=== release 0.10.36 === 2012-02-20 Tim-Philipp Müller * configure.ac: releasing 0.10.36, "Better" 2012-02-20 23:19:49 +0000 Tim-Philipp Müller * po/ca.po: * po/id.po: po: update translations 2012-02-17 15:08:36 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/vorbis/gstvorbisparse.c: vorbisparse: pedantically recognize undefined headers too 2012-02-01 16:32:24 +0000 Vincent Penquerc'h * 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 * gst-libs/gst/interfaces/propertyprobe.c: propertyprobe: fix documentation 2012-01-18 14:58:08 +0000 Vincent Penquerc'h * 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 * 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 * gst/playback/gststreamsynchronizer.c: streamsynchronizer: avoid unlikely NULL dereference 2012-01-19 16:35:54 +0100 Mark Nauwelaerts * 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 * tools/gst-discoverer.c: gst-discoverer: remove extraneous variable 2012-01-19 16:32:37 +0100 Mark Nauwelaerts * gst/playback/gstplaysink.c: playsink: verify linking to overlay element 2012-01-19 16:32:05 +0100 Mark Nauwelaerts * gst/playback/gstplaysink.c: playsink: avoid finding sink in NULL bin in corner case 2012-01-19 16:29:53 +0100 Mark Nauwelaerts * gst-libs/gst/tag/gstexiftag.c: tag: exif: add missing break 2012-01-17 18:19:30 +0100 Mark Nauwelaerts * ext/ogg/gstoggstream.c: oggstream: initialize variable ... to help out challenged compiler. 2012-01-16 11:43:25 +0000 Vincent Penquerc'h * 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 * 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 * 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 * 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 * ext/vorbis/gstvorbisdec.c: Fix TREMELO -> TREMOLO typo 2012-01-12 16:24:01 +0000 Vincent Penquerc'h * ext/theora/gsttheoraparse.c: theoraparse: fix array leak 2012-01-12 14:26:05 +0000 Vincent Penquerc'h * 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 * 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 * 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 * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: fix caps and discoverer object ref leaks 2012-01-11 11:55:59 +0000 Vincent Penquerc'h * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: add a few consts where appropriate 2012-01-11 11:55:36 +0000 Vincent Penquerc'h * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: fix pad leak 2012-01-10 18:27:19 +0000 Tim-Philipp Müller * 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 * 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 * 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 * ext/ogg/gstoggstream.c: oggstream: fix tag list leak 2012-01-10 16:51:09 +0000 Vincent Penquerc'h * ext/ogg/gstoggdemux.c: oggdemux: fix pad leak 2012-01-10 16:14:29 +0000 Vincent Penquerc'h * 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 * gst/playback/gstplay-enum.h: playback: document DEINTERLACE flag 2011-12-16 15:27:24 +0000 Vincent Penquerc'h * 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 * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: make is_subtitle_caps thread-safe 2011-11-01 17:57:59 +0100 Havard Graff * 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 * 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 * 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 * