
pad is not linked. Fixes #399948.
* tests/check/gst-plugins-base.supp:
Add suppression for pango on edgy/x86 for textoverlay test.
2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add some more fixed payloads.
2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
Error out properly if we get an error from libogg while reading the
BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin2.c: Don't leak mutex.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
Don't leak mutex.
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_urisource_handler),
(test_missing_suburisource_handler),
(test_missing_primary_decoder), (playbin_suite):
Run all tests once with decodebin and once with decodebin2.
One test does not pass yet with decodebin2.
2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
Fix the cases where oggmux doesn't properly figure out that all
sinkpads have gone EOS, and therefore doesn't push out the remaining
buffers and the final EOS event.
Fixes #363379
2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
sys/: Don't lock on navigation event push, just on keysym to string.
Original commit message from CVS:
2007-01-23 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Don't lock on navigation event push, just on keysym to string.
Fixes #397673 again.
2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin2.c: Cleanups.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(get_current_group), (group_demuxer_event_probe),
(gst_decode_group_expose), (deactivate_free_recursive),
(gst_decode_group_free):
Cleanups.
Don't forget to emit 'no-more-pads' once a group is exposed.
Cleanup elements from a DecodeGroup once we remove it.
Protect call to gst_decode_group_expose() with the decodebin lock.
2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
Original commit message from CVS:
2007-01-22 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Looking at Xorg code i can't figure out if that XKeysymToString
function is thread sensible or not. Lock it just in case as
recommended by Radek Doulik <rodo at ximian dot com>.
2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
sys/: Lock that X Call as well. Fixes #397673.
Original commit message from CVS:
2007-01-22 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Lock that X Call as well. Fixes #397673.
2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
Don't go into an endless loop if the file starts with 00 00 01 2X,
like quicktime redirect files might. Fixes #396042.
* tests/check/Makefile.am:
* tests/check/gst/.cvsignore:
* tests/check/gst/typefindfunctions.c: (GST_START_TEST),
(typefindfunctions_suite):
Add unit test for the above.
2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
On second thought, use "depth" field rather than "bpp" field.
2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Camtasia caps apparently need a bpp field (#398875).
2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element), (gst_play_base_bin_change_state):
Attempt at a better error message in case we don't have the required
URI handler installed; post missing-plugin message also when we're
missing an URI handler for the subtitle URI; clean up properly also
when an error occurs and we never made it to PAUSED state.
* tests/check/elements/playbin.c: (GST_START_TEST),
(playbin_suite):
Check that we're also getting a missing-plugin messsage for a
missing subtitle URI handler (and clean up properly).
2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Plug a few reference leaks.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
Plug a few reference leaks.
2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Lower probability a bit if the marker isn't right at the start,
to decrease the chance of false positives.
2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Small mpeg2 system stream typefinding improvement: make typefinder
probe a bit into the stream instead of just looking for a marker
at the beginning. Fixes #397810.
2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
Remove compatibility cruft for prehistoric GLib versions.
2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin.c: (close_pad_link):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_handle_message_func), (unknown_type):
Let decodebin be the element to post missing-plugin messages for
missing decoders (rather than playbin); make playbin implement
GstBin::handle_message so we can suppress missing-plugin messages
for types we're not handling on purpose (don't want to bring up an
installer in those cases).
2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/: Fix potentially unaligned access (#397207).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
* gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
Fix potentially unaligned access (#397207).
2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
Original commit message from CVS:
* tests/examples/seek/seek.c: (set_scale), (update_scale),
(do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
(rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
(main):
Allow to toggle looping while it plays. Fix callback prototype. Clean
up code a bit more. Add copyright header.
2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Red and blue mask was swapped (spotted by Dan Williams).
2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/tag/: Use new beats-per-minute tag from core.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Use new beats-per-minute tag from core.
2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
Original commit message from CVS:
* po/POTFILES.in:
Add new files with translatable strings, so they actually make it
into the template file one day.
2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-01-12 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
(gst_base_audio_sink_activate_pull): Remove the handwavey nego
stuff, as the base class handles this now. Actually tell the ring
buffer to start.
(gst_base_audio_sink_callback): Cast the ring buffer correctly.
How did this work before? Maybe I'm not as awesome a programmer as
I think.
* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
of a pad function.
2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
Original commit message from CVS:
* gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
Remove more fields so that the application can better blacklist
formats that have been tried before.
2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
add latest files
Original commit message from CVS:
add latest files
2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.h:
Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
used when compiling with c++ compilers as well.
2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Fix comment.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Fix comment.
2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (post_missing_element_message),
(gen_video_element), (gen_text_element), (gen_audio_element),
(gen_vis_element):
Post missing-plugin messages also when we error out because
converters, textoverlay or auto*sinks are missing (#161922).
2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
(is_demuxer_element), (new_caps):
* gst/playback/gstplaybasebin.c: (source_new_pad):
Fix the case where we try to ref a NULL element when we delay a link
because of unfixed caps.
Set the state of autoplugged decodebins to PAUSED.
RTSP now works in playbin, we can remove it from the blacklist.
2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplaybasebin.c: (string_arr_has_str),
(unknown_type), (setup_subtitle), (gen_source_element):
* gst/playback/gstplaybin.c: (plugin_init):
Post missing-plugin messages on the bus for missing sources and
missing decoders/demuxers/depayloaders; fix error code used when
we're missing an URI handler source; for media types that we are not
handling on purpose at the moment, don't print "don't know how to
handle xyz" messages to the terminal or post missing-plugin
messages on the bus.
* tests/check/elements/playbin.c: (create_playbin),
(GST_START_TEST), (gst_codec_src_uri_get_type),
(gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
(gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
(gst_codec_src_init_type), (gst_codec_src_base_init),
(gst_codec_src_create), (gst_codec_src_class_init),
(gst_codec_src_init), (plugin_init), (playbin_suite):
Add some tests for the missing-plugin stuff.
2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
API: add new libgstbaseutils library with functions
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
(find_format_info), (caps_are_rtp_caps),
(gst_base_utils_get_source_description),
(gst_base_utils_get_sink_description),
(gst_base_utils_get_decoder_description),
(gst_base_utils_get_encoder_description),
(gst_base_utils_get_element_description),
(gst_base_utils_add_codec_description_to_tag_list),
(gst_base_utils_get_codec_description), (gst_base_utils_list_all):
* gst-libs/gst/utils/descriptions.h:
* gst-libs/gst/utils/missing-plugins.c:
(missing_structure_get_type), (copy_and_clean_caps),
(gst_missing_uri_source_message_new),
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new),
(missing_structure_get_string_detail),
(missing_structure_get_caps_detail),
(gst_missing_plugin_message_get_installer_detail),
(gst_missing_plugin_message_get_description),
(gst_is_missing_plugin_message):
* gst-libs/gst/utils/missing-plugins.h:
API: add new libgstbaseutils library with functions
- to create and parse missing-plugins messages
- that provide (translated) descriptions for caps/decoders/sources/etc.
Closes #392393.
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Add new lib.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Generate docs for new lib and API.
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/utils.c: (missing_msg_check_getters),
(GST_START_TEST), (libgstbaseutils_suite):
Add some basic unit tests.
2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
Original commit message from CVS:
* ext/ogg/Makefile.am:
Dist gstoggdemux.h to fix 'make distcheck'.
* sys/v4l/Makefile.am:
Fix 'make distcheck' even more.
2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
Added docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
(gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
(gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
(gst_ogg_demux_perform_seek):
* ext/ogg/gstoggdemux.h:
Added docs.
Add some more comments.
Small cleanups.
2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
Small documentation updates/fixes
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/vorbis/vorbisdec.c:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/tag/gstvorbistag.c:
Small documentation updates/fixes
2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
Original commit message from CVS:
* configure.ac:
Require core CVS HEAD for Andy's basesrc/sink API additions.
2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
Original commit message from CVS:
Patch by: Günter Thelen <daedalus dot inc at gmx net>
* gst/typefind/gsttypefindfunctions.c: (flac_type_find),
(plugin_init):
Add typefinder for flac-in-ogg in conformance with the ogg-mapping
on flac.sf.net (there appear to be other versions of the first
ogg page in the wild) (#391365).
2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Check if localtime_r() is available.
Original commit message from CVS:
* configure.ac:
Check if localtime_r() is available.
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
If localtime_r() is not available, fall back to localtime(). Should
fix build on MingW (#393310).
2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* gst/subparse/gstsubparse.h:
Remove spurious 1000 subtrahend when calculating the timestamp from
the frame number and the frame rate . Also, use the frames/second
value specified in the first line of the file, if one is specified
there. Should fix #357503.
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
(subparse_suite):
Add some basic unit tests for the microdvd subtitle format.
2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
sys/xvimage/xvimagesink.c: Fixes : #390076.
Original commit message from CVS:
2007-01-07 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_xvimage_put),
(gst_lookup_xv_port_from_adaptor),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
(gst_xvimagesink_set_xwindow_id),
(gst_xvimagesink_set_event_handling),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
Patch by : Young-Ho Cha <ganadist at chollian dot net>
Fixes : #390076.
Add an adaptor property to select a specific XV adaptor.
* sys/xvimage/xvimagesink.h:
2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
sys/: Use flow_lock much more to protect every access to xwindow.
Original commit message from CVS:
2007-01-07 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
(gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
(gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
(gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
(gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
(gst_xvimagesink_change_state),
(gst_xvimagesink_set_xwindow_id),
(gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
Use flow_lock much more to protect every access to xwindow.
Try to catch erros while creating images in case some drivers
are
just generating an XError when the requested image is too big.
Should fix : #354698, #384008, #384060.
* tests/icles/stress-xoverlay.c: (cycle_window),
(create_window):
Implement some stress testing of setting window xid.
2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/common/libgsaudio.def: Add new exported function.
Original commit message from CVS:
* win32/common/libgsaudio.def:
Add new exported function.
* win32/common/libgstogg.dsp:
Add gstoggaviparse.c to the build.
* win32/common/libgstvideoscale.dsp:
Add vs_4tap.c to the build.
* win32/common/libgstvorbis.dsp:
Add vorbistag.c to the build.
2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* gst-libs/gst/audio/gstbaseaudiosink.c:
gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
Original commit message from CVS:
2007-01-06 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_class_init)
(gst_base_audio_sink_init):
(gst_base_audio_sink_activate_pull): Add an activate_pull function
to baseaudiosink, and tell basesink that we can work in pull mode.
This way the ring buffer thread drives the pipeline directly, if
pull mode is possible. There is some lingering nastiness regarding
capsnego, however.
(gst_base_audio_sink_callback): Implement the callback to pull
data. This interface is a bit light, though -- it should get a
GstFlowReturn return value at least.
2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
Printf format and missing argument fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst/playback/gstdecodebin2.c:
(gst_decode_group_check_if_blocked):
Printf format and missing argument fixes.
2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
(gst_ogm_parse_change_state):
Activate pads before adding them to the element.
2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (main):
* tests/examples/seek/seek.c: (main):
Call g_thread_init() first thing in main() (see #391278).
2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/netbuffer.c: (GST_START_TEST),
(netbuffer_suite):
Add test for GstNetBuffer + gst_buffer_copy(). Disabled
for the time being, since it's broken, see #393099.
2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
Original commit message from CVS:
* tests/check/Makefile.am:
Update to use GST_PLUGINS_BASE_CFLAGS as well.
2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
Original commit message from CVS:
* configure.ac:
split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
so that GST_BASE_CFLAGS can go inbetween them, making sure
we use uninstalled gst-libs headers
* docs/libs/Makefile.am:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/volume/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
* tests/icles/Makefile.am:
adapt
2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
Original commit message from CVS:
2007-01-04 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/interfaces/xoverlay.c:
(gst_x_overlay_handle_events):
* gst-libs/gst/interfaces/xoverlay.h:
* sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
(gst_ximagesink_set_xwindow_id),
(gst_ximagesink_set_event_handling),
(gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
(gst_ximagesink_get_property), (gst_ximagesink_init),
(gst_ximagesink_class_init):
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
(gst_xvimagesink_set_xwindow_id),
(gst_xvimagesink_set_event_handling),
(gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
(gst_xvimagesink_get_property), (gst_xvimagesink_init),
(gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
* tests/icles/stress-xoverlay.c: (toggle_events),
(create_window):
Add a method to the XOverlay interface to allow disabling of
event handling in x[v]imagesink elements. This will let X events
propagate to parent windows which can be usefull in some cases.
Be carefull that the application is then responsible of pushing
navigation events and expose events to the video sink.
Fixes: #387138.
2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c: (GST_START_TEST):
Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
(fixes #392070).
2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
Dist design docs.
Original commit message from CVS:
* configure.ac:
* docs/Makefile.am:
* docs/design/Makefile.am:
Dist design docs.
2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
Original commit message from CVS:
2006-12-27 Julien MOUTTE <julien@moutte.net>
* docs/libs/gst-plugins-base-libs-sections.txt: Fix a
documentation
typo. Fixes: #390063.
2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
sys/: Plug a caps leak.
Original commit message from CVS:
2006-12-27 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
caps leak.
* win32/common/config.h: Updated.
2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (cleanup_gdppay),
(setup_gdppay_streamheader):
Fix the dp tests, but activating the pads for the streamheader tests
too and cleaning up conditionaly
2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
Original commit message from CVS:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
(img_get_alpha_info):
Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
other end of the word. Fixes: #387073.
Add some inconsequential branch hints in a couple of places.
2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_smpfmt):
The "signed" field in raw audio caps is of boolean type, trying to
extract the value with _get_int() will fail (fix to keep in sync with
the copy in gst-ffmpeg)
2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/: consistent pad (de)activation
Original commit message from CVS:
* tests/check/elements/audioresample.c: (cleanup_audioresample):
* tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(cleanup_gdpdepay):
* tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
* tests/check/elements/subparse.c: (teardown_subparse):
* tests/check/elements/textoverlay.c: (cleanup_textoverlay):
* tests/check/elements/videorate.c: (cleanup_videorate):
* tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
* tests/check/elements/volume.c: (cleanup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec),
(cleanup_vorbisdec):
* tests/check/elements/vorbistag.c: (setup_vorbistag),
(cleanup_vorbistag):
consistent pad (de)activation
2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Forgot to register the extensions.
2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
(plugin_init):
Add typefinder for VIVO files (my christmas present to the 90s).
2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found):
Special-case the text/plain media type: we only want to recognise it
as a 'raw' decoded media type if it comes from a demuxer or subtitle
parser, but not if the entire stream is of text/plain type. If the
entire stream is text/plain, we should just error out.
This fixes playback of audio files with lyrics in totem. Totem can't
distinguish between text files and subtitle files and passes any
.txt file with the same basename as the main file to playbin as
suburi, and playbin will then throw a 'subtitle found, but no video
stream' error, which isn't entirely helpful. See #380342.
Also, with this change we'll show a slightly more correct error
message in case totem passes a playlist file to us (although a
custom error message wording instead of the default text would
probably not be a bad idea either).
Same problem also needs to be fixed for playbin+decodebin2.
* tests/check/Makefile.am:
* tests/check/elements/decodebin.c: (src_handoff_cb),
(decodebin_new_decoded_pad_cb), (GST_START_TEST),
(decodebin_suite):
Add simple unit test for decodebin for the above.
2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
Refuse to change state to READY when we failed to create any of the
required elements in our instance init function.
2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Small docs fixes/updates.
* gst-libs/gst/video/gstvideosink.h:
Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
removed from the base sink API between 0.9.6 and 0.9.7).
API: add GST_VIDEO_SINK_CAST and use it for the height/width
accessor macros, so we don't do a runtime GObject type check every
time we use them.
2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
add doap file
Original commit message from CVS:
* Makefile.am:
* gst-plugins-base.doap:
* gst-plugins-base.spec.in:
add doap file
2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
Declare variables at the beginning of a block. Fixes #383195.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
Declare variables at the beginning of a block. Fixes #383195.
2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Bump version nano - back to CVS.
Original commit message from CVS:
* configure.ac:
Bump version nano - back to CVS.
=== release 0.10.11 ===
2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: releasing 0.10.11, "Dumb things"
Original commit message from CVS:
=== release 0.10.11 ===
2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:
releasing 0.10.11, "Dumb things"
2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
(close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
Handle the case where an element has multiple pads with
unfixed caps as well as still possibly producing more dynamic
pads by storing each case as a distinct entry in the dynamic list.
Fixes #38223 again.
2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
Fix #382223, add more dynamic caps handling.
2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
* po/.gitignore:
Ignore all pot files
Original commit message from CVS:
Ignore all pot files
2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
gst/audiorate/gstaudiorate.c: Delete bad debug code.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Delete bad debug code.
Fixes #381219
2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
Fix compilation on win32 under VS8
Original commit message from CVS:
* gst/videoscale/vs_4tap.c:
* win32/MANIFEST:
* win32/common/config.h:
* win32/vs8/libgstvideoscale.vcproj:
Fix compilation on win32 under VS8
Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
Partially fixes #381175
2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
(GST_START_TEST):
It would be very bad if, after a discont buffer, we thought every
single following buffer was also discont. So, add to the test to
ensure that this isn't the case.
* ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
... it was the case. So fix it.
2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Improve debug.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (check_queue_event):
Improve debug.
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
padtemplate caps. Refixes #357577.
2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (check_queue_event),
(queue_threshold_reached), (queue_out_of_data),
(gen_preroll_element):
Add event probe to see when EOS is in a queue and we can disable the
underrun signals. Fixes #357577.
2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/: New decodebin2 element.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
(_gst_boolean_accumulator), (gst_decode_bin_class_init),
(gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_finalize), (gst_decode_bin_set_property),
(gst_decode_bin_get_property), (gst_decode_bin_set_caps),
(gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
(connect_element), (expose_pad), (type_found),
(pad_added_group_cb), (pad_removed_group_cb),
(no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb), (find_compatibles), (is_demuxer_element),
(are_raw_caps), (multi_queue_overrun_cb),
(multi_queue_underrun_cb), (gst_decode_group_new),
(get_current_group), (group_demuxer_event_probe),
(gst_decode_group_control_demuxer_pad),
(gst_decode_group_control_source_pad),
(gst_decode_group_check_if_blocked),
(gst_decode_group_check_if_drained), (gst_decode_group_expose),
(gst_decode_group_hide), (gst_decode_group_free),
(gst_decode_group_set_complete), (source_pad_blocked_cb),
(source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
(remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
(plugin_init):
New decodebin2 element.
Closes #370092
* gst/playback/gstplay-marshal.list:
Added marshallers for new signals in decodebin2
* gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
is set.
2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state):
Disable rtsp:// uris for the release, it's not good enough yet.
Remove unused var.
2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Implement reverse playback.
Original commit message from CVS:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_dec_push_forward), (theora_dec_push_reverse),
(theora_handle_data_packet), (theora_dec_decode_buffer),
(theora_dec_flush_decode), (theora_dec_chain_reverse),
(theora_dec_chain_forward), (theora_dec_chain):
Implement reverse playback.
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
(vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
(vorbis_dec_chain_forward):
Clear buffers used for reverse playback in _reset.
No need to set the eos flag, we clip samples using the segment.
2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Some cleanups.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
(gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
(gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
(gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
Some cleanups.
Handle continued pages in reverse mode.
2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Small cleanups.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
(vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
(vorbis_dec_flush_decode):
Small cleanups.
Don't try to add invalid timestamps.
Clipping will unref the buffer.
2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/: remove obsolete _factory_init protos
Original commit message from CVS:
* gst/adder/gstadder.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
remove obsolete _factory_init protos
2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: Fix spacing in debug message.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
Fix spacing in debug message.
2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
(gst_ogg_demux_chain):
Don't just ignore return values from _pad_push().
Small debug improvements.
2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
If our incoming buffer is marked as DISCONT, then increment the page
number (so that the discontinuity is marked in the final ogg
bitstream) and flush the previous page.
2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
Original commit message from CVS:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: (gst_theora_enc_init),
(theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
(theora_buffer_from_packet), (theora_enc_is_discontinuous),
(theora_enc_chain), (theora_enc_change_state):
Mark discontinuities of > 3/4 of a frame, reinit encoder.
* tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
(GST_START_TEST), (theoraenc_suite):
Enable discontinuity test, fix it.
2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
(gst_text_overlay_video_event), (gst_text_overlay_pop_text),
(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
(gst_text_overlay_change_state):
* ext/pango/gsttextoverlay.h:
Some textoverlay fixes: for one, in the video chain function,
actually wait for a text buffer to come in if there is none at the
moment and there should be one; also, deal more gracefully with
incoming buffers that do not have a timestamp or duration; discard
text buffer when not needed any longer. Fixes #341681.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
(setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
(create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
(test_video_waits_for_text_send_text_newsegment_thread),
(test_video_waits_for_text_shutdown_element),
(test_render_continuity_push_video_buffers_thread),
(textoverlay_suite):
Add some unit tests for textoverlay.
2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
Avoid integer underflow when the found probability for mp3 is
smaller than the 'penalty' we subtract if there's not a clean
mp3 header sync at offset 0.
2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add some new symbols to the docs
2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/ffmpegcolorspace.c:
(ffmpegcolorspace_suite):
Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
(for now not for valgrinding though, since it takes too long).
2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
Fix RGBA32 caps. Fixes #357038.
2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Add FIXME so we can add some padding here in 0.11
2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Fix GstBaseRTPAudioPayload structure so the whole GObject
inheritance business actually works (parent class instance structure
must always come first in the derived class instance structure).
2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
Original commit message from CVS:
* gst/videotestsrc/Makefile.am:
* tests/check/Makefile.am:
Make sure our checks and the videotestsrc plugin link against the
local uninstalled gst libs and not any installed gst libs that
might happen to exist as well.
* tests/check/elements/adder.c: (message_received),
(test_event_message_received), (test_play_twice_message_received):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
Fix compiler warnings when compiling against core with disabled
debugging system.
2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
(gst_audio_rate_sink_event), (gst_audio_rate_chain):
Fix audiorate, so that it accurately sets offsets and timestamps.
Doesn't change the fundamental algorithmic decisions; so should be
safe.
* tests/check/Makefile.am:
Enable audiorate test now that it passes.
2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
clear xv when going to NULL, remove // commented non-existant proto
* tests/examples/seek/seek.c: (main):
add missing tooltip description for scrub and play_scrub
2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
configure.ac: Bump liboil requirement to 0.3.8.
Original commit message from CVS:
* configure.ac:
Bump liboil requirement to 0.3.8.
* gst-libs/gst/riff/riff-media.c:
Add Dirac fourcc.
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.h:
Use liboil's stdint.h.
* gst/videotestsrc/videotestsrc.c:
Remove liboil related ifdef's, since they aren't needed now, and
won't work with future versions.
2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
Original commit message from CVS:
* gst/videoscale/Makefile.am:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* gst/videoscale/vs_image.c:
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.c:
* gst/videoscale/vs_scanline.h:
Add a 4-tap image scaler. Theoretically looks much prettier.
The tap calculation could use some improvement.
2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
Various gsize and gssize printf fixes. Fixes #372507.
Original commit message from CVS:
Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
(gst_riff_parse_strf_iavs):
* gst/subparse/gstsubparse.c: (convert_encoding):
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
(gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
(gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
Various gsize and gssize printf fixes. Fixes #372507.
2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_dec_push_forward), (vorbis_dec_push_reverse),
(vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
(vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
(vorbis_dec_chain_forward), (vorbis_dec_chain):
* ext/vorbis/vorbisdec.h:
First stab at vorbis reverse playback.
2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggdemux.c: Improve a debug line slightly.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
Improve a debug line slightly.
* ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
Call gst_riff_init() in plugin_init, to avoid getting errors from
the debug system (unrelated changes to another plugin made this turn
up; not sure why).
2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
win32/common/libgsttag.def: Add missing symbol (#366492).
Original commit message from CVS:
Patch by: Sergey Scobich <sergery.scobich at gmail com>
* win32/common/libgsttag.def:
Add missing symbol (#366492).
2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gststreamselector.c: Don't unref a NULL pad.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
Don't unref a NULL pad.
2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
(gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
(gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
(gst_ogg_demux_loop):
Implement first stab at reverse playback.
2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
add h263/h264 variants to the caps, Fixes #363118
2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Use g_strerror instead of strerror so we get UTF-8.
2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
Add/remove KW-DIRAC header here, since it is ogg-specific.
2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
Recognise more mpeg4 elementary video streams.
2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
Lower the probability of mp3 typefinding functions if we don't find a
valid mp3 header at the start of the file.
Closes #369482
2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_init),
(theora_dec_sink_event), (theora_dec_chain_forward),
(theora_dec_flush_decode), (theora_dec_chain_reverse),
(theora_dec_chain):
Document and partially implement an algorithm for doing reverse playback
of theora video.
2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
Original commit message from CVS:
Patch by: Sergey Scobich <sergey.scobich at gmail com>
* win32/common/config.h:
* win32/common/interfaces-enumtypes.c:
* win32/common/libgsttag.def:
* win32/vs8/gst-plugins-base.sln:
* win32/vs8/libgstaudioresample.vcproj:
* win32/vs8/libgstinterfaces.vcproj:
* win32/vs8/libgstogg.vcproj:
* win32/vs8/libgstriff.vcproj:
* win32/vs8/libgsttag.vcproj:
* win32/vs8/libgsttheora.vcproj:
* win32/vs8/libgstvideoscale.vcproj:
* win32/vs8/libgstvorbis.vcproj:
Misc. VS8 build fixes: fix syntax in config.h, add missing entries
to libgsttag.def; add missing dependencies for some vs8 projects;
re-arrange placement of .def files in vs8 projects (#366334).
2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstogg.c: Remove unused variable.
Original commit message from CVS:
* ext/ogg/gstogg.c:
Remove unused variable.
* ext/ogg/gstoggdemux.c:
Fix Wim's surname in plugin description.
2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
Original commit message from CVS:
* gst-plugins-base.spec.in:
spec new .h file. Fixes #368310.
2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
(gst_multi_fd_sink_get_stats),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_queue_buffer),
(gst_multi_fd_sink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Make using the remove or clear signals threadsafe.
Make calling get-stats with an invalid fd not segfault.
Fixes 368273.
2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/: Fix and activate base audio payloader.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_init):
Fix and activate base audio payloader.
2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
(plugin_init):
Add typefinder for QuickTime Image Files (see #366156).
2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Another typo fix (#366212).
2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Use stream time to synchronize volume property instead of rather random
timestamps. This is needed when gnonlin does its time shifting.
2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
* ChangeLog:
I'm too lazy to comment this
Original commit message from CVS:
*** empty log message ***
2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet dot be>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
Remove the pad from the element in release_pad.
2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
Explicitly create our custom buffer classes at a thread-safe
location as well, since g_type_class_ref() doesn't seem to be
entirely thread-safe either (#365501; also see #349410).
2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
(gst_riff_parse_info):
If strings in INFO chunk are not UTF-8, do something similar to
what we do for ID3v1 tags: check a number of environment variables
(GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
character sets to try, otherwise try the current locale and/or fall
back on ISO-8859-1. Fixes #360552.
2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_pattern_get_type),
(gst_video_test_src_set_pattern):
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
(gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
(gst_video_test_src_checkers8):
* gst/videotestsrc/videotestsrc.h:
Add a bunch of exciting new checkers patterns.
2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
Original commit message from CVS:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (handle_buffer),
(gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
* gst/subparse/gstsubparse.h:
* gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
(parse_tmplayer):
* gst/subparse/tmplayerparse.h:
Add support for TMPlayer-type subtitles (#362845).
* tests/check/elements/subparse.c: (test_tmplayer_do_test),
(GST_START_TEST), (subparse_suite):
Add some basic unit tests for the above.
2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
Original commit message from CVS:
* tests/check/elements/audiorate.c: (test_injector_base_init),
(test_injector_class_init), (test_injector_chain),
(test_injector_init), (probe_cb), (do_perfect_stream_test),
(GST_START_TEST), (audiorate_suite):
More tests for audiorate: inject buffers to check behaviour when
buffers overlap.
2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiorate.c: (probe_cb), (got_buf),
(do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
Add some basic unit tests for audiorate. Disabled at the moment
since it doesn't pass yet (see bug #363119).
2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
(parse_subrip), (handle_buffer):
Add missing closing tags for markup and fix broken markup,
otherwise pango won't render anything (fixes #357531). Also,
make sure the text we send out is always NUL-terminated
(better safe than sorry etc.).
* tests/check/elements/subparse.c: (test_srt_do_test),
(test_srt):
Some more tests for .srt incl. tests for the above stuff.
2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
Original commit message from CVS:
2006-10-20 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
Patch by: Stefan Kost <ensonic@users.sf.net>
Try to redraw borders only when needed. Apparently this consumes
resources on small devices... :-O (#363607)
2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
gst/tcp/gstmultifdsink.c: If caps change, then update the client's idea of the caps so that we don't end up re-sendin...
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_client_queue_buffer):
If caps change, then update the client's idea of the caps so that we
don't end up re-sending streamheaders for every single buffer after
the caps change.
2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
Original commit message from CVS:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
(gst_ogg_parse_append_header), (gst_ogg_parse_chain):
Set caps on pushed buffers; fix up refcounting of caps objects.
2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
(plugin_init):
Typefind mmsh header data packet to application/x-mmsh (#362625).
2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add very simple unit test for subparse.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/subparse.c: (buffer_from_static_string),
(setup_subparse), (teardown_subparse), (test_srt_do_test),
(GST_START_TEST), (subparse_suite):
Add very simple unit test for subparse.
2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (strip_trailing_newlines),
(parse_subrip):
Strip trailing newlines from subtitle text output.
2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_change_state):
Fix memleak; clear subparse->textbuf n state change function.
2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
Don't require subrip (.srt) files to start with a chunk number of 1.
2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Extract rate from the NEWSEGMENT event.
Use commit_full to also take rate adjustment into account when writing
samples to the ringbuffer.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Added _commit_full() to also take rate into account.
Use simple interpolation algorithm to resample audio.
API: gst_ring_buffer_commit_full()
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
* tests/examples/seek/seek.c: (segment_done):
Don't try to seek with 0.0 rate, just pause instead.
Remove bogus debug line.
2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
(setup_source):
Catch async errors when starting up the subtitle bin, so we can
stop waiting and continue with the main film instead of hanging
forever. Fixes #339366.
* tests/check/elements/playbin.c: (playbin_suite):
Enable unit test for the above.
2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/playbin.c: (GST_START_TEST),
(gst_red_video_src_uri_get_type),
(gst_red_video_src_uri_get_protocols),
(gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
(gst_red_video_src_uri_handler_init),
(gst_red_video_src_init_type), (gst_red_video_src_base_init),
(gst_red_video_src_create), (gst_red_video_src_class_init),
(gst_red_video_src_init), (plugin_init), (playbin_suite):
Some small and basic unit tests for playbin; not very useful yet,
but at least a start.
2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: The old pad activation spiel.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (setup_sinks):
The old pad activation spiel.
2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Don't hang forever if the subbin already fails to start up in
the state change to PAUSED (#339366).
2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
Original commit message from CVS:
* gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
(gst_tuner_set_channel), (gst_tuner_get_channel),
(gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
(gst_tuner_set_frequency), (gst_tuner_get_frequency),
(gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
(gst_tuner_find_channel_by_name):
Fix some function guards, add some more function guards.
2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (get_our_ghost_pad),
(remove_element_chain):
Don't return a pad from get_our_ghost_pad unless it is actually the
one we want.
Change a cast in remove_element_chain slightly.
2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
Original commit message from CVS:
2006-10-13 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (do_seek), (start_seek),
(rate_spinbutton_changed_cb), (segment_done),
(msg_state_changed):
Segment seeking needs to use the rate and set stop to -1.
2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Don't crash when ringbuffer is not yet created.
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
Fixes #361634.
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Activate pads befre adding them to running elements.
2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
Original commit message from CVS:
2006-10-13 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (do_seek), (start_seek),
(rate_spinbutton_changed_cb), (msg_state_changed): Stop the
scale
updater when we start grabing the slider. Don't wait for the
pipeline to be PAUSED.
2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
(gst_mixer_set_volume), (gst_mixer_get_volume),
(gst_mixer_set_mute), (gst_mixer_set_option),
(gst_mixer_get_option), (gst_mixer_mute_toggled),
(gst_mixer_record_toggled), (gst_mixer_volume_changed),
(gst_mixer_option_changed):
Guard mixer interface functions against bogus arguments.
2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
Original commit message from CVS:
2006-10-12 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (do_seek), (start_seek),
(stop_seek),
(play_cb), (pause_cb), (stop_cb),
(rate_spinbutton_changed_cb),
(msg_state_changed), (main): Use state-changed messages to
trigger
start/stop of scale update timer. Indeed the scale slider was
jumping here and there because the update timer was activated
before seek completed. This fixes instant applying of rate
changes
by pressing the spinbutton like a crazy man !
2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
Original commit message from CVS:
Patch by: Sebastien Cote <sebas642 at yahoo.ca>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_finalize):
Fix two small memory leaks (#361456).
2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
Original commit message from CVS:
2006-10-10 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (do_seek),
(rate_spinbutton_changed_cb): When changing spinbutton we try
to change the rate on the fly.
2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/: Add WMS caps.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add WMS caps.
2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
ext/gnomevfs/: Fix URI interface implementation return type.
Original commit message from CVS:
2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
Patch by: Josep Torre Valles <josep@fluendo.com>
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix URI interface implementation return type.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
Fix what looks like a copy/paste issue when assigning values.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_get_type):
Cast to prevent Forte warnings.
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Fix URI interface implementation return type.
gst_pad_query_position requires a signed integer pointer as
3rd parameter, GstClockTime is unsigned.
* gst/audioconvert/audioconvert.c:
Fix integer overflow when treated as signed.
* gst/audioresample/resample.c: (resample_add_input_data):
Cast to prevent warnings on Forte.
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
Fix integer overflow when treated as signed.
* gst/ffmpegcolorspace/imgconvert_template.h:
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
* gst/playback/gstdecodebin.c: (queue_filled_cb),
(cleanup_decodebin):
Who initialises a guint to -1!
Cast function pointers to prevent warnings on Forte.
* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
(queue_threshold_reached):
Cast function pointers correctly to prevent warnings on Forte.
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Cast function pointers correctly to prevent warnings on Forte.
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
Obvious change to unsigned, 0xEF > max signed char.
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
GstClockTime is unsigned, initialise correctly.
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
Cast so pointer arithemetic doesn't cause warnings on Forte.
* gst/videorate/gstvideorate.c:
Use correct return value.
* tests/examples/seek/scrubby.c:
GstClockTime is unsigned, initialise correctly.
2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
Original commit message from CVS:
Patch by: Ferenc Gerlits <fgerlits at gmail com>
* gst/typefind/gsttypefindfunctions.c:
Recognise XML files and XML-like files shorter than 256 bytes as
well (fixes #359237).
2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
* ChangeLog:
* common:
* gst/typefind/gsttypefindfunctions.c:
Added typefind functions to video/x-nuv media.
Original commit message from CVS:
Added typefind functions to video/x-nuv media.
2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
(gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
Some more guards against invalid input.
2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Useless goto.
Original commit message from CVS:
2006-10-07 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
Useless goto.
* tests/examples/seek/seek.c: (do_seek),
(rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
seek example to experiment with rates != 1.0 (reverse playback
!)
2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Unref message in doc-example (spotted by Robert McQueen)
2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/typefind/gsttypefindfunctions.c: printf fix.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mpeg1_parse_header), (mpeg1_sys_type_find):
printf fix.
2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/: Activate dynamic pads before adding them to the element.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(close_pad_link):
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
Activate dynamic pads before adding them to the element.
2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Fix obviously-bogus macros; use the correct types.
2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_change_state):
Also call parent state change function to activate pads.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mpeg1_parse_header), (mpeg1_sys_type_find):
Add some more debug info in mpeg typefinding.
2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_chain):
Zero byte theora packets are valid and well-defined; don't warn on
them.
2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_get_stats), (find_limits),
(gst_multi_fd_sink_queue_buffer):
API: add dropped_buffers to the get-stats GValueArray
2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
Printf format fixes.
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_device_property_probe_get_values):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
(gst_ogg_mux_process_best_pad):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
(gst_ogg_parse_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
(gst_vorbis_enc_buffer_check_discontinuous):
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_push_full):
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
* gst/audioresample/resample.c: (resample_input_pushthrough):
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(wavpack_type_find):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
* tests/check/elements/volume.c: (GST_START_TEST):
Printf format fixes.
2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
Original commit message from CVS:
* gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
Fix a simple mistake (see the docs)
Fixes #359580
2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* win32/common/config.h:
bump version
Original commit message from CVS:
bump version
2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Add vorbistag element to docs; update version numbers to 0.10.10.1.
2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
Original commit message from CVS:
Patch by: James "Doc" Livingston <doclivingston at gmail com>
* ext/vorbis/Makefile.am:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
(vorbis_parse_parse_packet), (vorbis_parse_chain):
* ext/vorbis/vorbisparse.h:
* ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
(gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
(gst_vorbis_tag_parse_packet):
* ext/vorbis/vorbistag.h:
Add new vorbistag element which derives from vorbisparse
and is essentially the same as well, only that it implements
the GstTagSetter interface and can modify the stream's
vorbiscomment on the fly (#335635).
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/vorbistag.c: (setup_vorbistag),
(cleanup_vorbistag), (buffer_probe), (start_pipeline),
(get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
(_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
Add unit test for new vorbistag element.
2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
Original commit message from CVS:
* ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
(vorbis_parse_push_headers), (vorbis_parse_chain):
Set BOS flag in packet structure to fix 'jump depends
on unitialized value' errors in valgrind; various minor
clean-ups.
2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: Fix typo in a debug statement.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
Fix typo in a debug statement.
* gst/playback/gstplaybasebin.c: (probe_triggered),
(new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
(gen_source_element), (source_new_pad), (analyse_source),
(setup_source):
When handling no_more_pads in new_decoded_pad, make sure to treat
subtitle pads correctly. Fixes playback with subtitle files.
Move a recurring message to LOG level.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
which ends up as -1 when cast to an int. Make the logic handle the
max value as an unsigned mask and only change the colorkey when it's
a value we recognise.
2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Removed empty * between paragraphs
2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/: Moved some documentation into .c file
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/README:
Moved some documentation into .c file
2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Fix compilation.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (no_more_pads):
Fix compilation.
2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Remove g_print
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_caps):
Remove g_print
* gst/playback/gstplaybin.c:
Add some docs.
2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
Original commit message from CVS:
* tests/check/Makefile.am:
Re-enable cddabasesrc test to see if it works again
now.
2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element):
Handle invalid URIs a bit more gracefully.
2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/oggmux.c: Remove obsolete comment.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c:
Remove obsolete comment.
2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
(gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
(gst_ogg_mux_collected):
Commit patch from James "Doc" Livingston, adds proper EOS handling
in oggmux. GStreamer can, for the first time ever, create a valid
Ogg file! Yay!
* tests/check/pipelines/oggmux.c: (check_chain_final_state),
(oggmux_suite):
Reenable tests now that they pass.
2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
Stop reading commands when EOF (we read 0) as well.
2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
(close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
(find_dynamic), (unlinked), (close_link):
Implement delayed caps linking needed for element with a lot of
different caps on the src pads that get fixed at runtime.
Improve management of dynamic elements.
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
(group_destroy), (group_commit), (check_queue), (queue_overrun),
(gen_preroll_element), (remove_groups), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
(new_decoded_pad), (setup_subtitle), (array_has_value),
(gen_source_element), (source_new_pad), (has_all_raw_caps),
(analyse_source), (remove_decoders), (make_decoder),
(remove_source), (setup_source), (finish_source), (prepare_output),
(gst_play_base_bin_change_state):
* gst/playback/gstplaybasebin.h:
Use more _CAST instead of full type checking casts.
Small cleanups, plug some leaks.
Handle dynamic sources.
Add some helper functions to create lists of strings used for
blacklisting and other stuff.
Refactor some code dealing with analysing the source.
Re-enable sources without pads (like cd:// or other selfcontained
elements).
2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Set caps on outgoing buffers.
* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
(gst_video_rate_event), (gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
Fix videorate some more. Fixes #357977
2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
Original commit message from CVS:
* tests/check/elements/adder.c: (adder_suite):
Don't set timeout to 6 seconds when we're running
in valgrind ... (and how is 6 seconds longer than
the default anyway?)
2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
(gst_audio_rate_sink_event), (gst_audio_rate_convert),
(gst_audio_rate_convert_segments), (gst_audio_rate_chain):
Keep sink and src segment to keep track of time and support more
input formats.
Fix bogus next_offset and run_time calculation, don't understand how
this could have worked before. Fixes #357976.
Remove some unneeded vars.
2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (remove_sinks):
Only remove visualisation from visbin if there is a visbin (or:
don't throw warnings when closing totem without playing a file).
2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
add new header file to spec
Original commit message from CVS:
add new header file to spec
2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
Original commit message from CVS:
* tests/check/Makefile.am:
Another attempt to make the gen64 buildbot happy.
2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
Original commit message from CVS:
Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
* ext/libvisual/visual.c: (gst_visual_clear_actors),
(gst_visual_chain), (gst_visual_change_state):
Libvisual plugin was not passing audio data to libvisual 0.4.0
correctly. Fixes #357800
2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
Original commit message from CVS:
* tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
Add timeout to _get_state() so we see which pipeline it is
that causes trouble on the gen64 build bot.
2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
(gst_base_rtp_depayload_set_gst_timestamp):
the source pad always uses fixed caps.
2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
Added docs for the audio libs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
Added docs for the audio libs.
2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
Original commit message from CVS:
* tests/check/Makefile.am:
Temporarily disable test that fails on the bots for unknown reasons.
2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Moved AudioCodecType into priv
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
(add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
(is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
(new_pad):
Cleanups and small leak fixes.
Added Depayloaders to valid list of autopluggable elements.
2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property),
(gen_video_element), (gen_text_element), (gen_audio_element),
(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
(gst_play_bin_set_clock_func), (gst_play_bin_change_state):
Detect NO_PREROLL state change returns and disable clock distribution to
the sinks so that sync is disabled.
Avoid some type checking and do simple casts instead.
Small cleanups, fix some FIXMEs.
Be more robust when linking user specified elements, catch an report
errors. Fixes #357404.
Fix some leaks in the error paths.
2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
* ChangeLog:
ChangeLog surgery for missing bug-number
Original commit message from CVS:
ChangeLog surgery for missing bug-number
2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/playback/test.c:
Fix compilation with uClibc and -Werror (#357591).
2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse dates that are followed by a time as well (#357532).
* tests/check/libs/tag.c: (test_vorbis_tags):
Add unit test for this.
2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/: A few array const-ifications.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
* gst/videotestsrc/videotestsrc.h:
A few array const-ifications.
2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: See if this makes the build bots happy.
Original commit message from CVS:
* tests/check/Makefile.am:
See if this makes the build bots happy.
* tests/check/libs/cddabasesrc.c:
UTF8-ise my name.
2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_font),
(fix_invalid_entities):
More case-insensitivity for certain tags; recognise entities with
decimal codes as special entities as well (#357330).
2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
Original commit message from CVS:
* gst-libs/gst/Makefile.am:
Need to build tag directory before cdda.
2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_base_init):
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
(gst_tag_register_musicbrainz_tags):
Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
depend on libgsttag. This is required so we can extract/read tags like
DISCID without depending on libgstcddabasesrc (which used to register
them).
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
tags (also see #347848).
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
Log vorbis comments we are actually writing. Const-ify array.
2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
Improve buffering a bit by avoiding a deadlock because we cannot assume
the underrun is always called.
2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Added MPEG-4 AAC and id and caps. Fixes #357289
Added WMA9 Lossless id.
2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix misleading docs addition.
* tests/check/elements/videotestsrc.c: (check_rgb_buf):
Get rid of compiler warning the right way.
2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_process),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_queue_release):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Small cleanups.
Fix some leaks.
Refactored the process method and added methods to push from the process
vmethod.
Use _scale functions.
API: gst_base_rtp_depayload_push_ts
API: gst_base_rtp_depayload_push
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
timestamps are uint.
2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Remove unused statement from doc example.
2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/videorate/gstvideorate.c:
update docs
Original commit message from CVS:
update docs
2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
Original commit message from CVS:
* gst-libs/gst/interfaces/videoorientation.c:
(gst_video_orientation_iface_init),
(gst_video_orientation_get_hflip),
(gst_video_orientation_get_vflip),
(gst_video_orientation_get_hcenter),
(gst_video_orientation_get_vcenter),
(gst_video_orientation_set_hflip),
(gst_video_orientation_set_vflip),
(gst_video_orientation_set_hcenter),
(gst_video_orientation_set_vcenter):
Add since tags to new API docs, ChangeLog surgery (forgot API keyword
in ChangeLog)
2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: but disable for now since it doesn't pass (something wrong with
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
(create_rgb_conversions), (rgb_conversion_free),
(right_shift_colour), (fix_expected_colour), (check_rgb_buf),
(got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
but disable for now since it doesn't pass (something wrong with
RGBA somewhere).
2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
(queue_out_of_data), (gen_preroll_element),
(preroll_remove_overrun), (probe_triggered):
Refactor handling of overrun detection.
Separate handling of group completion and deadlock detection when doing
network buffering. This should fix some deadlocks that were not detected
because the group was completed.
Add more comments, improve debugging.
2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/: Some more compilation fixes.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/libs/audio.c:
Some more compilation fixes.
2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Early morning compilation fix.
2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
bump nano
Original commit message from CVS:
bump nano
2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/: Fix some warnings.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/multifdsink.c: (GST_START_TEST):
* tests/check/elements/videorate.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (eos_buffer_probe):
Fix some warnings.
2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_get_times):
change colorkey behaviour back according to #354773 comment 6/7
2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery: remove junk
Original commit message from CVS:
ChangeLog surgery: remove junk
2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
(gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
(gst_multi_fd_sink_recover_client),
(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Implement stubbed out properties unit-type, units-soft-max,
units-max, to allow specifying maximum sizes in units other than
buffers.
Fixes #355935
2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Reorder the audio formats a bit for clarity.
Detect and create caps for MSGSM and MSN (WAV49).
Fixes #356596.
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
Small cleanups, move error handling out of normal flow for clarity.
2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Add new interface to control video orientation (fixes #354908)
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/videoorientation.c:
(gst_video_orientation_get_type),
(gst_video_orientation_iface_init),
(gst_video_orientation_get_hflip),
(gst_video_orientation_get_vflip),
(gst_video_orientation_get_hcenter),
(gst_video_orientation_get_vcenter),
(gst_video_orientation_set_hflip),
(gst_video_orientation_set_vflip),
(gst_video_orientation_set_hcenter),
(gst_video_orientation_set_vcenter):
* gst-libs/gst/interfaces/videoorientation.h:
Add new interface to control video orientation (fixes #354908)
2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
Use G_UNLIKELY in _create and log one more detail.
(gst_video_test_src_get_times), (gst_video_test_src_create):
* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
Use gst_util_uint64_scale_int in _get_times().
2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
Give better warning message (add object and detail).
2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_get_times):
xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
#354773), use gst_util_uint64_scale_int in _get_times()
2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
always true, leading to dropping all timestamps.
2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/libvisual/visual.c: update to work also with libvisual 0.4 API
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_vis_src_negotiate),
(gst_visual_chain), (gst_visual_change_state):
update to work also with libvisual 0.4 API
* tools/gst-launch-ext.1.in:
* tools/gst-visualise.1.in:
remove references to old man-pages
* tests/examples/seek/seek.c: (main):
add real meadi-buttons, add tool-tips for the seek-options, arrange
seek options in a table
2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
(gst_ogg_mux_push_buffer):
Don't generate out-of-order timestamps from oggmux, instead clamp
output timestamps to be >= the previously output ts.
Fixes #355595
2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
(gst_multi_fd_sink_class_init):
Updates, fixes, and typo corrections for multifdsink. No functional
changes.
2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
Don't crash on truncated files - check that we got an 8 byte buffer
before trying to memcmp it.
2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (get_active_source):
Make stream-switching appear instant to the application
(ie. make sure that a g_object_get on 'current-foo' returns
the stream previously set with g_object_set(). Totem needs
this to update stream-related meta-info (like audio-codec)
correctly when switching streams.
2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
(gst_alsa_mixer_ensure_track_list):
Try harder to guess which mixer track is the master mixer
track (instead of just taking the first one that has a pvolume).
Fixes #342228.
2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(gst_audio_convert_transform_caps):
Get structure-name just once.
2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/: Fix big batch of compiler warnings.
Original commit message from CVS:
* tests/check/elements/audioresample.c: (GST_START_TEST):
* tests/check/elements/videotestsrc.c: (check_rgb_buf):
* tests/check/elements/volume.c: (GST_START_TEST):
* tests/check/elements/vorbisdec.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
(test_pipeline), (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
Fix big batch of compiler warnings.
2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Add docs about icydemux usage in connection with gnomevfssrc
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/audiorate/gstaudiorate.c:
More G_OBJECT macro fixing.
* gst/audiotestsrc/gstaudiotestsrc.h:
Fix wrong info in header due to copy & paste
2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Do the delay calculation in the source/sink base classes as this is
specific for the capture/playback mode.
Try to fixate a bit better, like round depth up to a multiple of 8
bigger than width.
Handle underruns correctly by marking DISCONT on buffers and adjusting
timestamps to handle the gap.
Set offset/offset_end correctly on buffers.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Remove resync and underrun recovery from the ringbuffer.
Fix ringbuffer read code on under/overrun.
2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (fill_buffer), (check_queue),
(queue_threshold_reached), (gst_play_base_bin_set_property),
(gst_play_base_bin_get_property):
* gst/playback/gstplaybasebin.h:
Don't use a 0 low watermark when buffering, it is catching starvation
way too late. Instead, use a 3 second queue with 30 and 95
percent low/high watermarks.
Added queue-min-threshold property to configure low watermark.
Use new _buffering message API.
Make queue_threshold variable big enough to store a uint64 time value.
API: playbin::queue-min-threshold property.
2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
configure.ac: We require 0.10.10.1 now because of _wait_preroll().
Original commit message from CVS:
* configure.ac:
We require 0.10.10.1 now because of _wait_preroll().
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Use gst_base_sink_wait_preroll().
2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/: Use DEBUG_OBJECT more.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
* ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
Use DEBUG_OBJECT more.
=== release 0.10.10 ===
2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* common:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/theora/theoraparse.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst/playback/gstplaybin.c:
* tests/check/Makefile.am:
* win32/common/config.h:
releasing 0.10.10
Original commit message from CVS:
releasing 0.10.10
2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
* win32/common/config.h:
second prerelease
Original commit message from CVS:
second prerelease
2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
update bug in changelog
Original commit message from CVS:
update bug in changelog
2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
Fix implementation of sync-method 'next-keyframe'
Original commit message from CVS:
patch by: Michael Smith <msmith at fluendo dot com>
* gst/tcp/gstmultifdsink.c: (is_sync_frame),
(gst_multi_fd_sink_client_queue_buffer),
(gst_multi_fd_sink_new_client):
* tests/check/elements/multifdsink.c: (GST_START_TEST),
(multifdsink_suite):
Fix implementation of sync-method 'next-keyframe'
2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
Original commit message from CVS:
patch by: Wim Taymans <wim at fluendo dot com>
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
This patch removes the RANDOM flag that was incorrectly introduced with
revision 1.91. Fixes #354590
2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
* win32/common/config.h:
first prerelease
Original commit message from CVS:
first prerelease
2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
update po files
Original commit message from CVS:
update po files
2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
Original commit message from CVS:
* tests/check/Makefile.am:
Random variation in Makefile line to see if it makes the
gen64-base-full bot any happier.
2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (oggmux_suite):
Disable test that fails at the moment (killed after timeout).
2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
tests/check/: Add simple unit test for oggmux from #337026 with checking for the
Original commit message from CVS:
Patch by: James Livingston <doclivingston at gmail.com>
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/oggmux.c: (get_page_codec),
(check_chain_final_state), (fail_if_audio), (validate_ogg_page),
(eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
(test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
(test_theora_vorbis), (oggmux_suite):
Add simple unit test for oggmux from #337026 with checking for the
EOS flags disabled for the time being.
2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
Original commit message from CVS:
patch by: Alessandro Dessina <alessandro nnva org>
* ext/ogg/gstoggmux.c:
Add cmml caps to oggmux. Fixes #353912
2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
Original commit message from CVS:
* tests/check/elements/videotestsrc.c: (check_rgb_buf):
Returning a return value often helps. In this case, we
don't need the return value anyway, so just get rid of it.
Should make build bots much happier.
2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
(paint_get_structure), (gst_video_test_src_get_size),
(gst_video_test_src_smpte), (gst_video_test_src_snow),
(gst_video_test_src_unicolor), (paint_setup_AYUV),
(paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
(paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add support for AYUV and the various RGBA formats. Initialise
fields of paintinfo structs allocated on the stack.
* tests/check/elements/videotestsrc.c: (right_shift_colour),
(fix_expected_colour), (check_rgb_buf), (got_buf_cb),
(GST_START_TEST), (videotestsrc_suite):
Add unit tests for videotestsrc's RGB output.
2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_pattern_get_type),
(gst_video_test_src_set_pattern):
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
(gst_video_test_src_black), (gst_video_test_src_white),
(gst_video_test_src_red), (gst_video_test_src_green),
(gst_video_test_src_blue):
* gst/videotestsrc/videotestsrc.h:
Add more uni-colour patterns ("white", "red", "green", and "blue").
2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
Fix stride for YVYU, should be word-aligned (#353658).
2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/adder/gstadder.c: Fix build.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_src_event):
Fix build.
2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
Original commit message from CVS:
* gst/adder/gstadder.c: (forward_event_func),
(gst_adder_src_event), (gst_adder_collected),
(gst_adder_change_state):
* gst/adder/gstadder.h:
Remember the start position asked in the incoming seeks, so we can
output GST_EVENT_NEW_SEGMENT with a correct position value (instead
of assuming it will always be 0).
2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
(gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_loop):
Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
Return FALSE instead of returning a random false unit
size when the format isn't known/supported (even if
this shouldn't happen under normal circumstances).
2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
Original commit message from CVS:
Patch by: Tim-Philipp Müller <tim at centricular dot net>
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
(gst_gnome_vfs_src_start):
Try harder to get the size from a uri by using _info_uri() when
_info_from_handle() does not give us enough info.
Also follow symlinks when getting the size.
Partially Fixes #332864.
2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
Original commit message from CVS:
Patch by: Viktor Peters <viktor dot peters at gmail dot com>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
(gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
(gst_alsa_mixer_set_record):
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities),
(alsa_track_has_cap), (gst_alsa_mixer_track_new),
(gst_alsa_mixer_track_update):
* ext/alsa/gstalsamixertrack.h:
Improve and fix mixer track handling, in particular better handling
of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
track objects for tracks that have both capture and playback volume
(and label them differently as well so they're not mistakenly
assumed to be duplicates); classify mixer tracks that only affect
the audible volume of something (rather than the capture volume)
as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
for capture tracks to correspond to alsa-pswitch alsa-cswitch
(following the meaning documented in the mixer interface header
file); add support for alsa's exclusive cswitch groups; update/sync
state/flags better if mixer settings are changed by another
application. Fixes #336075.
2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Improve docs: add section about BUFFERING messages sent by playbin.
2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
(gst_vorbis_enc_buffer_check_discontinuous),
(gst_vorbis_enc_chain):
Ignore explicit DISCONT marked on buffers (which is often spurious,
particularly when using multiple segments), in favour of solely
using the timestamps/durations.
2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Don't rely on incoming buffers offset anymore, since it is completely
broken when using multiple segments.
Instead convert the incoming buffers timestamp to running time, and
then convert that value to the offsets.
Also inform GstSegment of the last outputted stop position, which is
needed if we received several segments with an unknown stop value.
2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
fix buffer unreffing on a header push failure
2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
(gst_audio_rate_chain):
Make the metadata of the buffer writable before changing its
flags.
2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
* ChangeLog:
Fix changelog with bugzilla bug it fixed.
Original commit message from CVS:
Fix changelog with bugzilla bug it fixed.
2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audiorate/gstaudiorate.c: Fix audiorate some more.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
(gst_audio_rate_setcaps), (gst_audio_rate_init),
(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
(gst_audio_rate_chain), (gst_audio_rate_change_state):
Fix audiorate some more.
Reset and resync counters on flush and READY.
Handle the DISCONT flag correctly.
Use GstSegment to track position.
Fail when not negotiated.
2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
gst/tcp/gstmultifdsink.c: Fix spelling.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Fix spelling.
Remove accidently included debug line.
2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstmultifdsink.c: Small cleanups.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Small cleanups.
If a buffer is received with no caps, make the buffer metadata
writable and set the caps, making sure that we don't screw up the
refcounts.
2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
Fix memory leaks and misleading debug messages, add a couple of
comments.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
(gst_multi_fd_sink_render):
Do not use gst_buffer_make_writable() in a basesink render method,
as it may incorrectly unref the buffer. Instead, use convoluted
dance to avoid copying the buffer except when we need to.
2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c:
(gst_vorbis_enc_buffer_check_discontinuous):
Allow very small discontinuities in the timestamps. These we can't
do anything useful with anyway (because vorbis's timestamps have
only sample granularity), and are commonly produced by elements with
minor bugs. Allow up to 1/2 a sample out.
Fixes #351742.
2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
Original commit message from CVS:
* tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
(play_scrub_toggle_cb), (main):
Add a checkbox to enable play scrubbing. Makes it possible to disable
normal scrubbing.
2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/.cvsignore: make buildbot happy
Original commit message from CVS:
* tests/check/elements/.cvsignore:
make buildbot happy
2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
(gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
(gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
(gst_ogm_text_parse_strip_trailing_zeroes),
(gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
(gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
Refactor ogm parse, do better input checking, misc. clean-ups.
Cache incoming events and push them once the source pad has
been created. Don't pass unterminated strings to sscanf().
Strip trailing zeroes from subtitle text output, since they
are not valid UTF-8. Don't push vorbiscomment packets on
the subtitle text pad. Output perfect streams if possible.
2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
Original commit message from CVS:
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
Waits for tasks to settle down so that we clean up correctly for
valgrind.
2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
Original commit message from CVS:
* tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
Unit test fixes: \377 is more likely to fit into 8 bits than \777;
actually return return value in taglists_are_equal.
2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
Fix crash due to broken bitstream parsing on x86-64: can't make
any assumptions about sizeof(struct) due to alignment/packing
differences on different architectures. Fixes #351790.
2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
(gst_riff_parse_chunk), (gst_riff_parse_file_header),
(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
(gst_riff_parse_info):
Protect public functions against bad input.
Do some cleanups.
Fix documentation.
2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add voxware audio IDs (even if we can't play it) (#351795).
2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps),
(gst_riff_create_iavs_template_caps):
Const-ify some arrays and use G_N_ELEMENTS instead
of wasting oodles of RAM on terminator bits.
2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
* tests/check/libs/tag.c: (GST_START_TEST):
And the same for _to_vorbiscomment_buffer(): allow
id_data_len == 0 for speex.
2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/gdp/README:
adding a README
Original commit message from CVS:
adding a README
2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
Move GDP plugin to -base from -bad. Closes #347783.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/inspect/plugin-gdp.xml:
* gst/gdp/Makefile.am:
* tests/check/Makefile.am:
Move GDP plugin to -base from -bad. Closes #347783.
2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer):
Allow id_data_len == 0 (needed for vorbis comments in Speex files).
Also add some checks to make sure we don't memcmp() beyond the end of
vorbiscomment buffer if the ID to check for is larger than the buffer.
* tests/check/libs/tag.c: (GST_START_TEST):
Some more tests for gst_tag_list_from_vorbiscomment_buffer().
2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
(gst_vorbis_enc_set_metadata):
Use vorbis comment utility functions from libgsttag
instead of re-inventing the wheel (partially fixes #347091).
2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix leaks. Wait for state transitions that might happen ASYNC, as well
as some that won't.
2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Don't try to GObject scan the netbuffer as it's not a GObject.
Fixes #351308.
* gst-libs/gst/netbuffer/gstnetbuffer.c:
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Document GstNetBuffer.
2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Add testcase for caps-size-explosion
2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_unit_size), (set_structure_widths):
Lower debug, use g_assert in _get_unit_size
* gst/audioresample/gstaudioresample.c:
(audioresample_get_unit_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
use g_assert in _get_unit_size
2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery: fix bug number
Original commit message from CVS:
ChangeLog surgery: fix bug number
2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
Document GstRTPBuffer.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
(gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_buffer):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Document GstRTPBuffer.
Added function to efficiently strip payload headers.
API: gst_rtp_buffer_get_payload_subbuffer()
2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
(gst_tag_to_vorbis_comments):
Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
tags and deserialise them properly as well (#351768).
Add some more gtk-doc blurbs and also some g_return_if_fail().
* tests/check/libs/tag.c: (GST_START_TEST),
(back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
More tests.
2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
Original commit message from CVS:
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
(gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
(gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
(gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
(gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
(gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
Added ogg-in-avi parser element. Fixes #140139.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
Fixed a bug in oggdemux debug code.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Recognise Ogg in the AVI extensible wave format.
2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Make buffer durations add up (duration should be next_ts-ts for
perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
from CVS.
* tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
(test_buffer_timestamps), (cddabasesrc_suite):
Add unit test for the above.
* tests/check/Makefile.am:
Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
to see what happens.
2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/: Avoid setting and using a NULL device name.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
(gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
(gst_alsasrc_open):
Avoid setting and using a NULL device name.
Print more info when we fail to open a device.
2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
API: add gst_tag_parse_extended_comment() (#351426).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
API: add gst_tag_parse_extended_comment() (#351426).
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
Add unit test for gst_tag_parse_extended_comment().
2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
sys/: Fix leak (#351502).
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
Fix leak (#351502).
2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
Document playbin.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* gst/playback/gstplaybin.c:
Document playbin.
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update to CVS version.
2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(value_list_append_structure_list),
(gst_play_bin_handle_redirect_message),
(gst_play_bin_handle_message):
Add "connection-speed" property; re-order redirect messages with
multiple redirect locations depending on the minimum bitrate if
that information is available and a connection speed is set
(#350399).
2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Update max volume to the same value that the volume element uses.
2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsamixer.c: Less uglyness..
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
Less uglyness..
2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Add some more debug info.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
(gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
Add some more debug info.
Don't crash when a seek failed.
Actually return the result of the seek instead of TRUE.
Ignore multiple BOS pages with the same serial so that we don't create
the same stream multiple times.
Post an error when we fail to do the initial seek.
2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsa.c: Small code cleanup.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
Small code cleanup.
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
(gst_alsa_mixer_new):
Remove hack that always set the device to hw:0*.
Properly find the card name for whatever device was configured.
Do some better debugging.
Fixes #350784.
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_change_state):
Cleanups.
Handle setting of a NULL device name better.
2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
Original commit message from CVS:
* gst/adder/gstadder.c:
Don't clip float values. Fixes #350900.
2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
gst/tcp/gsttcp.c: Really fix the build?
Original commit message from CVS:
2006-08-11 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcp.c: Really fix the build?
2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
Original commit message from CVS:
2006-08-11 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcp.h: For now, always disable deprecation here --
fixes the build.
2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
Float caps shouldn't have a "signed" field.
2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
Implement SEEKING query in its most basic form, so that we can
at least check if we're seekable or not (#350655).
2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
The checks here are not even close to anything that would
justify MAXIMUM probability, lowering to POSSIBLE until someone
fixes the checks (case at hand: quicktime redirection files
might start with 00 00 01 XX and pass the checks here just
fine, see #350399).
2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
I forgot to include the file containing the #define :)
Now includes "config.h"
2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
Ignore test known to fail on PPC64. See #348114.
2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon net>
* gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
Better detection for multipart/x-mixed-replace: accept leading
whitespaces before the boundary marker as well (as our very own
multipartmux used to produce) (#349068).
2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
gst-libs/gst/riff/: Detect DTS audio streams (#350157).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Detect DTS audio streams (#350157).
2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
Original commit message from CVS:
2006-08-05 Andy Wingo <wingo@pobox.com>
* ext/theora/gsttheoraparse.h:
* ext/theora/theoraparse.c (gst_theora_parse_class_init)
(theora_parse_dispose, theora_parse_set_property)
(theora_parse_get_property, theora_parse_munge_granulepos)
(theora_parse_push_buffer, theora_parse_change_state): Add a
property 'synchronization-points' to fix badly synchronized oggs.
2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
Original commit message from CVS:
2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Fix event parsing by gdpdepay. Fixes #349916.
2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add a few tests for the channel position stuff in libgstaudio.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/audio.c: (structure_contains_channel_positions),
(fixed_caps_have_channel_positions), (GST_START_TEST),
(audio_suite), (main):
Add a few tests for the channel position stuff in libgstaudio.
2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
(gst_alsa_detect_channels):
* ext/alsa/gstalsasink.c:
Add support for cards that (only) do more than 8 channels,
like the Delta 44 (#345188).
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
unspecified channel position and cannot be combined with any
of the other audio channel positions; adjust position layout
checks accordingly (#345188).
2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Recognise ancient RealAudio files (see #349779).
2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for Interplay's MVE format (#348973).
2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
Original commit message from CVS:
Patch by: Marcel Moreaux <marcelm at luon dot net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_add_to_queue):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Handle RTP sequence number rollover.
Disable jitterbuffer by default.
2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/gdp/gstgdpdepay.c: Disable seeking.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
(gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
(gst_gdp_depay_src_event), (gst_gdp_depay_chain),
(gst_gdp_depay_change_state):
Disable seeking.
Small cleanups.
Clear adapter on disconts.
Clear caps when going to READY instead of NULL
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
(gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
(gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
(gst_gdp_queue_buffer), (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
(gst_gdp_pay_change_state):
* gst/gdp/gstgdppay.h:
Reset payloader when going to READY.
Fix leaked buffers in ->queue on push errors.
Disable seeking.
Code cleanups.
Create packetizer in _init, free in _finalize.
2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
(gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
Consume all events except EOS because we generate events from
the gdp payload instead. Fixes #349204
2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (audioresample_stop),
(audioresample_set_caps):
Don't leak references to the incoming caps. Clean them up when
stopping.
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_finalize):
Don't leak our temporary pixel buffer.
* tests/check/Makefile.am:
* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
(GST_START_TEST), (simple_launch_lines_suite):
Fix leaks and re-enable the test for valgrind checking.
2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon net>
* gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
(plugin_init):
Add typefind function for multipart/x-mixed-replace (#348916).
2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.c: Fix leak in duration query.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_query_duration):
Fix leak in duration query.
Reflow some docs and notes.
2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
Original commit message from CVS:
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
(vorbisenc_suite):
Enable Andy's extra vorbisenc test, now that it passes. Also fix one
aspect of it.
2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
(gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
(gst_vorbis_enc_push_buffer),
(gst_vorbis_enc_buffer_check_discontinuous),
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Handle discontinuities in the input vorbis stream correctly,
so that the output is properly timestamped (and has good granulepos
values). Needs some oggmux fixes too.
2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
Original commit message from CVS:
patch by: Kai Vehmanen <kv2004 eca cx>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_change_state):
Don't send multiple newsegments with different formats.
Fixes #348677.
2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
Make seeking in ogg more accurate again by doing the more correct
granuletime to stream time conversion.
2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_new_client):
debug a little more understandably
do not use goto as a substitute for break, especially if
break is also being used
2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gsttcp.c:
move a recurring normal event to LOG, where it should be
Original commit message from CVS:
move a recurring normal event to LOG, where it should be
2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/vorbis/vorbisdec.c:
tweak debug output
Original commit message from CVS:
tweak debug output
2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
proxying get/set caps is the wrong thing to do, since we really
do change caps quite fundamentally
* tests/check/elements/gdpdepay.c:
* tests/check/elements/gdppay.c:
remove declaration of buffers, it's already done in gstcheck.h
2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Remove GLib-2.6 compatibility cruft.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Remove GLib-2.6 compatibility cruft.
2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to align a sample to an unknown value.
2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
When the audio clock is slaved to another clock, never try to align
samples but trust the rate interpolation algorithm.
2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
Don't try to calculate silence samples, base class does this much
better now.
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
Calculate silence samples correctly.
* gst-libs/gst/audio/gstringbuffer.h:
Add _CAST macro.
2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
Limit search for the first markup tag to the first few kB of
the file. If we don't find one there, it's highly unlikely that
this is an XML(-ish) file.
2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
Original commit message from CVS:
2006-07-21 Andy Wingo <wingo@pobox.com>
* tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
test to the one in vorbisenc. Also commented out.
2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
Original commit message from CVS:
2006-07-21 Andy Wingo <wingo@pobox.com>
* tests/check/pipelines/vorbisenc.c:
(test_discontinuity): New test, commented out until Mike lands
some elite vorbisenc patches.
2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
tests/check/pipelines/: Port to bufferstraw.
Original commit message from CVS:
2006-07-21 Andy Wingo <wingo@pobox.com>
* tests/check/pipelines/vorbisenc.c:
* tests/check/pipelines/theoraenc.c: Port to bufferstraw.
Bufferstraw was actually factored out of these tests. Now we share
code yay.
2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Better clipping.
Original commit message from CVS:
* ext/theora/theoradec.c: (clip_buffer):
Better clipping.
2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Fix leak.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
Fix leak.
Avoid type casting when we can.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
Fix mem leak.
2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_change_state):
Make state change fail if the specified device can't be opened
for some reason.
2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/test.c: Example of a small audio/video player using decodebin.
Original commit message from CVS:
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad), (main):
Example of a small audio/video player using decodebin.
2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add 'fact' chunk id
2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_change_state):
Don't assert when not negotiated but post a meaningfull
error message. Fixes #347918.
* gst-libs/gst/rtp/gstbasertppayload.c:
Add comment about better default MTU size.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
Small cleanups, start docs.
2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
Original commit message from CVS:
Patch by: Martin Szulecki
* sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
If "device-name" is requested and the device is not
open, try to temporarily open it to obtain this
information (#342494).
2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
Some more random const-ifications.
2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps):
Add more FOURCCs (sort list to make stuff easier to find),
add comment what those 16 bytes in struct _gst_riff_strh according to
one avi-dumper are
2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
Original commit message from CVS:
2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
remove parent_class setting, BOILERPLATE does this
(gst_gdp_pay_reset_streamheader):
fix typo in comment
2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_fixate_channel_positions):
Const-ify two arrays.
2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
Fix typo, so that alsasink also advertises 8 channels
if that's supported (tags: can, worms, open, alsa, ph34r).
2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
*sigh*, when is the compiler going to warn when the comments
are out-of-sync with the code.. Refix case of busted theora
headers with 0 granule pos.
2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_wait),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
Fix 99% cpu load by waiting for absolute times on the
clock. Fixes #347300.
2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
Original commit message from CVS:
2006-07-14 Andy Wingo <wingo@pobox.com>
* ext/theora/gsttheoraparse.h:
* ext/theora/theoraparse.c (theora_parse_drain_event_queue)
(theora_parse_push_headers, theora_parse_clear_queue)
(theora_parse_drain_queue_prematurely, )
(theora_parse_sink_event, theora_parse_change_state): Queue events
until we initialized our state, like in vorbisparse.
2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
Original commit message from CVS:
2006-07-14 Andy Wingo <wingo@pobox.com>
* ext/vorbis/vorbisparse.h:
* ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
(vorbis_parse_push_headers, vorbis_parse_clear_queue)
(vorbis_parse_drain_queue_prematurely, )
(vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
until we have initialized our state. Fixes seeking after an
initial pad block.
2006-07-14 Andy Wingo <wingo@pobox.com>
Patch by: Iain * <iaingnome@gmail.com>
* ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Bump nano back to CVS
Original commit message from CVS:
* configure.ac:
Bump nano back to CVS
=== release 0.10.9 ===
2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: releasing 0.10.9, "I walk the line"
Original commit message from CVS:
2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:
releasing 0.10.9, "I walk the line"
2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
Original commit message from CVS:
* tests/check/pipelines/vorbisenc.c: (stop_pipeline):
Move a g_cond_signal to earlier to avoid sometimes deadlocking
(commonly happens when running this test under valgrind) when trying
to remove the buffer probe.
2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/gdp/Makefile.am:
build as a plugin, not a lib
Original commit message from CVS:
build as a plugin, not a lib
2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
Fix missing g_unlock from the previous commit
2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
(gst_ximagesink_change_state):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_change_state):
Implement a locking order to ensure we always take the object lock
before the x_lock and never vice-versa.
2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
docs/plugins/: add more plugins and elements to docs
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
add more plugins and elements to docs
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
fix segfaults due to wrong g_free
add example
* gst/gdp/gstgdppay.c:
add example
2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (find_compatibles):
Fix a caps leak when linking (#347304)
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
Don't leak shared memory resources. Use the object lock to protect
against the xcontext disappearing while returning a buffer from the
pipeline. (#347304)
2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
(vorbis_handle_comment_packet):
gst_tag_list_merge() returns a new object. Take that into account when
using it. This avoids memleak.
Revert previous commit which is not needed.
2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
Reset the decoder in finalize so that all fields get cleared.
2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_clock),
(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
Don't try to post an error message when setting the clock fails
as this can happen when adding an element to a bin which will then
deadlock. Fixes #347296.
2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
(vorbis_handle_type_packet):
Post tag messages on the bus even if we're not initialized.
If we're not initialized, we still postpone the event pushing of tags.
2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
Revert last two changes that broke the freeze.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Revert last two changes that broke the freeze.
2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
basesink calculates silence sample correctly for us.
2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Calculate correct silence samples so we don't fill our ringbuffer
with noise.
2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
(gst_vorbis_dec_reset), (vorbis_dec_sink_event),
(vorbis_handle_comment_packet), (vorbis_handle_type_packet):
* ext/vorbis/vorbisdec.h:
Delay sending events (newsegment, tags) until the decoder is properly
initialized.
Fixes #347295
2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (get_float_mc_caps),
(get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
Patch from #347221 adding a test for audioconvert
channel remappings.
2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
Original commit message from CVS:
* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
(gst_ssa_parse_parse_line):
Don't include the terminating NUL in the buffer size,
it's only there for extra paranoia (would add random
'*' characters at the end of each subtitle since the
terminator itself is not valid UTF-8 technically).
Also fix indenting after boilerplate macro.
2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
Also emit 'unknown-type' signal (which should really be
called unhandled-type) if we found potential decoders/demuxers
in the registry but none of them worked in the end (as in the
case where the plugins don't exist any longer but are still
listed in the registry). Fixes #329798.
2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* ext/theora/theoraparse.c:
theoraparse.c (theora_parse_push_buffer)
Original commit message from CVS:
2006-07-08 Andy Wingo <wingo@pobox.com>
* theoraparse.c (theora_parse_push_buffer)
(theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
Add some more debugging. Fix granulepos reconstruction in the face
of discontinuities.
2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init),
(gst_base_audio_sink_provide_clock):
Use gobject_class instead of G_OBJECT_CLASS (klass)
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
(gst_base_audio_src_get_time),
(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
(gst_base_audio_src_create_ringbuffer):
Fix latency and buffer-time constants and properties ala basesink.
Implement pull based scheduling. Fixes #346527.
Set default blocksize in GstBaseSrc to 0, we default to pushing out
one segment.
Refuse slaving to another clock instead of silently not working.
Only provide a clock when we are actually able to do so.
Various small cleanups and compiler hints.
2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
Original commit message from CVS:
Patch by: Lutz Mueller <lutz at topfrose de>
* gst/typefind/gsttypefindfunctions.c: (html_type_find),
(plugin_init):
Add typefinding for text/html (#346581).
2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(xml_check_first_element), (xml_type_find), (smil_type_find):
Fix SMIL typefinding, make xml_check_first_element() more
useful.
2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
(gst_play_base_bin_finalize), (decodebin_element_added_cb),
(decodebin_element_removed_cb), (gst_play_base_bin_set_property):
* gst/playback/gstplaybasebin.h:
Protect list of elements with a subtitle-encoding property and
the subtitle encoding member itself with a lock of their own
instead of using the object lock. This prevents a dead-lock in
the element-remove callback in some circumstances when shutting
down playbin.
2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/common/libgsttag.def: Export some new functions.
Original commit message from CVS:
* win32/common/libgsttag.def:
Export some new functions.
* win32/vs6/libgstogg.dsp:
Add a link to libgsttag-0.10.lib.
2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsamixertrack.c: Some const-ification.
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Some const-ification.
2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
Improve checking if we are dealing with a stream. Added some
more uris that need buffering.
2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
ext/vorbis/vorbisdec.c: Remove unused variable.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_do_clip):
Remove unused variable.
2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
Makefile.am: include lcov.mak
Original commit message from CVS:
* Makefile.am:
include lcov.mak
* configure.ac:
add GCOV_LIBS to GST_LIBS
2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
Original commit message from CVS:
Patch by: Michael Sheldon <webmaster at mikeasoft com>
* ext/alsa/gstalsasrc.c:
Add 32 bps to template caps and increase channels range
from [1,2] to [1,MAX]. See #346326.
2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Recognise 'WMVA' video codec fourcc (#345879).
2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Fixed nasty memory leak
2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/gsttcp.c: fix logging
Original commit message from CVS:
* gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
(gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
fix logging
2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
(remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
Protect remove_fakesink using a mutex, so that we don't try and
remove the fakesink simultaneously from multiple threads.
When going from READY to PAUSED, restore the fakesink, so that
it is there when decodebin gets reused.
2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/videorate/gstvideorate.c:
* gst/videotestsrc/gstvideotestsrc.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lsrc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
Original commit message from CVS:
* ext/directfb/dfbvideosink.c:
* ext/gsm/gstgsmdec.c:
* ext/gsm/gstgsmenc.c:
* ext/libmms/gstmms.c:
* ext/neon/gstneonhttpsrc.c:
* ext/theora/theoradec.c:
* gst/freeze/gstfreeze.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* sys/glsink/glimagesink.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
and fix one GObject boilerplate macro.
2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Second field in GEnumValue shouldn't be a description,
but a stringified version of the enum value.
2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Avoid type checking in buffer casts.
Avoid caps copy in buffer_alloc when we can.
Use pad_peer_accept.
2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
Oops, make that 'Since: 0.10.9'.
2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
API: add GstTagImageType enum to describe images contained in image tags (#345641).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type):
API: add GstTagImageType enum to describe images contained
in image tags (#345641).
2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Fix warnings with gst-inspect: "buffers-min" property
should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
typo in property description.
2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
gst/: Avoid unnecessary class cast check in class_init functions (#337747).
Original commit message from CVS:
Patch by: Cody Russell <bratsche at gnome org>
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
Avoid unnecessary class cast check in class_init
functions (#337747).
2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
(gst_text_overlay_video_chain):
g_markup_escape_text() REALLY doesn't like non-UTF8 input
and doesn't validate its input either (and neither did
textoverlay it seems). Let's do that then and fix #345206.
2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
(gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
(find_syncframe), (find_limits), (assign_value),
(count_burst_unit), (gst_multi_fd_sink_new_client),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
(gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
(gst_multi_fd_sink_change_state):
* gst/tcp/gstmultifdsink.h:
Added shiny new burst-on-connect methods.
Add properties to control the minimal amount of data queued.
Small cleanups.
API: bytes-min property
API: time-min property
API: buffers-min property
API: burst-unit property
API: burst-value property
API: add-full signal
* gst/tcp/gsttcp-marshal.list:
Added new marshaller code for the new signal.
* tests/check/elements/multifdsink.c: (GST_START_TEST),
(multifdsink_suite):
Added testcases for new burst methods.
2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update for latest changes
Original commit message from CVS:
update for latest changes
2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
ext/theora/theoradec.c: Implement clipping for accurate seeking.
Original commit message from CVS:
* ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
Implement clipping for accurate seeking.
Closes #345225
2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
Original commit message from CVS:
Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
* gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
(gst_video_scale_transform):
Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery
Original commit message from CVS:
ChangeLog surgery
2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
Original commit message from CVS:
* configure.ac:
Fix --disable-extern (can't set conditionals conditionally,
#343602).
2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
Original commit message from CVS:
* tests/check/elements/audioresample.c: (test_reuse),
(audioresample_suite):
Add test case for bug #342789 fixed below.
2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
Parse extra data better, apparently it's right behind
the normal strf header size. Fixes #343500.
2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams):
If we fail to set the buffer_time and period_time alsa
parameters, post a warning and leave alsa select a
default instead of failing. Fixes #342085
2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
Original commit message from CVS:
ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
out in the header file and shouldn't be listed in the docs.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
Fix it so that it doesn't crash in the debug statement.
2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/: add remaining symbols into correct setions
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
add remaining symbols into correct setions
* gst-libs/gst/audio/gstringbuffer.c:
fix incomplete docs
* gst-libs/gst/audio/gstringbuffer.h:
comment out not yet implemented function
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
add short descriptions
* gst-libs/gst/interfaces/propertyprobe.c:
fix return value docs
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
simplify debug logging
* gst-libs/gst/riff/riff-read.h:
sync function prototype and docs
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
remove left over symbol
2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
Use GST_PLUGIN_DOCS macro in configure.ac, add
Original commit message from CVS:
* autogen.sh:
* configure.ac:
* docs/Makefile.am:
Use GST_PLUGIN_DOCS macro in configure.ac, add
--enable-plugin-docs default to autogen.sh and use
ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
(gst_ogg_demux_loop):
Combine GstFlowReturn from the source pads to give a
meaningfull result to the upstream peer or to stop the
processing task in case of errors.
2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Try GST_TAG_CODEC as fallback when extracting the
codec name; more debug info.
2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
Original commit message from CVS:
* ext/ogg/Makefile.am:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
Extract language tags from ogm subtitle streams, so that
the subtitle menu choices are labelled correctly in
Totem (fixes #344708).
2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
(gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
(gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
Fix various leaks. Fixes #343699.
Add x-smoke mime type.
2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add IDs for 'bext' chunks (see #343837).
2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (sami_context_pop_state),
(handle_start_font), (end_sami_element):
Honour font face tags in SAMI subtitles (#344503).
2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
po/POTFILES.in: add missing files containing translatable strings
Original commit message from CVS:
* po/POTFILES.in:
add missing files containing translatable strings
2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
Original commit message from CVS:
* docs/libs/tmpl/.cvsignore:
we don't want those *.sgml files in CVS either
2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ignore more
Original commit message from CVS:
* docs/libs/.cvsignore:
* tests/check/elements/.cvsignore:
* tests/check/libs/.cvsignore:
ignore more
2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
Original commit message from CVS:
* docs/libs/Makefile.am:
also commiting the changed Makefile.am (added more libs to the
doc-build)
2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
docs/libs/: first batch of reordering things, add index & hierarchy
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
first batch of reordering things, add index & hierarchy
2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* sys/v4l/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
further clean up build
Original commit message from CVS:
further clean up build
2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
Original commit message from CVS:
* configure.ac:
use GST_PKG_CHECK_MODULES, cleans up output
2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* win32/common/config.h:
update to cvs
Original commit message from CVS:
update to cvs
2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
Add support for burn:// URIs (#343385); const-ify things a bit,
use G_N_ELEMENTS instead of hard-coded array size.
2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
Fix up broken entities before passing them to libxml *sigh*.
(#343303).
2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* configure.ac:
back to trunk
Original commit message from CVS:
back to trunk
=== release 0.10.8 ===
2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/config.h:
releasing 0.10.8
Original commit message from CVS:
releasing 0.10.8
2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
0.10.7.2 prerelease
Original commit message from CVS:
* configure.ac:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* win32/common/config.h:
0.10.7.2 prerelease
2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
move last template doc snippets to source code and delete them
Original commit message from CVS:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* docs/libs/tmpl/gsttuner.sgml:
* docs/libs/tmpl/gstxoverlay.sgml:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/xoverlay.c:
move last template doc snippets to source code and delete them
2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/gdp/gstgdppay.c:
adapt to new api
Original commit message from CVS:
adapt to new api
2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
configure.ac: enable building of GDP elements
Original commit message from CVS:
* configure.ac:
enable building of GDP elements
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
(gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
(gst_gdp_pay_change_state):
* gst/gdp/gstgdppay.h:
add version 1.0
2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
Original commit message from CVS:
* ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
(theora_parse_drain_queue):
Mark DELTA_UNIT on non-keyframes.
2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps):
* tests/check/elements/audioresample.c:
* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
* tests/check/elements/videorate.c:
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/elements/volume.c:
* tests/check/elements/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
Don't busy-wait in tests; this was causing test timeouts very
frequently when running under valgrind.
2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.h:
small fixes
Original commit message from CVS:
small fixes
2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
fail_if_can_read is racy
Original commit message from CVS:
fail_if_can_read is racy
2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/: make multifdsink properly deal with streamheader:
Original commit message from CVS:
* gst/tcp/README:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_client_queue_caps),
(gst_multi_fd_sink_client_queue_buffer),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_render):
* gst/tcp/gstmultifdsink.h:
make multifdsink properly deal with streamheader:
- streamheader is taken from caps
- buffers marked with IN_CAPS are not sent
- streamheaders are sent, on connection, from the caps of the
buffer where the client gets positioned to
- further streamheader changes are done every time the client
will receive a buffer with different caps
* tests/check/elements/multifdsink.c: (GST_START_TEST),
(gst_multifdsink_create_streamheader):
add tests for this
2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Reinstate limit on channel count. Vorbis does not define the meaning
of > 6 channels, so they're just independent channels. Gstreamer
currently has no mechanism to represent N independent channels.
2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Don't arbitrarily restrict channel counts and rate in vorbis.
In terms of effects likely on real-world files, this fixes 96kHz
playback of vorbis.
2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
gst/audioconvert/audioconvert.c: More correct float->int conversion.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float):
More correct float->int conversion.
2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
value. Fixes g-critical on trying to play back ogg containing
unknown codec.
2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_create), (group_commit),
(setup_source):
* gst/playback/gstplaybasebin.h:
Make the subtitle detection work from any thread so we don't
deadlock. Fixes #343397.
2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
(gst_gdp_pay_get_property):
add crc-header and crc-payload properties
don't error out on some things that are recoverable
* tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
add test for crc
2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gsttcp.c:
show type number when packet is of the wrong type
Original commit message from CVS:
show type number when packet is of the wrong type
2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
Original commit message from CVS:
* gst/volume/Makefile.am:
Seriously, it's not *that* hard to get compilation right. Even
a drunk can do it ! Add LIBOIL CFLAGS and LIBS
2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.h:
* ext/amrwb/gstamrwbdec.h:
* ext/amrwb/gstamrwbenc.h:
* ext/amrwb/gstamrwbparse.h:
* ext/arts/gst_arts.h:
* ext/artsd/gstartsdsink.h:
* ext/audiofile/gstafparse.h:
* ext/audiofile/gstafsink.h:
* ext/audiofile/gstafsrc.h:
* ext/audioresample/gstaudioresample.h:
* ext/bz2/gstbz2dec.h:
* ext/bz2/gstbz2enc.h:
* ext/dirac/gstdiracdec.h:
* ext/directfb/dfbvideosink.h:
* ext/divx/gstdivxdec.h:
* ext/divx/gstdivxenc.h:
* ext/dts/gstdtsdec.h:
* ext/faac/gstfaac.h:
* ext/gsm/gstgsmdec.h:
* ext/gsm/gstgsmenc.h:
* ext/ivorbis/vorbisenc.h:
* ext/libfame/gstlibfame.h:
* ext/nas/nassink.h:
* ext/neon/gstneonhttpsrc.h:
* ext/polyp/polypsink.h:
* ext/sdl/sdlaudiosink.h:
* ext/sdl/sdlvideosink.h:
* ext/shout/gstshout.h:
* ext/snapshot/gstsnapshot.h:
* ext/sndfile/gstsf.h:
* ext/swfdec/gstswfdec.h:
* ext/tarkin/gsttarkindec.h:
* ext/tarkin/gsttarkinenc.h:
* ext/theora/theoradec.h:
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackparse.h:
* ext/xine/gstxine.h:
* ext/xvid/gstxviddec.h:
* ext/xvid/gstxvidenc.h:
* gst/cdxaparse/gstcdxaparse.h:
* gst/cdxaparse/gstcdxastrip.h:
* gst/colorspace/gstcolorspace.h:
* gst/festival/gstfestival.h:
* gst/freeze/gstfreeze.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/modplug/gstmodplug.h:
* gst/mpeg1sys/gstmpeg1systemencode.h:
* gst/mpeg1videoparse/gstmp1videoparse.h:
* gst/mpeg2sub/gstmpeg2subt.h:
* gst/mpegaudioparse/gstmpegaudioparse.h:
* gst/multifilesink/gstmultifilesink.h:
* gst/overlay/gstoverlay.h:
* gst/playondemand/gstplayondemand.h:
* gst/qtdemux/qtdemux.h:
* gst/rtjpeg/gstrtjpegdec.h:
* gst/rtjpeg/gstrtjpegenc.h:
* gst/smooth/gstsmooth.h:
* gst/smoothwave/gstsmoothwave.h:
* gst/spectrum/gstspectrum.h:
* gst/speed/gstspeed.h:
* gst/stereo/gststereo.h:
* gst/switch/gstswitch.h:
* gst/tta/gstttadec.h:
* gst/tta/gstttaparse.h:
* gst/videodrop/gstvideodrop.h:
* gst/xingheader/gstxingmux.h:
* sys/directdraw/gstdirectdrawsink.h:
* sys/directsound/gstdirectsoundsink.h:
* sys/dxr3/dxr3audiosink.h:
* sys/dxr3/dxr3spusink.h:
* sys/dxr3/dxr3videosink.h:
* sys/qcam/gstqcamsrc.h:
* sys/vcd/vcdsrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_class_init),
(gst_volume_init), (volume_process_float), (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps),
(volume_transform_ip), (plugin_init):
* gst/volume/gstvolume.h:
rewrite the passthrough check, split _int16 and _int16_clamp, fix
another property desc., remove unused param from process function
* tests/check/elements/volume.c: (volume_suite):
reactivate the passthrough test
2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
remove wrong commit
Original commit message from CVS:
remove wrong commit
2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/libvisual/visual.c: Handle DISCONT.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_reset),
(gst_visual_sink_setcaps), (gst_visual_sink_event),
(gst_visual_src_event), (get_buffer), (gst_visual_chain):
Handle DISCONT.
Use running time before doing QoS.
Handle mono too.
2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
Original commit message from CVS:
* docs/libs/Makefile.am:
set a magic variable to indicate we know the docs are incomplete
2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
Original commit message from CVS:
* win32/common/libgstvideo.def:
export gst_video_calculate_display_ratio
* win32/vs6/libgstvideoscale.dsp:
add link to libgstvideo-0.10.lib
2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_source_element):
Throw a more comprehensible error for rtsp:// URIs (rather
than erroring out with a negotiation error later on) until
we fix playbin to handle rtspsrc etc.
2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/pango/gsttextoverlay.c: Added some FIXMEs.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
(gst_text_overlay_text_event):
Added some FIXMEs.
2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.*: Implement release_request_pad.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
(gst_adder_request_new_pad), (gst_adder_release_pad):
* gst/adder/gstadder.h:
Implement release_request_pad.
Make padcounter atomic.
* tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
Added check for release_pad in adder.
2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Fix build again.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
Fix build again.
2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
(gst_ogg_demux_seek), (gst_ogg_demux_get_data),
(gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
(gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
add more debugging
clean up printf formats for granulepos and serialno
2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
* tests/check/generic/states.c:
properly fail if we can't make an element
Original commit message from CVS:
properly fail if we can't make an element
2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (raw_caps_factory),
(gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
(gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
(gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Multi-channel caps negotiation, so we can do proper multichannel
vorbis encoding, negotiated through audioconvert.
2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
Original commit message from CVS:
* tests/check/elements/adder.c: (test_event_message_received),
(test_play_twice_message_received), (GST_START_TEST),
(adder_suite):
Added check to show that #339935 is fixed with ongoing
adder and collectpads fixes.
2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.c: Don't leak pad name.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_request_new_pad):
Don't leak pad name.
2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.c: Fix adder seeking.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(forward_event_func), (forward_event), (gst_adder_src_event):
Fix adder seeking.
Make query/seeking code threadsafe.
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (test_event_message_received),
(GST_START_TEST), (test_play_twice_message_received):
Fix adder test case.
2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (gst_play_base_bin_dispose),
(set_encoding_element), (decodebin_element_added_cb),
(decodebin_element_removed_cb), (setup_subtitle), (setup_source),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
* gst/playback/gstplaybasebin.h:
Add 'subtitle-encoding' property to playbin, so applications can
force a subtitle encoding for non-UTF8 subtitles (#342268).
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
(gst_sub_parse_set_property):
Rename recently-added 'encoding' property to 'subtitle-encoding'
(so it can be proxied by playbin/decodebin in a generic way
with less danger of false positives).
2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(append_with_other_format), (set_structure_widths),
(gst_audio_convert_transform_caps):
Patch from #341562: give more specific audio caps in get_caps, so
that basetransform can make better decisions on what caps to
negotiate.
2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/volume.c: make it compile again
Original commit message from CVS:
* tests/check/elements/volume.c:
make it compile again
2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/volume.c: disable test until #343196 gets resolved
Original commit message from CVS:
* tests/check/elements/volume.c: (volume_suite):
disable test until #343196 gets resolved
2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/adder/gstadder.c: Make it easier to copy&paste
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_get_type):
Make it easier to copy&paste
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_set_mute),
(gst_volume_class_init), (volume_process_int16), (volume_set_caps),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume):
* gst/volume/gstvolume.h:
Add own debug category, move duplicate code to helper function, fix
property texts, add more comments and prepare ffor liboil-goodness
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
add test for mute and passtrough case, be a bit more verbose to track
failure
* tests/check/generic/states.c: (GST_START_TEST):
catch elements that fail to instantiate
2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
Original commit message from CVS:
* tests/check/pipelines/simple-launch-lines.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisenc.c:
Comment out tests using parse_launch() if core was built without
parsing capabilities.
2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
Original commit message from CVS:
* tests/check/Makefile.am:
Extra bonus points for whoever explains to ensonic that you are meant
to test unit tests thoroughly before commiting them, especially if
you know it's going to break.
De-activated element/adder tests.
2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
Marking caps conversion issues as GST_WARNING is way too verbose,
Moving them to GST_LOG.
2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
Original commit message from CVS:
* README:
Replace current README (containing the release notes from
some 0.9.x version) with a proper README taken from the core.
2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Small cleanups.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
(vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
(vorbis_dec_change_state):
Small cleanups.
Add some FIXMEs
Clip output samples to segment boundaries.
2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
(gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
Improve the errors produced on bad output, including some human
readable description strings.
Handle the (theoretical for ximagesink) case where the XServer
has a different idea about the size required for a particular
frame and gives us too small a memory allocation.
2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
Mention bugs fixed by previous commit
Original commit message from CVS:
Mention bugs fixed by previous commit
2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
Improve the errors produced on bad output, including some human
readable description strings.
Handle RGB Xv formats properly by transforming them into our
big-endian caps description.
Use gst_caps_truncate to ensure that we never try and choose a
non-fixed caps in buffer_alloc.
Handle the case where the XServer has a different idea about the size
required for a particular frame and gives us too small a memory
allocation.
Use -1 to indicate 'no image format', because 0 is a valid XServer
image format number.
Put RGB Xv formats at the end of the caps, so that we always prefer
YUV format frames.
Iterate the available Xv Encodings to determine the maximum width and
height, and then return that in our caps.
2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
When there is only one unfinished pad and it receives an event that
doesn't match our requirements, we need to set alldone=FALSE so that
the fakesink is not removed yet.
2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
Use gst_type_find_helper_for_buffer() to find the type
of stream from the first packet.
* configure.ac:
Bump requirements to core CVS (needed for vorbis
typefinding to work).
2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
Else they play perfectly fine with qtdemux.
2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
make more debug catagories static
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* gst/audiorate/gstaudiorate.c:
make more debug catagories static
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (message_received),
(test_event_message_received), (GST_START_TEST),
(test_play_twice_message_received), (adder_suite):
added test case for using element twice, extra bonus points for anyone
who can make these test run reliably
2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_chain):
Make work with time-stamped input buffers that do not
have a granulepos in BUFFER_OFFSET_END (like theora
buffers coming from matroskademux). Fixes #342448.
2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
(gst_gdp_depay_change_state):
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
(gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
(gst_gdp_pay_change_state):
* gst/gdp/gstgdppay.h:
Handle error cases when calling functions
do downwards state change after parent's change_state
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gdppay.c: (GST_START_TEST):
clean up more
2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
Original commit message from CVS:
* gst/gdp/Makefile.am:
* gst/gdp/gstgdp.c: (plugin_init):
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
(gst_gdp_depay_class_init), (gst_gdp_depay_init),
(gst_gdp_depay_finalize), (gst_gdp_depay_chain),
(gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
(gst_gdp_pay_class_init), (gst_gdp_pay_init),
(gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
(gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
(gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
(gst_gdp_queue_buffer), (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
(gst_gdp_pay_plugin_init):
* gst/gdp/gstgdppay.h:
* tests/check/Makefile.am:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
(setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
* tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
(GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
(main):
adding GDP payloader and depayloader. Build integration will
follow later when the GDP issues for core are sorted out.
2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/tcp/Makefile.am:
fdstresstest doesn't need Gtk+, fix compilation if
gtk is not available (#342566).
2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
80 line columns
Removed redundant floor()
2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
On second thought, just skip JUNK chunks automatically, so
the caller doesn't have to handle this. Fixes #342345.
Also, return GST_FLOW_UNEXPECTED if we get a short read,
not GST_FLOW_ERROR.
2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
Don't bail out on JUNK chunks with a size of 0 (would try to
pull_range 0 bytes before, which sources don't like too much).
See #342345.
2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Use the gstutil scaling function to preserve 64 bits while calculating
output width and height from the display-aspect-ratio. (A continuation
of #341542)
2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_buffer_alloc):
* sys/xvimage/xvimagesink.h:
When performing buffer allocations, remember the caps and image format
we return so that if the same caps are asked for next time we can
return them immediately without doing any caps intersections.
2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/README: Some new documentation
Original commit message from CVS:
2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/README:
Some new documentation
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs.
Not enabled in Makefile.am until approved.
2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
Original commit message from CVS:
* tests/check/elements/alsa.c: (test_device_property_probe):
Fix test case: don't try to free NULL GValueArray when there
are no devices.
2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/alsa.c: (test_device_property_probe),
(alsa_suite), (main):
Add simple test that runs a device property probe on alsasrc,
alsasink and alsamixer. Disable valgrind check for now (too
many leaks in libasound, and valgrind ignored my suppressions
additions).
2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
(gst_alsa_device_property_probe_probe_property),
(gst_alsa_device_property_probe_needs_probe),
(gst_alsa_device_property_probe_get_values),
(gst_alsa_type_add_device_property_probe_interface):
* ext/alsa/gstalsadeviceprobe.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_init_interfaces):
* ext/alsa/gstalsamixerelement.h:
Clean up and simplify alsa device probing. Make it actually work
for multiple classes. Don't cache results any longer.
* ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
(gst_alsasink_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
(gst_alsasrc_interface_supported), (gst_implements_interface_init),
(gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
Make alsasink and alsasrc implement the GstPropertyProbe interface
for device probing (#342181).
Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_font):
Don't ignore return value of strtol (++compiler_happiness).
2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist chollian net>
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_sub_parse_set_property), (gst_sub_parse_get_property),
(convert_encoding):
* gst/subparse/gstsubparse.h:
Add 'encoding' property (#341681).
* gst/subparse/samiparse.c: (characters_sami):
Output is pango markup, so we need to escape text
between tags (#342143).
2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
It's okay to have caps with channels=1 and a channel position
different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
(deinterleavers might want to keep the position in the caps,
so that they can be re-interleaved again properly later).
Leave check for unexpected 2-channel layouts intact for now.
2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
Original commit message from CVS:
2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/tcp/gsttcp.c: (gst_tcp_socket_read):
Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
basesrc can do its job correctly.
2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
(gst_alsa_detect_formats), (get_channel_free_structure),
(caps_add_channel_configuration), (gst_alsa_detect_channels),
(gst_alsa_probe_supported_formats):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
Refactor and improve caps probing code: probe signedness
when we probe the supported formats/widths; set endianness
to the one we actually probed for (ie. cpu endianness).
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
(gst_alsasrc_close):
* ext/alsa/gstalsasrc.h:
Implement caps probing for alsasrc.
2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
Original commit message from CVS:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_dec_src_query), (theora_dec_src_event),
(theora_dec_sink_event), (theora_handle_comment_packet),
(theora_handle_data_packet), (theora_dec_change_state):
Cleanups, add some G_LIKELY.
Use segment helpers instead of our own wrong code.
Clear queued buffers on seek and READY.
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
(vorbis_dec_convert), (vorbis_dec_src_query),
(vorbis_dec_src_event), (vorbis_dec_sink_event),
(vorbis_handle_comment_packet), (vorbis_dec_push),
(vorbis_handle_data_packet), (vorbis_dec_chain),
(vorbis_dec_change_state):
* ext/vorbis/vorbisdec.h:
Remove old useless packetno variable.
Do position query properly.
Add some G_LIKELY.
Do cleanup of queued buffers in new helper function
and use it.
2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
Query supported sample rates. Fixes #341732.
2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
Original commit message from CVS:
2006-05-15 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstdecodebin.c: (cleanup_decodebin),
(gst_decode_bin_change_state): Make decodebin reusable
when going from PAUSE_TO_READY and then back to PAUSED.
Fixes #331678.
2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
(vorbis_dec_convert), (vorbis_dec_src_query),
(vorbis_dec_sink_query), (vorbis_dec_src_event),
(vorbis_dec_sink_event), (vorbis_handle_identification_packet),
(vorbis_dec_clean_queued), (vorbis_dec_push),
(vorbis_handle_data_packet), (vorbis_dec_change_state):
Cleanups. Use refcounting and DEBUG_OBJECT.
Reset segment on flush, use code methods instead of our
own wrong version.
Fix potential memleak.
2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_init):
* ext/alsa/gstalsasink.h:
Don't leak allocated snd_output_t structure if there's
more than one alsasink instance at a time (#341873).
Also fix GObject macros in header file.
2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
Don't use libxml functions in the typefinding code.
2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
Fix seeking performance in the case where a non-header
packet has a 0 granulepos (busted theora case).
Fixes #341719
2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
Improve SAMI typefinding: handle case where there are
whitespaces or newlines in front of the first <SAMI>
tag (#169936).
2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
Original commit message from CVS:
* configure.ac:
Build video4linux plugin even if there's no XVIDEO, just
without implementing the GstXOverlay interface (#334002).
2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
Add tentative support for libvisual-0.4 (#336881).
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
(plugin_init):
Add tentative support for libvisual-0.4 (#336881).
2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (handle_start_font):
Need to map "silver" colour explicitly (#169936).
2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
gst/subparse/: Add support for SAMI subtitles (#169936).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(parser_state_dispose), (gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (feed_textbuf),
(gst_subparse_type_find), (plugin_init):
* gst/subparse/gstsubparse.h:
* gst/subparse/samiparse.c:
* gst/subparse/samiparse.h:
Add support for SAMI subtitles (#169936).
2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* win32/common/config.h:
update config.h
Original commit message from CVS:
update config.h
2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/README:
fix mistakes in README
Original commit message from CVS:
fix mistakes in README
2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix #341696: crash when mixing L+R+C to mono or stereo.
* tests/check/Makefile.am:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
(audioconvert_suite):
Add test for the above, including some generic framework bits for
testing multichannel things.
2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:
Back to CVS
Original commit message from CVS:
Back to CVS
=== release 0.10.7 ===
2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: releasing 0.10.7, "Leave the gun"
Original commit message from CVS:
2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:
releasing 0.10.7, "Leave the gun"
2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* common:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Fix the build.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Fix the build.
2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
* gst-libs/gst/video/video.h:
* gst/videoscale/Makefile.am:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
* tests/check/Makefile.am:
* tests/check/libs/video.c: (GST_START_TEST), (video_suite),
(main):
Fix integer overflow problem with pixel-aspect-ratio calculations
in videoscale and xvimagesink (#341542)
2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/MANIFEST: update win32 files listing
Original commit message from CVS:
* win32/MANIFEST:
update win32 files listing
2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
disable failing check on gentoo64
Original commit message from CVS:
disable failing check on gentoo64
2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
disable failing check on gentoo64
Original commit message from CVS:
disable failing check on gentoo64
2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
macros show the correct line
Original commit message from CVS:
macros show the correct line
2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
macros show the correct line
Original commit message from CVS:
macros show the correct line
2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
Original commit message from CVS:
2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
patch by: Sjoerd Simons (sjoerd@luon.net)
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(group_create), (group_destroy), (add_stream),
(gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
API: GstPlayBaseBin::stream-info-value-array property
use a more bindings-friendly way of exposing streaminfo
using a GValueArray. Tested in ipython.
Closes #341114
2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
fix some type warnings
Original commit message from CVS:
fix some type warnings
2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
(queue_underrun_cb), (queue_filled_cb):
Also catch queue underruns but don't do anything yet.
Refactor and comment queue enlarging code a bit.
* gst/playback/gstplaybasebin.c: (queue_overrun),
(queue_threshold_reached), (queue_out_of_data),
(gen_preroll_element):
If a queue over/underruns check that we don't create nasty
deadlocks when the min-threshold is not reached but the
max-bytes is. In those cases disable max-bytes when we
know that the queue is fed timed data.
Add more comments.
2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Make playbin automatically plug an 'audioresample'
element before the audio sink as well. This solves
problems with sinks that only accept a very specific
sample rate, like esdsink (e.g. #340379).
2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_source_element):
Make http sources send special headers so that we receive
icecast metadata if the http stream is an icecast stream
(otherwise the server will just ignore them). This also
means that from now on users will need the 'icydemux'
element from gst-plugins-good installed if they want to
listen to icecast radio streams. (#341432, #333657).
2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
more commenting
Original commit message from CVS:
more commenting
2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
remove stupid example from docs - it should come with a simple
C program instead.
Clean up/fix docs
* tests/check/elements/multifdsink.c: (wait_bytes_served),
(fail_if_can_read), (GST_START_TEST),
(gst_multifdsink_create_streamheader), (multifdsink_suite):
add a test for changing streamheader which exposes a bug in
multifdsink
2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_received_headers_callback):
* ext/gnomevfs/gstgnomevfssrc.h:
Don't set icy-caps unless we have a sane interval value. Move
interval to a local variable; we never use it outside this function.
2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
Register special buffer types along with the objects so
that they are not registered at runtime from N different
streaming threads since they are not threadsafe.
2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/multifdsink.c:
set caps and plug leaks
Original commit message from CVS:
set caps and plug leaks
2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
Original commit message from CVS:
* tests/check/elements/multifdsink.c: (wait_bytes_served),
(GST_START_TEST), (fail_unless_read), (multifdsink_suite):
add two more tests, one doing streamheader
2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
clean up the bufqueue when shutting down
* tests/check/Makefile.am:
* tests/check/elements/multifdsink.c: (setup_multifdsink),
(cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
(main):
add a test for the leak that was just fixed
2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
DEBUG_FUNCPTR'ing
Original commit message from CVS:
DEBUG_FUNCPTR'ing
2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
whitespace fixes
Original commit message from CVS:
whitespace fixes
2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_query_duration), (gst_adder_query), (forward_event),
(gst_adder_src_event), (gst_adder_sink_event),
(gst_adder_class_init), (gst_adder_finalize),
(gst_adder_request_new_pad), (gst_adder_collected):
* gst/adder/gstadder.h:
Updated some docs. Added comments and FIXMEs all over the place.
Improve debugging info.
Fix leak on finalize by not calling the parent.
Implement duration query.
Make event forwarding threadsafe.
Correctly send NEWSEGMENT at start and after flush.
Handle EOS correctly.
Post error when not negotiated.
* tests/check/elements/adder.c: (GST_START_TEST):
Added FIXME in the test.
2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
Const-ify GEnumValue and GFlagsValue arrays. Use
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
(gst_text_overlay_halign_get_type),
(gst_text_overlay_wrap_mode_get_type):
* ext/theora/theoradec.c: (theora_handle_type_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (gst_border_mode_get_type),
(theora_enc_sink_setcaps), (theora_enc_chain):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_mode_get_type):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type):
* gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
* gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
(gst_sync_method_get_type), (gst_unit_type_get_type),
(gst_client_status_get_type):
* gst/videoscale/gstvideoscale.c:
(gst_video_scale_method_get_type):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_pattern_get_type):
* gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
(paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
(paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
(paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
(paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
(paint_setup_RGB565), (paint_setup_xRGB1555):
Const-ify GEnumValue and GFlagsValue arrays. Use
GST_ROUND_UP_* macros instead of home-made ones.
2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Require core CVS for the new newsegment stuff.
Original commit message from CVS:
* configure.ac:
Require core CVS for the new newsegment stuff.
2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon net>
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
Register nick for enum value (#341160).
2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
(plugin_init):
backout typefind patch #340375
* tests/check/elements/adder.c: (message_received),
(GST_START_TEST), (adder_suite):
redo, signal-handling of test
2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_request_new_pad),
(gst_adder_collected):
* gst/adder/gstadder.h:
Remove bogus segment merging and forwarding, we don't
care about timestamps anyway and we just produce a
continuous stream.
Also create a nice NEWSEGMENT event when we start.
Use _scale_int some more.
2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
Original commit message from CVS:
* tests/icles/stress-xoverlay.c:
Fix if core was built without parsing support.
2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add SEDG (Samsung MPEG-4) fourcc.
2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
tests/examples/volume/volume.c: Fox if core was built without parsing support.
Original commit message from CVS:
* tests/examples/volume/volume.c:
Fox if core was built without parsing support.
* tests/examples/seek/seek.c:
Disable the parse_launch example if core was built without parsing
support.
2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Disable the parse_launch example if core was built without parsing
support.
2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* docs/libs/tmpl/gstcolorbalance.sgml:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/tcp/gstmultifdsink.c:
* gst/videoscale/gstvideoscale.c:
doc reparagraphing and DEBUG_FUNCPTRing
Original commit message from CVS:
doc reparagraphing and DEBUG_FUNCPTRing
2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
Original commit message from CVS:
* autogen.sh: (CONFIGURE_DEF_OPT):
libtoolize on Darwin/MacOSX is called glibtoolize
2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
Disable the adder test, until the build-slaves posses the kindness to
either like it or to give valid reason for not doing so
2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
Original commit message from CVS:
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
(adder_suite):
Shuffle NULL state change around and raise timeout more
2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
(mp4_type_find), (plugin_init):
Add typefind to distinguish between "audio/x-m4a" and new type
"video/mp4". Fixes #340375
* tests/check/elements/adder.c: (adder_suite):
Raise timeout to make buildbot happy
2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_event),
(gst_adder_request_new_pad), (gst_adder_change_state):
* gst/adder/gstadder.h:
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
(adder_suite), (main):
Add sink-event handling to adder. It tries to merge incomming
newsegment-events. Added test to check if segment_done is comming
through.
2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* ext/theora/theoraparse.c:
* ext/vorbis/vorbisparse.c:
ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
Original commit message from CVS:
2006-05-05 Andy Wingo <wingo@pobox.com>
* ext/theora/theoraparse.c (gst_theora_parse_init)
(theora_parse_src_convert, theora_parse_src_query):
* ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
(vorbis_parse_convert, vorbis_parse_src_query): Add convert and
query functions on the source pads of the theora and vorbis parse
elements. Fixes position querying when doing a remux.
2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
ext/theora/theoraparse.c: Fix flushing.
Original commit message from CVS:
* ext/theora/theoraparse.c: (parse_granulepos),
(theora_parse_drain_queue_prematurely),
(theora_parse_queue_buffer), (theora_parse_sink_event):
Fix flushing.
Fix invalid granulepos outputs when starting with a non-keyframe.
2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
(mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
Rearrange MPEG system stream detection, fixing some memleaks in the
process.
Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
they clean up their data correctly.
Remove unused ogganx caps and move the 'is_annodex' check to inside
the 'is_ogg' if statement.
2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (cleanup_decodebin):
Properly remove ghostpads. Fixes #340392
2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
gst/typefind/gsttypefindfunctions.c:
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mpeg_ts_probe_headers), (mpeg_ts_type_find):
When typefinding an MP3 in push-based mode, don't penalise the
probability down to 74% when we found 5 valid frames just because we
can't peek the end of the file.
Make the probability for detecting MPEG Transport Streams based on the
number of sequential headers we successfully detected.
2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_dec_push), (vorbis_dec_chain):
Still produce an error when we receive an empty packet.
2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
(gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
Mark buffers with DISCONT after seek and after activating new
chains.
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_get_query_types), (theora_dec_sink_event),
(theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
(theora_dec_change_state):
Fix frame counter.
Detect and mark DISCONT buffers.
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
(vorbis_dec_change_state):
* ext/vorbis/vorbisdec.h:
Use GstSegment.
Detect and mark DISCONT buffers.
Don't crash on 0 sized buffers.
2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
(volume_transform_ip):
Increase "volume" property to 10.0. Fixes #340369.
Set the process function to NULL when capsnego fails so that
we properly error out.
2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink):
* gst/playback/test.c: (main):
* gst/playback/test5.c: (dump_element_stats):
* gst/playback/test6.c: (main):
free cpas using gst_caps_unref, don't leak caps-strings
2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-libs/gst/rtp/gstbasertppayload.c:
some RTP debug
Original commit message from CVS:
some RTP debug
2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Refine musepack typefinding a bit. Return MAXIMUM
probability when we detect stream version 7 to make
sure the mpeg audio typefinder doesn't trump us.
2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Protect against unexpected NULL strf_data buffer.
2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
interpret the out[] buffer in the order the bytes are actually
put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
Other tests should use BYTE_ORDER since the array is filled in
with actual values
2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/elements/audioconvert.c:
dump expected data when audioconvert test fails
Original commit message from CVS:
dump expected data when audioconvert test fails
2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
when a test fails, give an indication of which it is
2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
* ext/theora/theoraenc.c:
add another include
Original commit message from CVS:
add another include
2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/subparse/gstssaparse.c:
atoi() needs stdlib.h
Original commit message from CVS:
atoi() needs stdlib.h
2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
exit needs stdlib.h
Original commit message from CVS:
exit needs stdlib.h
2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
compile fix; strtol() needs <stdlib.h>
2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* common:
* docs/Makefile.am:
* docs/libs/Makefile.am:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/plugins/Makefile.am:
* docs/upload.mak:
use common upload.mak
Original commit message from CVS:
use common upload.mak
2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
make GstElementDetails const
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_init):
send events from src-pad to all sink-pads fixes #338657
2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
(alsasink_parse_spec):
query witdh capabilities from alsa, fixes #338919
2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_remove_client_link):
* gst/tcp/gstmultifdsink.h:
Fix race condition in multifdsink that can lead to spurious
duplicate clients. this patch adds a new signal that is fired when
multifdsink has removed all references to the fd.
Fixes #339574.
Updated documentation.
API: client-fd-removed signal added
2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
When asking g_value_array_new to prealloc elements, we may as well
ask for the right number of elements.
2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
patch to make timestamp checking more tollerant to rounding
errors given that real discontinuities are to be marked on
buffers. Fixes some asf files and #338778.
Also avoid some crashers when we receive an event in the
NULL state.
2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
(gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_get_property),
(gst_gnome_vfs_src_send_additional_headers_callback),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
(gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Remove ICY handling (mostly) from gnomevfssrc, in favour of
proper shared support within icydemux.
2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_swap_prev), (gst_video_rate_chain):
fix up docs
fix a leak when no caps negotiated
fix counting of input frames
* tests/check/elements/.cvsignore:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(GST_START_TEST), (videorate_suite):
add tests for these
2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
(gst_ring_buffer_clear), (gst_ring_buffer_may_start):
Check arguments passed to public functions instead of
crashing.
2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
GstBaseAudioSrc must be live or it does not work.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
Don't set live to TRUE as this is the default in the parentclass.
2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* win32/common/config.h:
update config.h
Original commit message from CVS:
update config.h
2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
(gst_video_scale_fixate_caps), (gst_video_scale_src_event):
Videoscale doesn't pass on pixel-aspect ratio. Handle all
fixation cases better. Fixes #338991
2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
Handle 0/1 framerate correctly Fixes #331901.
2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (get_float_caps),
(GST_START_TEST), (audioconvert_suite):
Added check for correct clipping when doing float samples
in audioconvert.
2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videorate/gstvideorate.c: Print more debugging info.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_event),
(gst_video_rate_chain):
Print more debugging info.
2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(resample_set_state_from_caps):
Add support for other formats audioresample can handle such as
32 bits in and float and 64 bits float. Fixes #301759
2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float):
correctly clip float samples > 1.0. Fixes #338718
2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_render_text):
Don't strip newlines from the text. Also, center lines
within multi-line paragraphs (#339405).
2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
Fix wavpack typefinding to work in more cases (don't peek
for chunks of multiple hundred kBs at once, but process
things step-by-step in smaller units). Fixes #339786.
2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* configure.ac:
back to HEAD
Original commit message from CVS:
back to HEAD
=== release 0.10.6 ===
2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* docs/upload.mak:
releasing 0.10.6
Original commit message from CVS:
releasing 0.10.6
2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* win32/MANIFEST:
* win32/common/config.h:
dist more win32 files
Original commit message from CVS:
dist more win32 files
2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
gst/videoscale/gstvideoscale.c: Add call to oil_init().
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: Add call to oil_init().
Fixes #338897.
2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* configure.ac:
* win32/common/config.h:
new prerelease
Original commit message from CVS:
new prerelease
2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
Original commit message from CVS:
2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
patch by: Wim Taymans
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek):
make sure correct newsegments are sent, so that the decoder
and the demuxer agree on timestamps. Fixes playback of a lot
of Ogg files that do not start from 0. Fixes #339833.
2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
* tests/check/Makefile.am:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(setup_videorate), (cleanup_videorate), (GST_START_TEST),
(videorate_suite), (main):
Fix an infinite loop if frames are passed in with wrongly ordered
timestamps. Fixes #339013.
2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
* win32/common/config.h:
prerelease
Original commit message from CVS:
prerelease
2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
Original commit message from CVS:
Patch by: Tim-Philipp Müller <tim at centricular dot net>
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
fix typefinding on some ISO files. Fixes #339212.
2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
Original commit message from CVS:
Patch by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
add another H264 fourcc. Fixes #339047.
2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
Original commit message from CVS:
Patch by: Jan Schmidt
* gst/playback/gststreamselector.c:
(gst_stream_selector_bufferalloc):
Restore old StreamSelector behaviour.
Fixes #338419.
2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtpbuffer.h:
reverting rtp patches to fix freeze break on -base as explained on the list
Original commit message from CVS:
reverting rtp patches to fix freeze break on -base as explained on the list
2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* configure.ac:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
update libtool versioning
Original commit message from CVS:
update libtool versioning
2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* configure.ac:
* win32/common/config.h:
prerelease
Original commit message from CVS:
prerelease
2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
Fix some memory leaks: on finalize, free buffers left in the queue
before destroying the queue; in _push(), unref rtp_buf even if
the process vfunc returned a NULL buffer as output buffer (#337548);
demote some recuring debug messages to LOG level.
2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
fix version number macro
Original commit message from CVS:
fix version number macro
2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: More cleanups.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_chain_free), (gst_ogg_demux_sink_event),
(gst_ogg_demux_loop):
More cleanups.
Respect segment stop when emiting EOS or SEGMENT_DONE.
Fixes (#337945).
2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gststreamselector.c: Don't leak pad name.
Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_get_property):
Don't leak pad name.
2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
Mention bug #336617 closed by recent commit
Original commit message from CVS:
Mention bug #336617 closed by recent commit
2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
tests/check/: so that FC4 buildslaves can pass.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/gst-plugins-base.supp:
Suppress an old libtheora bug (fixed in more recent versions), so
that FC4 buildslaves can pass.
2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Don't leak events.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_demux_receive_event), (gst_ogg_pad_event),
(gst_ogg_demux_init), (gst_ogg_demux_finalize),
(gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
(gst_ogg_demux_loop):
Don't leak events.
Remember what error we got when finding chains, if we
were shutdown, that would not be an error.
2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event):
Starting the ringbuffer when we did not acquire it can cause
a deadlock, is pointless and causes nasty things for
subclasses.
Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Add some more debugging.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_demux_receive_event), (gst_ogg_pad_event),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
(gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
Add some more debugging.
2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* ext/theora/theoraenc.c:
fix width of docs
Original commit message from CVS:
fix width of docs
2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Some more debug info.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_event),
(theora_handle_data_packet):
Some more debug info.
* tests/examples/seek/seek.c: (start_seek), (main):
Print element messages too.
2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
Original commit message from CVS:
* gst/audioresample/debug.h:
replace debug macros with variable number of parameters
by a simple alias to gstreamer standard debug macros
(#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
supported by MSVC 6.0 and 7.1)
* gst/audioresample/resample.h:
define M_PI and rint for WIN32
* win32/common/libgstaudio.def:
* win32/common/libgstriff.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
add new exported functions
* win32/vs6:
update project files
2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
Original commit message from CVS:
* ext/alsa/gstalsamixeroptions.c:
(gst_alsa_mixer_options_class_init):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstaudiosrc.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
* gst-libs/gst/interfaces/colorbalancechannel.c:
(gst_color_balance_channel_class_init):
* gst-libs/gst/interfaces/mixeroptions.c:
(gst_mixer_options_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/interfaces/tunerchannel.c:
(gst_tuner_channel_class_init):
* gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netbuffer_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_channel_class_init):
* sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
(gst_v4l_tuner_norm_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
* tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Fix broken GObject macros
Original commit message from CVS:
* ext/pango/gsttextrender.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
* gst/playback/gstplaybasebin.h:
* gst/tcp/gstmultifdsink.h:
* sys/v4l/gstv4lelement.h:
Fix broken GObject macros
2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
More debug to trace why my USB headset is not working with gst
2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy):
Clean up our group elements properly in the case where it never
got committed - it still got added unconditionally to the bin.
2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Unref unhandled events.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_sink_event),
(theora_handle_data_packet), (theora_dec_chain):
Unref unhandled events.
Protect against empty buffers.
Perform QoS on running time.
2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
(gst_vorbis_enc_chain):
Remove leaks from vorbisenc.
Mostly minor changes, the only significant one is that now the
buffers we set as 'streamheader' on the caps are copies of the
original buffers, to avoid circular refcounting problems.
2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
Don't remove our mute-probe if someone else already did so.
Don't set a 2nd one if there is already one pending on the pad.
* gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
(do_playbin_seek):
When a seek fails, ensure that playbin is still set back to playing.
* gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
(mpeg_ts_type_find), (plugin_init):
Add a typefind function for mpeg-ts streams.
2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/videorate/gstvideorate.c:
gst/videorate/gstvideorate.c (gst_video_rate_reset)
Original commit message from CVS:
2006-04-06 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_reset)
(gst_video_rate_init): Caps-related parameters should not be reset
by a flush -- move their inits to the instance init function.
(gst_video_rate_flush_prev): Don't complain if gst_pad_push
is not OK, just return the result.
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_class_init)
(gst_audio_test_src_get_times): Re-enable is-live=true, as was
broken by Stefan's commit on 24 March.
2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
Original commit message from CVS:
2006-04-06 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
buffers being pushed out. Fixes oggmux ! multifdsink.
2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
(gst_vorbis_dec_init), (vorbis_dec_finalize):
* ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
(gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
(gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
(gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
(gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
(gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
(gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
(gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
(gst_vorbis_enc_buffer_from_packet),
(gst_vorbis_enc_buffer_from_header_packet),
(gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
(gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
(gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
(gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
(gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
vorbisenc adhere to the official nomenclature; use boilerplate
macro.
2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
Original commit message from CVS:
2006-04-04 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
Whoops, fix bug introduced. Bad hacker!
2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
Original commit message from CVS:
2006-04-04 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
Properly handle the case where you get EOS before any buffers are
received. Use gst_buffer_make_metadata_writable where appropriate.
2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
Original commit message from CVS:
2006-04-04 Andy Wingo <wingo@pobox.com>
* ext/theora/theoradec.c (theora_handle_data_packet): This value
is often negative -- make it signed so as not to wrap around.
Fixes segfaults introduced on 9 March.
2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/: Don't try to store a gdouble in a gboolean.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (theora_dec_src_event):
Don't try to store a gdouble in a gboolean.
Small cleanups.
2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggmux.c: Oggmux sucks.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
Oggmux sucks.
Make it suck slightly less by writing out the final page.
Still can't encode a vorbis-in-ogg file correctly, though.
2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
Original commit message from CVS:
2006-04-03 Andy Wingo <wingo@pobox.com>
* ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
a g_print.
2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
ext/theora/theora.c (plugin_init): Register theoraparse.
Original commit message from CVS:
2006-04-03 Andy Wingo <wingo@pobox.com>
* ext/theora/theora.c (plugin_init): Register theoraparse.
* ext/theora/gsttheoraparse.h:
* ext/theora/theoraparse.c: New files implementing a theora
parser. Now we can properly remux ogg/theora+vorbis, yay.
2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
ext/vorbis/vorbisparse.c: Add some docs and a copyright.
Original commit message from CVS:
2006-04-03 Andy Wingo <wingo@pobox.com>
* ext/vorbis/vorbisparse.c: Add some docs and a copyright.
2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* configure.ac:
don't use AS_LIBTOOL_TAGS, it doesn't work
Original commit message from CVS:
don't use AS_LIBTOOL_TAGS, it doesn't work
2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* ext/pango/gsttextoverlay.c:
* sys/v4l/gstv4lsrc.c:
remove BT8x8 from description, works for more devices
Original commit message from CVS:
remove BT8x8 from description, works for more devices
2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Fixed the sample pipeline (see #323798)
2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
use AS_VERSION and AS_NANO more cleanups
Original commit message from CVS:
* configure.ac:
* win32/common/config.h:
* win32/common/config.h.in:
use AS_VERSION and AS_NANO
more cleanups
2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
Original commit message from CVS:
2006-03-31 Andy Wingo <wingo@pobox.com>
* ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
uninitialized variable return that would happen.
2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
Original commit message from CVS:
2006-03-31 Andy Wingo <wingo@pobox.com>
* ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
uninitialized variable return that would never happen.
2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
Original commit message from CVS:
2006-03-31 Andy Wingo <wingo@pobox.com>
* ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
(vorbis_parse_sink_event): Add an event function to flush our
state on a seek, and to drain buffers on a premature EOS.
(vorbis_parse_push_headers, vorbis_parse_clear_queue)
(vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
(vorbis_parse_chain, vorbis_parse_queue_buffer)
(vorbis_parse_drain_queue): Queue up buffers until we can set
their timestamps and granulepos values.
* ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
and keep track of data needed for deriving granulepos and
timestamps for buffers.
2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
expose pluginsdir so gonlin can use it for tests
Original commit message from CVS:
expose pluginsdir so gonlin can use it for tests
2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
add ccda to libraries
Original commit message from CVS:
add ccda to libraries
2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
better/unified long descriptions
Original commit message from CVS:
Patch by: j^ <j at bootlab dot org>
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
better/unified long descriptions
Fixes #336477
2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
Original commit message from CVS:
* tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
(stop_seek):
Don't let double and tripple clicks mess up our state.
2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element):
Error out gracefully when we can't create any of the usual
conversion elements for some reason. Also, don't try to
create an audioscale (sic) element that's not used anyway.
2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Don't post RESOURCE_NOT_FOUND error when we can't find a source
element for a particular protocol, that's confusing for users.
Instead, post a RESOURCE_FAILED error, so that our own error
message is actually shown in totem etc. (#336303).
2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_get_icy_metadata):
Fix some minor memory leaks (#336194).
2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfs.c:
(gst_gnome_vfs_location_to_uri_string):
* ext/gnomevfs/gstgnomevfs.h:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnome_vfs_sink_set_property):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
Make gnomevfssink accept filenames as well as URIs for the
"location" property, just like gnomevfssrc does (and
filesrc/filesink do) (#336190).
2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
Original commit message from CVS:
* tests/check/generic/clock-selection.c: (GST_START_TEST):
set to NULL before unreffing, fixes a valgrind leak.
Why was this not triggering the error that an object needs to
be NULL before unreffing ?
* win32/common/config.h:
update
2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (convert_encoding),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
Text subtitle files may or may not be UTF-8. If it's not, we
don't really want to see '?' characters in place of non-ASCII
characters like accented characters. So let's assume the input
is UTF-8 until we come across text that is clearly not. If it's
not UTF-8, we don't really know what it is, so try the following:
(a) see whether the GST_SUBTITLE_ENCODING environment variable
is set; if not, check (b) if the current locale encoding is
non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
the current locale encoding is UTF-8 and the environment variable
was not set to any particular encoding. Not perfect, but better
than nothing (and better than before, I think) (fixes #172848).
2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* docs/plugins/tmpl/.gitignore:
* tests/check/libs/.gitignore:
* tests/check/pipelines/.gitignore:
* tests/examples/volume/.gitignore:
ignore more
Original commit message from CVS:
ignore more
2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
Original commit message from CVS:
2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
* configure.ac:
update core requirement to 0.10.4.1 because of async_playback
vmethod on GstBaseSink
2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
use DEBUG_FUNCPTR for collectpads
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
* gst/adder/gstadder.c: (gst_adder_init):
use DEBUG_FUNCPTR for collectpads
2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* Makefile.am:
don't go through check-torture if no check installed
Original commit message from CVS:
don't go through check-torture if no check installed
2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
Original commit message from CVS:
* win32/common/libgstinterfaces.def:
Add a lot of export functions for gst-python
* win32/common/libgstinterfaces.dsp:
Add a missing include folder in the project configuration
2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
Fix audio sources, forgot to make the ringbuffer
startable...
2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
unparent instead of unref the ringbuffer.
2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
Implement new async_play vmethod to start slaving and allow
playback start in case of async PLAY state changes.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Enable QoS with new method in base class.
2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
Original commit message from CVS:
Patch by: Julien MOUTTE <julien at moutte dot net>
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
(gst_video_test_src_do_seek), (gst_video_test_src_create):
Partially handle 0 framerate, only EOS after the first frame
is missing.
2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
gst/: Patch for support of YVU9 AVI files (#334822)
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c:
Patch for support of YVU9 AVI files (#334822)
2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
Original commit message from CVS:
* docs/design/design-decodebin.txt:
Added design document for new decodebin
(Target Caps): text/x-pango-markup is also a default target caps.
2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
docs/design/design-decodebin.txt: Added design document for new decodebin
Original commit message from CVS:
* docs/design/design-decodebin.txt:
Added design document for new decodebin
2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_dispose):
Since we _parent the ringbuffer, we also need to
_unparent instead of a plain _unref.
2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/seek.c: Add scrub checkbox.
Original commit message from CVS:
* tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
(stop_seek), (scrub_toggle_cb), (main):
Add scrub checkbox.
2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
Original commit message from CVS:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
(gst_ogg_parse_chain):
Fix very inefficient usage of linked lists (#335365).
2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
gcc 4.1 unreferenced pointer fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* gst/playback/gstplaybin.c: (handoff):
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property):
gcc 4.1 unreferenced pointer fixes.
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
gst_buffer_ref() now takes a GstBuffer*.
2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
Original commit message from CVS:
2006-03-20 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_get_format_from_caps): Fix a memleak reported
by Jan Schmidt.
2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
(id3v1_type_find), (apetag_type_find), (plugin_init):
Can't do tag preferences via probability, as tags would then
lose against types that are recognised with MAXIMUM probability
(like .wav); so let all tag typefinders return MAXIMUM themselves
and order them via the rank. Split ID3v1 and ID3v2 typefinders so
that we can prefer APE to ID3v1 (fixes #335028).
2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix #330748.
2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Revert accidental commits to these files.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
* win32/common/config.h:
Revert accidental commits to these files.
2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
Original commit message from CVS:
Patch by: Michal Benes <michal dot benes at xeris dot cz>
* tests/Makefile.am:
Don't try to build tests in tests/icles if we
don't have X (#323852)
2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add TXXX frame identifiers for replaygain stuff as used
by some taggers (see #323721).
2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc):
Preserve the existing buggy streamselector behaviour by performing
a fallback buffer allocation when downstream isn't linked yet.
This should really be fixed in playbin by blocking pads until it's
linked them.
Also, use gst_pad_alloc_buffer instead of
gst_pad_alloc_buffer_and_set.
2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Don't crash on unknown ID3v2 TXXX frames.
2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
Chain up to the parent finalize method.
Add 32-bit sample size to the template caps.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add the fourcc that the VMWare codec uses.
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc),
(gst_stream_selector_request_new_pad):
For the active pad, forward buffer-alloc requests, otherwise
return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
having to memcpy every frame when used by playbin.
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
Get negotiated caps from the sink pad, rather than the sink
pad's peer.
2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
Don't forget to set src->callbacks_pushed to FALSE again when
popping them, otherwise re-activation in a different mode won't
work (#334620).
2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
Original commit message from CVS:
Patch by: Sebastien Moutte <sebastien moutte net>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
(gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
(gst_ffmpeg_smpfmt_to_caps):
Replace __VA_ARGS__ caps creation macros with varargs functions.
Makes things compile on MSVC (#320765), looks nicer, and we can
tell the compiler to check for the NULL terminator.
2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make sure the buffer we copy into is really always big
enough, this time for real (#333488).
2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for 24bpp DIB (#305279).
2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/: Re-enable QoS after the release.
Original commit message from CVS:
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_init), (gst_video_scale_src_event):
Re-enable QoS after the release.
Rework videoscale to use the base class src_event handler.
2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: back to CVS.
Original commit message from CVS:
* configure.ac:
back to CVS.
=== release 0.10.5 ===
2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/config.h:
releasing 0.10.5
Original commit message from CVS:
releasing 0.10.5
2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
Original commit message from CVS:
* docs/plugins/Makefile.am:
Part of previous cdparanoiasrc docs fixes, forgot to commit.
2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/plugins/: Add cdparanoiasrc to docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
Add cdparanoiasrc to docs.
* gst-libs/gst/cdda/gstcddabasesrc.c:
More GstCddaBaseSrc docs.
2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
Add new API to libgsttag: gst_tag_from_id3_user_tag().
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
* gst-libs/gst/tag/tag.h:
Add new API to libgsttag: gst_tag_from_id3_user_tag().
2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
NULL-terminate array of mpeg4 video file extensions.
Fixes crash on PPC (#334226).
2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
(gst_gnome_vfs_src_check_get_range):
gnome_vfs_uri_is_local() alone is not a good indicator
whether we can operate in pull-mode with a specific URI,
as it returns FALSE for file:// URIs that point to an
NFS-mounted path. Be more conservative here: whitelist
local files, blacklist http URIs and use the old
mechanism for anything else (fixes #334216).
2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
configure.ac: back to trunk
Original commit message from CVS:
* configure.ac:
back to trunk
=== release 0.10.4 ===
2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* docs/upload.mak:
* win32/common/config.h:
releasing 0.10.4
Original commit message from CVS:
releasing 0.10.4
2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
Original commit message from CVS:
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Disable max-lateness by setting it to -1 for now, so that
we can bed QoS stuff in thoroughly between now and the next
release.
2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make sure we don't read beyond the palette buffer in case of
broken or manipulated files (#333488, patch by: Fabrizio
Gennari)
2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
Fix for variable not initialized.
2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
* docs/libs/tmpl/gstringbuffer.sgml:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* win32/common/config.h:
prereleasing
Original commit message from CVS:
prereleasing
2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/libvisual/visual.c: Small cleanups.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_get_type),
(gst_visual_src_setcaps), (gst_vis_src_negotiate),
(gst_visual_chain):
Small cleanups.
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_init),
(gst_theora_dec_reset), (_theora_granule_time),
(theora_dec_src_convert), (theora_dec_sink_convert),
(theora_dec_src_query), (theora_dec_src_event),
(theora_dec_sink_event), (theora_handle_comment_packet),
(theora_handle_header_packet), (theora_dec_push),
(theora_handle_data_packet), (theora_dec_chain),
(theora_dec_change_state):
Add simple QoS.
2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
(audiocast_register_listener), (gst_gnome_vfs_src_start):
Some cleanups.
2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
Don't try to activate NULL chains.
2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
Fix invalid memory access to region before peek'd data (#332964).
2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
closes #333510.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
* ext/pango/gsttextrender.c: (gst_text_render_init):
* gst/adder/gstadder.c: (gst_adder_init):
Don't leak padtemplates, patch by Christophe Fergeau,
closes #333510.
2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Fix invalid memory access: make sure string passed to
regexec() is NUL-termianted.
2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mp3_type_find):
Refactor mpeg/audio typefinding to make it more maintainable
and easier to fine-tune. Make probing into middle of the file
work properly (fixes #333900, also see #152688).
2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(utf8_type_find_have_valid_utf8_at_offset):
Remove part from previous commit that was bogus:
g_utf8_validate() does in fact not accept embedded
zeroes, so we don't need to check for those (thanks
to Mike for the hint).
2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(utf8_type_find_count_embedded_zeroes),
(utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
Make plain/text typefinder more conservative: firstly, check
for embedded zeroes, which are perfectly valid UTF-8 characters,
but also a fairly good sign that something is not a plain text
file; secondly, probe into the middle of the file if possible.
If we can't probe into the middle, limit the probability value
to be returned to TYPE_FIND_POSSIBLE (see #333900).
2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Make typefind function name for mpeg4 video unique.
2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/libvisual/visual.c: Cleanups, post nice errors.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_init),
(gst_visual_clear_actors), (gst_visual_dispose),
(gst_visual_reset), (gst_visual_src_setcaps),
(gst_visual_sink_setcaps), (gst_vis_src_negotiate),
(gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Cleanups, post nice errors.
Handle sink and src events.
Implement simple QoS.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Use new basesink methods to configure max-lateness.
Small doc update.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
Debug statement cleanups.
* gst/volume/gstvolume.c: (gst_volume_class_init):
Simple cleanup.
2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
(gst_text_overlay_init), (gst_text_overlay_set_property),
(gst_text_overlay_get_property):
Revert API/ABI break from March 1. Keep 'halign' and 'valign'
as string type properties, but mark them deprecated. Add
'halignment' and 'valignment' properties that use enums
instead of strings.
2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Allow palettes with less than 256 colours in AVI files
(#333488, patch by: Fabrizio Gennari).
2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
Original commit message from CVS:
2006-03-07 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
(gst_text_overlay_video_event): Fix wrong EOS handling on text
pad. We were releasing the queued text buffer when we should keep
it until video pad gets EOS or discard the text buffer because it's
too old. That was eating the last subtitle buffer. Add some more
debug.
2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
(gst_text_overlay_video_chain):
Fix invalid memory access (we can't access a buffer after it's been
pushed downstream without taking a reference); fix memory leak (if
there's no text to render, bail out before allocating stuff).
2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
* ext/pango/gsttextoverlay.h:
If input is plain text, escape it before passing it to
pango_layout_set_markup().
2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
Don't ignore flow return from gst_pad_push().
2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
Don't leak references returned by gst_pad_get_parent()
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_getcaps),
(gst_visual_src_setcaps), (gst_visual_sink_setcaps):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
(gst_vorbisenc_convert_sink):
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
(gst_audio_filter_chain):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps):
* gst-libs/gst/video/video.c: (gst_video_frame_rate),
(gst_video_get_size):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Don't leak references returned by gst_pad_get_parent()
(#333663, based on patch by: Christophe Fergeau).
2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/gnomevfs/gstgnomevfssink.c: change location param details
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
change location param details
* gst/volume/gstvolume.c: (plugin_init):
correct plugin description
2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
(gst_gnome_vfs_src_check_get_range):
Override GstBaseSrc::check_get_range() in order to avoid opening
the resource just to check whether we can operate in pull-mode or
not - we can predict that pretty well from the URI alone. Should
fix problems with last.fm (#331690). (Requires latest core CVS).
2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
Original commit message from CVS:
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
(gst_video_sink_class_init):
Throw away frames that are later than 20 ms.
2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
gst-libs/gst/riff/riff-media.c:
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
put Theora BOS pages before others. This hardcodes
the Ogg/Theora I profile, but hey.
2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
changed more than 5 lines
Original commit message from CVS:
changed more than 5 lines
2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
Original commit message from CVS:
ogg muxing of vorbis and theora now has pages ordered correctly again,
even with delays.
* ext/ogg/README:
updated with some examples
* ext/theora/theoraenc.c: (granulepos_to_timestamp),
(granulepos_add), (theora_buffer_from_packet):
* ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
(granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
(gst_vorbisenc_chain):
implement strategy from ext/ogg/README
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
(gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
(gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
Fix muxer so that oggz-validate is happy with all streams;
except for no eos mark, and the BOS page ordering
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
(check_buffer_granulepos):
* tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
update tests to check for OFFSET being set as requested
fixed type of granulepos, it's not a ClockTime
2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
Original commit message from CVS:
2006-03-05 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
Check that the xvimage we are creating has a correct size before returning it. (#314897)
2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Give id3 and ape tag typefinders a rank slightly higher
than PRIMARY to ensure they're always run before any of
the other typefinders (in particular wav and mp3) (#324186).
2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for '3IVD' fourcc (#333403).
2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
configure.ac: Bump requirements to GStreamer CVS for the new error enum.
Original commit message from CVS:
* configure.ac:
Bump requirements to GStreamer CVS for the new error enum.
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
space left on the device (fixes #333352).
2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/vs6: add a project file for libgstvolume update the workspace
Original commit message from CVS:
* win32/vs6:
add a project file for libgstvolume
update the workspace
2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/README:
* ext/ogg/gstoggmux.c:
debug updates
Original commit message from CVS:
debug updates
2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
Original commit message from CVS:
2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
* ext/theora/theoraenc.c: (theora_set_header_on_caps):
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
(GST_START_TEST):
Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
Set IN_CAPS on header buffers
2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/plugins/: Add audioresample to docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add audioresample to docs.
* gst/audioconvert/gstaudioconvert.c:
Add revision date.
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_base_init), (gst_audioresample_class_init),
(gst_audioresample_init), (gst_audioresample_dispose),
(audioresample_get_unit_size), (audioresample_transform_caps),
(resample_set_state_from_caps), (audioresample_transform_size),
(audioresample_set_caps), (audioresample_event),
(audioresample_do_output), (audioresample_transform),
(audioresample_pushthrough), (gst_audioresample_set_property),
(gst_audioresample_get_property), (plugin_init):
* gst/audioresample/gstaudioresample.h:
Added docs.
Small code cleanups.
2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/videorate/Makefile.am:
fix wim's commit
Original commit message from CVS:
fix wim's commit
2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
debug using the actual GstPad, that allows us to see the serialno in the padname
Original commit message from CVS:
debug using the actual GstPad, that allows us to see the serialno in the padname
2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/plugins/: Added videoscale to docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added videoscale to docs.
* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain):
Fix typo in docs.
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_init), (gst_video_scale_prepare_size),
(gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
(gst_video_scale_fixate_caps), (gst_video_scale_transform):
* gst/videoscale/gstvideoscale.h:
Added docs, examples.
Some code cleanups.
Post errors instead of g_warning.
2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
clean up debug messages
Original commit message from CVS:
clean up debug messages
2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
extra debugging from older version, makes it easier to compare
Original commit message from CVS:
extra debugging from older version, makes it easier to compare
2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ext/ogg/gstoggmux.c:
some space cleanup and debug fixes
Original commit message from CVS:
some space cleanup and debug fixes
2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/: Added some more docs to libs and plugins.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes #331903)
Handle EOS correctly.
Added docs.
2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
In state change function, first chain up to parent class,
then handle downwards state change stuff. Remove some
commented out cruft from 0.8 code.
2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
(gst_ogm_parse_chain):
Don't remove/re-add source pad if the new caps are the same as
the old caps anyway (#333042). When removing source pad, don't
unref it afterwards - we didn't ref it when adding. Sprinkle some
GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
after using gst_pad_get_parent(). Return downstream flow return
value in chain function.
2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
docs/plugins/: Fix hierarchy, added some more elements to the docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.signals:
Fix hierarchy, added some more elements to the docs.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_type):
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
Fix docs for ffmpegcolorspace.
2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (id3_type_find),
(apetag_type_find), (ape_type_find), (plugin_init):
Some typefinding fine-tuning:
- rank ID3/APE tags in order of preference via probabilities, so that
ID3v2 > APEv2 > APEv1 > ID3v1.
- three or four bytes don't really justify MAXIMUM probability,
change those to 'very likely' (musepack and monkeysaudio).
2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
Added alsa docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasink.h:
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
(gst_alsasrc_init):
* ext/alsa/gstalsasrc.h:
Added alsa docs.
Small code cleanups.
2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/Makefile.am: Dist new header too,
Original commit message from CVS:
* ext/theora/Makefile.am:
Dist new header too,
2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
Fix some more docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
* ext/vorbis/vorbisenc.h:
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
(vorbis_parse_chain), (vorbis_parse_change_state):
* ext/vorbis/vorbisparse.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/tcp/gsttcpserversink.h:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Fix some more docs.
Added docs for vorbisdec and vorbisparse.
Fix vorbisparse.
2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
Updated/added documentation.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/pango/gstclockoverlay.h:
* ext/pango/gsttextoverlay.h:
* ext/pango/gsttextrender.h:
* ext/pango/gsttimeoverlay.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* gst/audioconvert/gstaudioconvert.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gstmultifdsink.h:
Updated/added documentation.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
(gst_text_overlay_halign_get_type),
(gst_text_overlay_wrap_mode_get_type),
(gst_text_overlay_base_init), (gst_text_overlay_class_init),
(gst_text_overlay_init), (gst_text_overlay_set_property),
(gst_text_overlay_get_property):
Fix up properties to be enums instead of string to make bindings,
introspection and automatic GUI creation possible.
Add getters for the properties.
2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
added defines of M_PI and M_PI_2
* gst/ffmpegcolorspace/avcodec.h:
removed #include "stdint.h" for win32 as _stdint.h is
autogenerated to win32/common
* win32/common/libgstaudio.def:
* win32/common/libgsttag.def:
added some exports
* win32/vs6:
some project files bugs corrected
* win32/vs7:
project files are reset to the default vs7 configuration
(they link to msvcr71.dll using default optimizations)
2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
Fix some docs.
2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
Original commit message from CVS:
* ext/alsa/gstalsasrc.c:
Set proper class on the ElementDetails:
Source/Audio instead of Src/Audio
2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
Original commit message from CVS:
* gst/videoscale/vs_scanline.c:
(vs_scanline_resample_nearest_RGBA):
Revert optimization in videoscale. It should go in liboil and have
an appropriate liboil function.
2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Don't try to provide a clock in the NULL state.
2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
(gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
(gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
(gst_ogg_demux_loop), (gst_ogg_demux_change_state):
Use GstSegment infrastructure to remove duplicated code
and handle more seek cases correctly.
2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform):
Don't ignore return code from ffmpeg convert function.
* gst/ffmpegcolorspace/imgconvert.c: (img_convert):
Split out some long statements to ease debugging.
2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_init),
(gst_vis_src_negotiate), (get_buffer), (plugin_init):
Don't use gst_pad_use_fixed_caps, because it prevents downstream from
being able to renegotiate the size. Instead, use the negotiation
algorithm from the goom plugin to pick an initial output caps.
Also, allow theoretical libvisual plugins that might support non-GL
output even if they also do GL.
2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
Original commit message from CVS:
2006-02-26 Julien MOUTTE <julien@moutte.net>
* ext/libvisual/visual.c: (gst_visual_init),
(gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
(plugin_init): Load only non GL plugins. Fix some memleaks and
possible negotiation issues.
2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
gst-libs/gst/tag/tag.h: Adding Annodex tags here.
Original commit message from CVS:
2006-02-25 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/tag/tag.h: Adding Annodex tags here.
2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
(cmml_type_find), (plugin_init):
Fix CMML type find function to not require a specific minor version
of the CMML header.
Add an MPEG4 video elementary stream typefind function.
2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
(gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
(gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
(gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
Annodex support in ogg demuxer. Doesn't do very much without the
other annodex patches (to come).
2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c:
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Pick up palette for MS video v1 (#327028, patch by:
Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_caps_remove_format_info),
(gst_ffmpegcsp_get_unit_size):
The 'palette_data' field from incoming RGB caps shouldn't be
proxied on outgoing YUV caps; also, restrict unit size
adjustment in case of paletted data only to the unit that
actually has a palette. Fixes #330711.
2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
(gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
(gst_ffmpegcsp_get_unit_size):
Plug some memory leaks.
2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
Original commit message from CVS:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
Add some _CFLAGS and _LIBS that seem to be missing
and/or required for Cygwin (see #317048).
2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
* ChangeLog:
ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
Original commit message from CVS:
ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
Original commit message from CVS:
* ext/alsa/gstalsasrc.c:
Fix description as pointed out by caugier.
2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
Original commit message from CVS:
Reviewed by : Edward Hervey <edward@fluendo.com>
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(qt_type_find):
Better 3gp typefinding.
2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
Don't send EOS event here, the base class will send one for us.
* gst/playback/gstplaybasebin.c: (prepare_output):
Subpictures without video stream aren't allowed either.
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Fix debug statement copy'n'paste-o.
2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
Fix issues with mixer keeping state when muting/unmuting
and when changing the volume whilst muted (see #331763
and #331765).
2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
(parse_subrip), (gst_sub_parse_format_autodetect):
Set right caps given that we send escaped text. Also,
honour <i></i>, <b></b> and <u></u> markers that can be found
in .srt files (fixes #310202).
2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.c:
(element_factory_rank_compare_func):
Make order in which elements are tried more determinable.
2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (get_our_ghost_pad),
(remove_element_chain), (cleanup_decodebin),
(gst_decode_bin_change_state): Make decodebin reusable by
fixing remove_element_chain first and then introduce a
cleaner in state change to ->NULL. (Closes #331678)
------------------------------------------------------
2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
use 0666 mask when creating files so umask gets applied
correctly. Fixes #331295.
2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
Original commit message from CVS:
* gst/subparse/Makefile.am:
* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
(gst_ssa_parse_dispose), (gst_ssa_parse_init),
(gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
(gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
(gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
(gst_ssa_parse_chain), (gst_ssa_parse_change_state):
* gst/subparse/gstssaparse.h:
* gst/subparse/gstsubparse.c: (plugin_init):
Add very basic parser for SSA subtitle streams (as often
found in matroska files).
2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (mimetype_is_raw):
That should be text/x-pango-markup, not text/x-pango-layout.
2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Polishing.
Original commit message from CVS:
2006-02-19 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
Polishing.
2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Fix state change deadlock.
Original commit message from CVS:
2006-02-19 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_finalize), (gst_text_overlay_init),
(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
(gst_text_overlay_render_text),
(gst_text_overlay_text_pad_link),
(gst_text_overlay_text_event), (gst_text_overlay_video_event),
(gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
(gst_text_overlay_video_chain), (gst_text_overlay_change_state):
Fix state change deadlock.
2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
Original commit message from CVS:
2006-02-19 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_finalize), (gst_text_overlay_init),
(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
(gst_text_overlay_render_text),
(gst_text_overlay_text_pad_link),
(gst_text_overlay_text_event), (gst_text_overlay_video_event),
(gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
(gst_text_overlay_video_chain), (gst_text_overlay_change_state):
* ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
and subtitles files.
2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
gst/playback/gstdecodebin.c: pango layout should be considered as row.
Original commit message from CVS:
2006-02-19 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
should be considered as row.
2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
gst/playback/gststreaminfo.*: Introduce language informations.
Original commit message from CVS:
2006-02-19 Julien MOUTTE <julien@moutte.net>
* gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
(cb_probe):
* gst/playback/gststreaminfo.h: Introduce language informations.
2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
Set shared memory segments to be deleted as soon as we have attached,
that way they get cleaned up automatically if we crash.
2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
ext/pango/: Those functions are called with lock held.
Original commit message from CVS:
2006-02-18 Julien MOUTTE <julien@moutte.net>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
* ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
functions are called with lock held.
2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
* ChangeLog:
Forgot Changelog.
Original commit message from CVS:
Forgot Changelog.
2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
Original commit message from CVS:
2006-02-18 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_finalize), (gst_text_overlay_init),
(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
(gst_text_overlay_render_text),
(gst_text_overlay_text_pad_link),
(gst_text_overlay_text_pad_unlink),
(gst_text_overlay_text_event),
(gst_text_overlay_video_event), (gst_text_overlay_pop_text),
(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
(gst_text_overlay_change_state): Refactoring of textoverlay
without collectpads. This now supports sparse subtitles coming
from a demuxer instead of a sub file. Seeking is still broken
though. Need to discuss with wtay some more on how to handle
seeking correctly.
* ext/pango/gsttextoverlay.h:
* gst/playback/gstplaybin.c: (setup_sinks): Support linking with
subtitles coming from the demuxer.
2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisenc.c: Use some more scaling functions.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
(gst_vorbisenc_convert_sink):
Use some more scaling functions.
2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
(gst_cd_paranoia_paranoia_callback),
(gst_cd_paranoia_src_signal_is_being_watched),
(gst_cd_paranoia_src_read_sector):
* ext/cdparanoia/gstcdparanoiasrc.h:
Add back 'transport-error' and 'uncorrected-error' signals and
make them actually be fired when bad stuff happens (#319340).
2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
(gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
(gst_ring_buffer_clear):
Small cleanups.
Added some G_LIKELY.
2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/TODO: Update TODO
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Update TODO
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset):
When trying to play samples ASAP and we don't have a
previous sample, try to play at position 0 instead of
an invalid position.
2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open),
(gst_alsasink_reset):
Also release lock when we get an error in _reset();
fix an error message.
2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
(gst_alsasink_init), (get_channel_free_structure),
(caps_add_channel_configuration), (gst_alsasink_getcaps),
(gst_alsasink_close):
* ext/alsa/gstalsasink.h:
Add support for more than 2 channels (#326720).
2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
with 4 or 6 channels, assume a default channel layout to make things
work (not sure there's anything else we can do in those cases).
2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: Minor docs fix.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Minor docs fix.
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
Add support for WAVEFORMATEX, eg. PCM audio with more than two
channels and a channel layout map.
2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
Original commit message from CVS:
Reviewed by Edward Hervey <edward@fluendo.com>
* gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
C-level optimization of the RGBA nearest neighbour function.
Eventually this might end up in liboil with vectorized versions.
2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions):
When we have more than 2 channels, but no channel layout is
specified in the caps, return some default channel layout
to the caller and warn about about a possibly buggy element
(could be buggy filtercaps as well of course) (#317038).
2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
Original commit message from CVS:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
Add gst-libs/gst/cdda to list of lib search paths.
2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
Original commit message from CVS:
2006-02-15 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
to the Lord Jesus that I do not have to touch the ogg muxer ever
again.
2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
quicktime movie files can also contain 'uuid' atoms.
2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
Original commit message from CVS:
* gst/audioconvert/plugin.c: (plugin_init):
Register the GstAudioChannelPosition enum type with the type
system in the plugin_init function, so that it is known before
any element actually makes use of multi-channel stuff. This is
required for example if one wants to be able to deserialise/use
a caps string with channel positions before any pipeline has
been setup and started, like with gst-launch.
2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
(gst_ring_buffer_samples_done), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_clear):
Add some compiler G_(UN_)LIKELY help.
SIGNAL the ringbuffer waiters when going to PAUSED as well to
make sure they can exit their functions. Should fix #330748
2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
Windows does not have long long; copy the generated _stdint.h
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* win32/MANIFEST:
* win32/common/_stdint.h:
Windows does not have long long; copy the generated _stdint.h
* win32/common/interfaces-enumtypes.c:
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
(gst_mixer_track_flags_get_type),
(gst_tuner_channel_flags_get_type):
* win32/common/multichannel-enumtypes.c:
(gst_audio_channel_position_get_type):
update
2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Always sync on first sample we receive when starting.
2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/playback/gstplaybin.c: Update vis bin docs.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_vis_element):
Update vis bin docs.
Move queue after tee so we don't queue video buffers but
audio samples instead. Fixes problems where the video queue
is filled and the audio queue empty.
2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
No need to push an EOS event here, GstBaseSrc will do that for us
when we return FLOW_UNEXPECTED.
2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Use scale functions when possible.
Fix error messages.
Free clockid when after waiting for EOS.
Use G_(UN_)LIKLY when it makes sense.
Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (prepare_output):
Remove stray semi-colon (fixes #330888).
2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
sys/: Fix up the XShm call testing so that we catch errors, and don't cause new ones by attempting to detach from a s...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
Fix up the XShm call testing so that we catch errors, and don't
cause new ones by attempting to detach from a segment we failed
to attach to. Fixes #312439.
2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Added flv file typefind (video/x-flv).
2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
Also added the caps to the default set of riff video caps.
2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
Original commit message from CVS:
2006-02-09 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
time and the end time of the last packet in the page.
(gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
on the pages in our queue, set the duration as well. Reflow a
debug statement.
(gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
Fixes bad muxing order.
2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_setcaps), (gst_basertppayload_push):
update seqnum before setting it on the packet; this makes sure
that the timestamp and seqnum properties match after pushing
a buffer
2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
changelog foo
Original commit message from CVS:
changelog foo
2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c:
* win32/common/config.h:
kapowpowpow
Original commit message from CVS:
kapowpowpow
2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
gst-libs/gst/audio/gstringbuffer.c
Original commit message from CVS:
2006-02-09 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c
(gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
overflow after 13.5 hours of recording. Kapow!
* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
the buffer size -- we don't care about underrun/overrun reporting
right now, just need to return a useful value.
2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
=== release 0.10.3 ===
2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/config.h:
Releasing 0.10.3
Original commit message from CVS:
Releasing 0.10.3
2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
configure.ac: Drat. Bump libtool version number for new API.
Original commit message from CVS:
* configure.ac:
Drat. Bump libtool version number for new API.
Prelease 0.10.2.3 (of 0.10.3)
2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
0.10.2.2 prerelease (of 0.10.3).
Original commit message from CVS:
* configure.ac:
* win32/common/config.h:
0.10.2.2 prerelease (of 0.10.3).
2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
Original commit message from CVS:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
Revert Andy's newsegment change pending a more correct
fix.
2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update .po files
Original commit message from CVS:
Update .po files
2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst/tcp/gstmultifdsink.c:
doc fixes
Original commit message from CVS:
doc fixes
2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
Original commit message from CVS:
:
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(qt_type_find), (plugin_init):
detect more files as 3gp
group and reorder the iso file formats
2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
Original commit message from CVS:
* ext/vorbis/vorbis.c: (plugin_init):
Register musicbrainz tags, so apps don't have to.
2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
(gst_tag_to_vorbis_tag):
Make sure we called gst_tag_register_musicbrainz_tags()
before possibly mapping a vorbiscomment string from/to a
musicbrainz tag.
2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
In case we can't find the required number of consecutive
mpeg audio frames to positively identify an MPEG audio
stream, check if there's at least a valid mpeg audio
frame right at offset 0 and if so suggest mpeg/audio
caps with a very low probability (#153004).
2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
Original commit message from CVS:
2006-02-07 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
a TIME segment if we get timestamped buffers. Requires recent
fixes in core to work properly.
2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (prepare_output):
Don't print the URI as part of the error message, it
makes error dialogs look rather ugly, especially if
the URI is very long or has characters in it that
need escaping.
2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (prepare_output):
Error out if we have only text or subtitles, but nothing
else. Also error out if we have subtitles but no video
stream.
2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
Post an error message on the bus when we encounter an
error, which will hopefully be more meaningful than the
'Internal Flow Error' message users get to see if we
just return GST_FLOW_ERROR.
2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
Original commit message from CVS:
2006-02-07 Andy Wingo <wingo@pobox.com>
* configure.ac (GST_MAJORMINOR): Update core version req to
0.10.2.2, for the collectpads API addition (#330244).
2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
Return FALSE from plugin_init() when GnomeVFS can't
be initialised for some reason (#328423).
2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
Original commit message from CVS:
2006-02-06 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
Stick to seeking theory until i find the bug.
* gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Make theoraenc and the tests leak free. Like, really.
Original commit message from CVS:
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
(theora_enc_finalize), (theora_enc_sink_setcaps),
(theora_set_header_on_caps), (theora_enc_chain),
(theora_enc_change_state):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
Make theoraenc and the tests leak free. Like, really.
2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
Original commit message from CVS:
(theora_enc_finalize), (theora_enc_sink_setcaps):
Add a finalize method to ensure we clean up state even if
someone omitted the state change back to NULL.
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
(gst_vorbisenc_chain):
Free some more leaked bits.
* tests/check/pipelines/theoraenc.c: (start_pipeline),
(stop_pipeline):
Wait for state changes to happen if they're ASYNC.
This ought to teach those fancy pants buildbots a lesson.
2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add mapping for ID3 International Standard Recording Code
tag "TSRC"
2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/vorbis/vorbisenc.c: Don't leak tag names.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
Don't leak tag names.
2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
Split libgsttag docs into multiple sections.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tags.c:
Split libgsttag docs into multiple sections.
2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
Add libgsttag to the docs.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
Add libgsttag to the docs.
2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
ext/pango/gsttextoverlay.c: Fix clockoverlay.
Original commit message from CVS:
2006-02-05 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
(gst_text_overlay_init), (gst_text_overlay_src_event),
(gst_text_overlay_collected): Fix clockoverlay.
2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
Original commit message from CVS:
* docs/libs/compiling.sgml:
Fix typo: it's pkg-config, not pkg-gconfig
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/tmpl/gstgconf.sgml:
There is no libgstgconf in 0.10, remove it
from the docs.
2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
docs/libs/tmpl/gstcolorbalance.sgml: Updated.
Original commit message from CVS:
2006-02-05 Julien MOUTTE <julien@moutte.net>
* docs/libs/tmpl/gstcolorbalance.sgml: Updated.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_src_event), (gst_text_overlay_collected):
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
(parse_mpsub), (parser_state_init), (handle_buffer),
(gst_sub_parse_chain), (gst_sub_parse_sink_event),
(plugin_init):
* gst/subparse/gstsubparse.h: Introduce seeking code.
2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Add comment about LANGUAGE tag inconsistency (we want
ISO-639-1, but extract three-letter identifiers?)
* po/POTFILES.in:
Add two translatable files.
2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
(gst_tag_register_musicbrainz_tags_internal),
(gst_tag_register_musicbrainz_tags):
Forward-port some tags stuff from the 0.8 branch. This is
mostly the addition of musicbrainz tags and their mapping
to vorbistags, and a vorbistag mapping of the language tag.
2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
gst/playback/gstplaybin.c: Fix broken code refactoring.
Original commit message from CVS:
2006-02-05 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
refactoring.
2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
Add Dirac typefinding and add dirac format to oggmux.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
* gst/typefind/gsttypefindfunctions.c:
Add Dirac typefinding and add dirac format to oggmux.
2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
* configure.ac:
Improve error message for liboil missingness.
Original commit message from CVS:
Improve error message for liboil missingness.
2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Don't put essential function call into
g_return_*() macro, otherwise it'll all be
replaced by NOOPs when compiling with
G_DISABLE_CHECKS defined.
2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
* ChangeLog:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggparse.c:
* gst/tcp/gsttcpserversink.c:
* sys/v4l/v4lsrc_calls.c:
* sys/v4l/v4lsrc_calls.h:
Just make it compile with --disable-gst-debug.
Original commit message from CVS:
Just make it compile with --disable-gst-debug.
2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init), (gst_alsasink_init),
(gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Add lock to protect alsa calls.
Implement reset to flush samples ASAP, does not work
with dmix though.
2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Ugh.. getting late I guess...
2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
Don't try to provide a clock when we are not negotiated since
we might not be able to make it run.
2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Unlinking two source pads is ... hard.
2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/TODO: Updated.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
On EOS, wait till the last sample is played before posting EOS.
2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* tests/check/pipelines/theoraenc.c:
comment on my understanding
Original commit message from CVS:
comment on my understanding
2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* common:
* tests/check/pipelines/theoraenc.c:
reformat to fit 80 chars
Original commit message from CVS:
reformat to fit 80 chars
2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
Original commit message from CVS:
2006-02-01 Philippe Kalaf <burger at speedy dot org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
setting queue_delay to zero. Also avoid thread being started if
queue_delay is zero.
2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
Original commit message from CVS:
* gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
Make test work again by connecting fakesinks to each decoded pad,
which makes the pipeline wait until each fakesink has a buffer
queued before going to PAUSED state. At that point we know the
decodebin pads are negotiated.
2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/: Pass unhandled queries to the parent class's query function.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
(gst_cdda_base_src_handle_event):
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Pass unhandled queries to the parent class's query function.
2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
(gst_ogg_pad_src_query):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_sink_query):
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_query):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
(gst_vorbisenc_sink_query):
* gst/adder/gstadder.c: (gst_adder_query):
Pass unhandled queries upstream instead of just
dropping them (#326447). Also, fix supported
query types list for some elements.
2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (au_type_find),
(paris_type_find), (ilbc_type_find), (plugin_init):
Fix typefinding for audio/x-au, audio/x-paris and
audio/iLBC-sh. We cannot use the START_WITH macros
here, because there can only be one typefind factory
with the same name (caps), so the second one would
replace the first one and the first one would never
be called when doing typefinding (see #161712).
2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
(vorbis_handle_header_packet), (vorbis_dec_push),
(vorbis_handle_data_packet):
Use scale_int when we can, add some more scaling.
Check packettype before parsing it.
2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Call right _scale functions.
Original commit message from CVS:
* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_src_convert), (theora_dec_sink_convert):
Call right _scale functions.
Use parameter instead of some other random value.
2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/theora/theoradec.c: Use higher precision timestamps calculation.
Original commit message from CVS:
* ext/theora/theoradec.c: (_theora_granule_frame),
(_theora_granule_time), (_inc_granulepos),
(theora_dec_src_convert), (theora_dec_sink_convert),
(theora_handle_type_packet), (theora_handle_data_packet),
(theora_dec_chain):
Use higher precision timestamps calculation.
Convert some other conversions to _scale.
2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/: initialize gst_controller before using
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_create_sine_table), (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
initialize gst_controller before using
2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisenc.c:
Define constant using G_GINT64_CONSTANT to avoid errors when
passing it around - otherwise it gets truncated to 32 bits.
Fixes failing tests.
2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
Original commit message from CVS:
2006-01-31 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
caps being set doesn't have a framerate value. Basically a stopgap
measure.
* ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
technically correct enough to put into core though.
(gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
DURATION. Fixes theoraenc ! oggmux.
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
fraction, not double.
2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update with latest files
Original commit message from CVS:
update with latest files
2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/vs7: add vs7 project files created by Sergey Scobich
Original commit message from CVS:
* win32/vs7:
add vs7 project files created by Sergey Scobich
2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
win32/vs8: add vs8 project files created by Sergey Scobich
Original commit message from CVS:
* win32/vs8:
add vs8 project files created by Sergey Scobich
2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
timestamp + duration, not just timestamp -- ogg pages should be
ordered by stop time. Necessary fix given the change in vorbis
timestamps.
2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
* ChangeLog:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
* tests/check/pipelines/theoraenc.c:
ext/theora/theoraenc.c (theora_enc_sink_setcaps)
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/theora/theoraenc.c (theora_enc_sink_setcaps)
(gst_theora_enc_init): Pull the granule shift out of the encoder.
(granulepos_add): New function, handles the messiness of adjusting
granulepos values.
(theora_buffer_from_packet):
(theora_enc_chain):
(theora_enc_sink_event): Use granulepos_add, not +.
* tests/check/pipelines/theoraenc.c
(check_buffer_granulepos_from_starttime): Just check the frame
count, not the actual granulepos -- we can't dictate to the
encoder when it should be placing keyframes.
2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
SERVICE_NOT_AVAILABLE happens for example when you're trying to
play an http:// stream from a server that's not serving
2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
* tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
remove the UINT64_CONSTANT macro, doesn't appear to be needed or
available.
2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
although theoraenc was timestamping correctly. Added handling of
streams that start with nonzero timestamps.
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c: New file, basically does same
tests as vorbisenc.
* tests/check/pipelines/vorbisenc.c: I claim these bugs.
2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.
2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
* ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
updated to timestamp from the first sample, not the last.
(gst_vorbisenc_buffer_from_header_packet): New function, takes
special care of granulepos and timestamp for header packets.
(gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
when the first buffer has a nonzero timestamp.
* ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
(GstVorbisEnc.subgranule_offset): New members. Take care of the
case when the first audio buffer we get has a nonzero timestamp.
(GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
properly timestamp vorbis buffers with the time of the first
sample, not the last.
* ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
vorbis_granule_time_copy -- now it takes the granule/subgranule
offset into account.
* tests/check/pipelines/vorbisenc.c: New test for correctness of
timestamps, durations, and granulepos on buffers produced by
vorbisenc.
2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Patch from Eric Jonas to support conversions to/from UYVY
(Fixes: #324626)
2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
gst/playback/: Implement subtitles.
Original commit message from CVS:
2006-01-30 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_overrun),
(setup_subtitle), (setup_source), (set_active_source):
* gst/playback/gstplaybin.c: (gst_play_bin_dispose),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
add a win32-update rule like in core, and copy over enumtypes files
Original commit message from CVS:
* Makefile.am:
* win32/MANIFEST:
* win32/common/interfaces-enumtypes.c:
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
(gst_mixer_track_flags_get_type),
(gst_tuner_channel_flags_get_type):
* win32/common/interfaces-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
(gst_audio_channel_position_get_type):
* win32/common/multichannel-enumtypes.h:
add a win32-update rule like in core, and copy over enumtypes files
2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
generate win32/common/config.h
Original commit message from CVS:
generate win32/common/config.h
2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
win32/: add config files just like in core
Original commit message from CVS:
* win32/MANIFEST:
* win32/common/config.h:
* win32/common/config.h.in:
add config files just like in core
2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
(set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
(gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
(set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
(gst_alsasrc_unprepare), (gst_alsasrc_read):
Update all error messages. All of them should either use
the default translated message, or actually provide a
translatable string.
Make the string for channel count problems meaningful.
2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
Make gcc-4.1 happy (part of #327357).
2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
Original commit message from CVS:
* sys/v4l/v4l_calls.c: (gst_v4l_open):
check for and throw RESOURCE_BUSY
2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
Original commit message from CVS:
* gst/videoscale/vs_scanline.c: Oops, *that's* why I never
checked in this change -- it requires liboil features not
in 0.3.6. Revert parts.
2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
update liboil requirement to 0.3.6
Original commit message from CVS:
* REQUIREMENTS:
* configure.ac: update liboil requirement to 0.3.6
* gst/videoscale/Makefile.am:
* gst/videoscale/vs_scanline.c: liboilify
2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
When pad_alloc returns a GstFlowReturn other
than GST_FLOW_OK, make sure it is passed upstream.
2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/alsa/gstalsasink.c: Free the device name string.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init):
Free the device name string.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
(gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
Don't remove a pad from the collectpads structure until it
is released - it's a request pad, and may receive data again
if the element gets moved back to PLAYING state.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Ensure we turn on double buffering on the Xv port, and
set the colour key to something dark and mysterious that
isn't black.
2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/: - a library should not call setlocale. see Libraries node in gettext manual
Original commit message from CVS:
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_base_init), (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- a library should not call setlocale. see Libraries node in
gettext manual
- make sure all plugins that use translation do bindtextdomain
to point to the localedir
* gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
(setup_sinks), (plugin_init):
all this, and check for NULL when creating sinks
2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
Original commit message from CVS:
2006-01-27 Julien MOUTTE <julien@moutte.net>
* gst/subparse/gstsubparse.c: (gst_subparse_type_find),
(plugin_init): Make typefinding of subtitles work again.
2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(wavpack_type_find), (m4a_type_find), (ircam_type_find),
(plugin_init):
Backport a bunch of typefinding fixes from the 0.8 branch.
Also, improve wavpack typefinding: if we can't peek the
entire wavpack block, try to parse the bits we can get and
see if we find what we're looking for in those.
2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
sys/: Handle some more cases of pixel aspect ratio.
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c:
(gst_ximagesink_calculate_pixel_aspect_ratio):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
more cases of pixel aspect ratio.
2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (pad_probe):
Also consider the flush-start and tag events as unblockers
for the pad probes.
2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstplaybin.c: (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
On the fly visualisation switch, works disabling, enabling as
well but it won't be able to enable vis in a playbin that was
created with no visualisation.
2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Undo previous commit, it breaks resume after pause.
2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
Improve debugging.
Post error when caps cannot be parsed.
Resync on discontinuity in the stream.
Clip samples to segment boundaries.
return WRONG_STATE sooner when we are flushing.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
Make audiosrc operate in TIME.
Set TIMESTAMP and DURATION on buffers.
2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/examples/seek/seek.c: Output tag messages as well.
Original commit message from CVS:
* tests/examples/seek/seek.c: (main):
Output tag messages as well.
2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(free_pad_probes), (remove_fakesink), (pad_probe),
(close_pad_link), (gst_decode_bin_change_state):
Replace GstPadBlockCallback with pad probes that detect
first buffer AND eos before removing fakesink.
Fixes hang with demuxers doing EOS while pre-rolling.
Solves #328279
2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
GCC 2.95 fixes (#328263).
Original commit message from CVS:
2006-01-23 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsasink.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
Patch by: Jens Granseuer <jensgr at gmx dot net>
2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
Original commit message from CVS:
2006-01-22 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
frames. We might get a frame destroyed after changing state to
NULL, adding a safety check on xcontext.
2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Fix prepare-xwindow-id code example in the docs - we need to
ignore all messages that aren't element messages as well.
2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
Original commit message from CVS:
2006-01-21 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
I think one day i'll completely undestand how caps negotiation
is supposed to work. This refactoring handles buffer_alloc
called with caps we can't handle. We definitely don't want a
set_caps with those caps, so we define and allocate a buffer
we would like to receive.
2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
* autogen.sh:
* common:
up automake requirement to 1.7
Original commit message from CVS:
up automake requirement to 1.7
2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Free iterator when done.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Free iterator when done.
2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.
Makes this work again:
gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
endianness=(int)4321, signed=(boolean)true, width=(int)16,
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
audioresample ! alsasink
2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
back to HEAD
Original commit message from CVS:
back to HEAD
=== release 0.10.2 ===
2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
releasing 0.10.2
Original commit message from CVS:
releasing 0.10.2
2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/: Comment out broken code that connects to the state-changed signal.
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property):
Comment out broken code that connects to the state-changed signal.
At this point, changing current stream selection is broken, but
stuff like gst-launch playbin current-audio=1 works and filters
to the chosen stream.
2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
Fix #327216 (null dereference in vorbisdec)
2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_comment_packet):
Post taglist actually on bus instead of just freeing it
(fixes #327114 and totem bug #327080).
* ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
Use gst_element_found_tags_for_pad(), so that the tags
are sent downstream as an event as well.
2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
sys/: move all regularly occurring messages to GST_LOG level add some more object logs
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
(gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
(gst_xvimagesink_buffer_alloc):
move all regularly occurring messages to GST_LOG level
add some more object logs
2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
prerelease
Original commit message from CVS:
prerelease
2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
ext/ogg/gstoggmux.c: fix a silly segfault
Original commit message from CVS:
2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
fix a silly segfault
2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
Add docs for mixerutils stuff.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/mixerutils.c:
* gst-libs/gst/audio/mixerutils.h:
Add docs for mixerutils stuff.
2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Fix playback for sources that emit raw audio or
raw video streams (e.g.: cd audio sources) (#325984).
2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst-libs/gst/audio/mixerutils.c: actually save the element we create
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.c:
(gst_audio_mixer_filter_do_filter):
actually save the element we create
2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
remove version suffix
Original commit message from CVS:
remove version suffix
2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
No need to post a tag message on the bus when seeking
within the same track, only post it when the current
track changes.
2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(probe_triggered), (new_decoded_pad), (mute_group_type),
(set_active_source):
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init),
(gst_stream_selector_set_property),
(gst_stream_selector_request_new_pad):
Reenable stream selection. These mechanisms need a complete overhaul
in the face of 0.8->0.10 changes though.
2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
Change the pad template to src_%d to match the pads that
are created from it. decodebin needs this information in order
to decide that oggdemux is capable of producing multiple pads
(and hence needs queues inserted).
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_collected):
Make debug output more useful by using GST_PTR_FORMAT.
2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
* gst-plugins-base.spec.in:
update spec.in file
Original commit message from CVS:
update spec.in file
2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
Set depth and width for alaw/mulaw (fixes #326601).
2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
tests/icles/Makefile.am: don't build the tests if we don't have the libs
Original commit message from CVS:
* tests/icles/Makefile.am:
don't build the tests if we don't have the libs
2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
(gst_cd_paranoia_paranoia_callback):
Don't try to free NULL pointers.
2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
gst/audiorate/gstaudiorate.c: Add debugging category.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
(gst_audio_rate_change_state), (plugin_init):
Add debugging category.
Fix type issues.
Add case for incoming buffers without valid offset/offset_end.
2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
Don't leak GCond in audio sources.
2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Don't leak an autoaudiosink/alsasink when we generate
a new audio element. (old code, I guess)
2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Support float audio in audiorate.
Use width rather than depth for selecting sample width.
2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.h:
Use GLib types here (that way we don't have to include the
generated _stdint.h header, which makes life easier for win32
folks that don't use autotools for the build) (#325990, patch
by: Sergey Scobich).
2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Name (private) union, makes Forte compiler happy (this time
for real) (#324900).
2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Link against libgstinterfaces, needed for mixer
and property probe stuff.
2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/Makefile.am:
Original commit message from CVS:
* gst-libs/gst/Makefile.am:
2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/mixerutils.c:
(gst_audio_mixer_filter_do_filter),
(gst_audio_mixer_filter_check_element),
(gst_audio_mixer_filter_probe_feature),
(element_factory_rank_compare_func),
(gst_audio_default_registry_mixer_filter):
* gst-libs/gst/audio/mixerutils.h:
Add gst_audio_default_registry_mixer_filter() utility
function.
2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
gst/audioresample/resample.h: As before, but for o_buf
Original commit message from CVS:
* gst/audioresample/resample.h:
As before, but for o_buf
2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
Original commit message from CVS:
* gst/audioresample/resample.h:
Declare struct _ResampleState.buffer as unsigned char *, not void *,
since we do arithmetic on it.
2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Sun's Forte compiler doesn't seem to like anonymous structs,
so use same setup as in GstBaseSrc (fixes #324900).
2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
move old example to tests/examples/volume/volune.c
Original commit message from CVS:
* configure.ac:
* gst/volume/Makefile.am:
* gst/volume/demo.c:
move old example to tests/examples/volume/volune.c
* tests/examples/Makefile.am:
* tests/examples/seek/seek.c: (main):
change window-close event from "delete-event" to "destroy"
* tests/examples/volume/Makefile.am:
* tests/examples/volume/volume.c: (value_changed_callback),
(setup_gui), (message_received), (eos_message_received), (main):
fix event handling and bus usage
2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
(gst_audio_test_src_query), (gst_audio_test_src_create_sine),
(gst_audio_test_src_create_square),
(gst_audio_test_src_create_saw),
(gst_audio_test_src_create_triangle),
(gst_audio_test_src_create_silence),
(gst_audio_test_src_create_white_noise),
(gst_audio_test_src_create_pink_noise),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_create_sine_table),
(gst_audio_test_src_change_wave),
(gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create), (gst_audio_test_src_set_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
update to basesrc changes, implement segmented seeking and eos handling,
add a 'sine-tab' waveform for performance critical playback
2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
po/POTFILES.in: ... and this time the other modified file that I missed last time.
Original commit message from CVS:
* po/POTFILES.in:
... and this time the other modified file that I missed last time.
2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad):
Fix non-C89 variable declaration not at the start of a block. Should
help some compilers.
2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
Original commit message from CVS:
* tests/check/Makefile.am:
And now fix 'make distcheck' (builddir != srcdir)
2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
Original commit message from CVS:
* configure.ac:
* ext/cdparanoia/Makefile.am:
* ext/cdparanoia/gstcdparanoia.c:
* ext/cdparanoia/gstcdparanoia.h:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
(gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
(gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
(gst_cd_paranoia_paranoia_callback),
(gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
(gst_cd_paranoia_src_set_property),
(gst_cd_paranoia_src_get_property), (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.h:
New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
plugin again (there are still fixes required to playbin to make
cdda:// uris work there).
2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/Makefile.am: Fix test case compilation.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix test case compilation.
2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_update_duration),
(gst_cdda_base_src_calculate_cddb_id):
An integer is not a string. Fix access to uninitialised variable.
* tests/check/Makefile.am:
Add cddabasesrc unit test; also actually enable the vorbis test.
* tests/check/generic/states.c:
Blacklist new cd audio elements as well.
* tests/check/libs/cddabasesrc.c:
Unit test for GstCddaBaseSrc (discid calculation mostly).
2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Add docs for libgstcdda/GstCddaBaseSrc.
* gst-libs/gst/interfaces/mixertrack.h:
Do one struct member per line with a semicolon at the end, that way
even gtk-doc might parse it without complaining.
2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
Add new libgstcdda with GstCddaBaseSrc class.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/base64.c:
* gst-libs/gst/cdda/base64.h:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
(gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
(gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
(gst_cdda_base_src_get_property),
(gst_cdda_base_src_get_track_from_sector),
(gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
(gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
(gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
(gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
(gst_cdda_base_src_uri_get_protocols),
(gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
(gst_cdda_base_src_uri_handler_init),
(gst_cdda_base_src_setup_interfaces),
(gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
(cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
(lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
(gst_cdda_base_src_add_tags),
(gst_cdda_base_src_add_index_associations),
(gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
(gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
(gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
(gst_cdda_base_src_create):
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-libs/gst/cdda/sha1.c:
* gst-libs/gst/cdda/sha1.h:
Add new libgstcdda with GstCddaBaseSrc class.
2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.h:
Use GstBaseSinkClass as parent_class member for class struct, not
GstBaseSink.
2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_start):
Add start method to reset running time and number of frames sent
when starting up (fixes #324696; patch by: Michal Benes).
2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.signals:
Add docs stuff for gnomevfssrc and gnomevfssink.
* ext/gnomevfs/gstgnomevfssrc.c:
Fix example pipeline in gtk-doc blurb.
2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
Original commit message from CVS:
* ext/gnomevfs/Makefile.am:
* ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
(gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
(gst_gnome_vfs_handle_get_type), (plugin_init):
* ext/gnomevfs/gstgnomevfs.h:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
(gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
(gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
(gst_gnome_vfs_sink_set_property),
(gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
(gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
(gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
(gst_gnome_vfs_sink_uri_get_type),
(gst_gnome_vfs_sink_uri_get_protocols),
(gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
(gst_gnome_vfs_sink_uri_handler_init):
* ext/gnomevfs/gstgnomevfssink.h:
Port gnomevfssink; add gtk-doc blurb.
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
(gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
(gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_uri_get_type),
(gst_gnome_vfs_src_uri_get_protocols),
(gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
(gst_gnome_vfs_src_uri_handler_init),
(gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
(gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
(gst_gnome_vfs_src_send_additional_headers_callback),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_push_callbacks),
(gst_gnome_vfs_src_pop_callbacks),
(gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
(gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
file; add gtk-doc blurb with example pipelines.
2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
back to HEAD
Original commit message from CVS:
back to HEAD
=== release 0.10.1 ===
2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
releasing 0.10.1
Original commit message from CVS:
releasing 0.10.1
2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
* ChangeLog:
* gst/typefind/gsttypefindfunctions.c:
iLBC30 and iLBC20 added to typefind.
Original commit message from CVS:
iLBC30 and iLBC20 added to typefind.
2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* configure.ac:
* docs/libs/tmpl/gstcolorbalance.sgml:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
prereleasing
Original commit message from CVS:
prereleasing
2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
stop making fun of older compilers
Original commit message from CVS:
stop making fun of older compilers
2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
Made a quack, forgot to add DUCK to the riff video template.
2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
(gst_ogm_parse_chain):
Make sure pads are initialized correctly.
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add a whole bunch of FOURCC <=> MimeType.
Extend the riff video pad template to support the newly added fourcc.
2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain):
Extra debug output when activating/deactivating chains.
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
(is_demuxer_element), (try_to_link_1), (remove_element_chain),
(unlinked):
Remove a queue from our list when it becomes unlinked.
Don't add queues to elements in class 'Demux' if they
can only produce one pad
2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
gst-libs/gst/video/gstvideosink.c: Add a debug category.
Original commit message from CVS:
2005-12-18 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
(gst_video_sink_get_type): Add a debug category.
2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
Original commit message from CVS:
2005-12-17 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
Handle downstream newsegment by sending our own newsegment before the
next buffer to be released. (#323900)
2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
Original commit message from CVS:
2005-12-17 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
add queue delay to new segment as well (as opposed to just the first
buffer). (bug #322347)
2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
ext/libvisual/visual.c: change some char* into char[]
Original commit message from CVS:
* ext/libvisual/visual.c: (make_valid_name):
change some char* into char[]
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
prepare to handle EOS and SEGMENT_DONE
2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
tests/check/generic/states.c: Blacklist cdparanoia element in state test.
Original commit message from CVS:
* tests/check/generic/states.c: (GST_START_TEST):
Blacklist cdparanoia element in state test.
2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
Original commit message from CVS:
* gst/tcp/gsttcp.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
Add <string.h> includes for memset and FD_ZERO (fixes #323878;
patch by: Benjamin Pineau).
2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
(gst_video_rate_chain):
Fix timestamping for videorate when the first buffer it sees has a
non-zero timestamp. Fix some misleading debug output.
2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Don't leak all input buffers to audioresample.
2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
Don't operate on empty text buffers. Strip newlines and
tabs only from the end of the text, but leave them intact
in the middle. Fix typo in gtk-doc description.
2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c: (handoff):
Make sure the video frame buffer we return to apps via the
"frame" property always has caps set on it. Modify
_gst_gvalue_set_object() macro to handle NULL objects
gracefully too.
2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Adjust to some recent api changes and add wtays new cool seeking
capabillities
2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsadeviceprobe.h:
Helper functions to add device probing via the GstPropertyProbe
interface to a class.
* ext/alsa/gstalsamixer.h:
Comment out GST_ALSA_MIXER, it returns a struct that's not
used.
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
Add some debug info.
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_interface_supported),
(gst_implements_interface_init),
(gst_alsa_mixer_element_init_interfaces),
(gst_alsa_mixer_element_class_init),
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_get_property),
(gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixerelement.h:
Add 'device' and 'device-name' properties. Add GstPropertyProbe
for device handling (gnome-volume-control will need that).
2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
* ChangeLog:
* ext/Makefile.am:
* gst-plugins-base.spec.in:
updates to activate cdparanoia plugin
Original commit message from CVS:
updates to activate cdparanoia plugin
2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
Use the correct function to free list of typefind factories.
2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_init),
(gst_video_test_src_parse_caps), (gst_video_test_src_query),
(gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
(gst_video_test_src_create):
* gst/videotestsrc/gstvideotestsrc.h:
Implement seeking in videotestsrc.
Small cleanups.
2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
Original commit message from CVS:
* ext/cdparanoia/Makefile.am:
* ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
(gst_paranoia_endian_get_type), (_do_init),
(cdparanoia_class_init), (cdparanoia_init),
(cdparanoia_set_property), (cdparanoia_get_property),
(cdparanoia_do_seek), (cdparanoia_is_seekable),
(cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
(cdparanoia_convert), (cdparanoia_get_query_types),
(cdparanoia_query), (cdparanoia_set_index),
(cdparanoia_uri_set_uri):
* ext/cdparanoia/gstcdparanoia.h:
Partially ported cdparanoia now that basesrc can support a
plugin like this..
2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (main):
Set higher priority for bus events so they don't get reordered with
gtk gui events.
* tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
(flush_toggle_cb), (main):
Added checkbox do disable flushing seeks.
Disable scrubbing when doing non flushing seeks.
2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_sub_parse_init),
(gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
(parser_state_init), (handle_buffer), (gst_sub_parse_chain),
(gst_sub_parse_sink_event), (gst_sub_parse_change_state):
Implement some sort of event handling that doesn't rely on
g_return_if_fail; make sure we always push the last chunk of an
.srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
state change function; remove some old cruft. Seeking is still
rather unlikely to work though.
* tools/.cvsignore:
Ignore more.
2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
Original commit message from CVS:
2005-12-11 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
Fixed a leak of the current image reference when cleaning up.
Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
Original commit message from CVS:
* tools/Makefile.am:
* tools/gst-launch-ext-m.m:
Remove gst-launch-ext. It doesn't work, and is no longer
particularly useful.
2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
don't pass random values to ogmparse convert function.
Make seeking possible in the exile1.ogm file.
2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Work around refcount problem with g_value_set_object() that occur
if the core has been compiled against GLib-2.6 (g_value_set_object()
will only g_object_ref() the element, but the caller will
gst_object_unref() it and bad things will happen due to the way
GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
totem for people on FC4 using Thomas's 0.10 RPMs.
2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
Time to welcome ogm to 0.10 :)
Original commit message from CVS:
Time to welcome ogm to 0.10 :)
* ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
(gst_ogg_pad_typefind):
Oggdemux can now properly typefind elements with dynamic pads.
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
Properly set caps on src pad, and set caps on outgoing buffers.
2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasink.h:
* ext/alsa/gstalsasrc.c:
* ext/alsa/gstalsasrc.h:
* ext/cdparanoia/gstcdparanoia.h:
* ext/gnomevfs/gstgnomevfsuri.h:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gsttextoverlay.h:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisenc.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
* gst-libs/gst/interfaces/navigation.h:
* gst-libs/gst/interfaces/propertyprobe.h:
* gst-libs/gst/interfaces/tuner.h:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/interfaces/xoverlay.h:
* gst-libs/gst/netbuffer/gstnetbuffer.h:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/video/video.h:
* gst/adder/gstadder.c:
* gst/adder/gstadder.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioconvert/gstchannelmix.c:
* gst/audioconvert/gstchannelmix.h:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/buffer.h:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.h:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c:
* gst/playback/gststreaminfo.h:
* gst/tcp/gstfdset.c:
* gst/tcp/gstfdset.h:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpplugin.h:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c:
* gst/videorate/gstvideorate.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.h:
* sys/v4l/gstv4lcolorbalance.h:
* sys/v4l/gstv4ltuner.h:
* sys/v4l/gstv4lxoverlay.h:
* sys/v4l/v4l_calls.h:
* sys/v4l/videodev_mjpeg.h:
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
* tests/check/elements/audiotestsrc.c:
* tests/check/elements/videotestsrc.c:
* tests/check/elements/volume.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
expand tabs
Original commit message from CVS:
expand tabs
2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstgconf.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* docs/libs/tmpl/gsttuner.sgml:
* docs/libs/tmpl/gstxoverlay.sgml:
put back stability level
Original commit message from CVS:
put back stability level
2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
back to HEAD
Original commit message from CVS:
back to HEAD
=== release 0.10.0 ===
2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
releasing 0.10.0
Original commit message from CVS:
releasing 0.10.0
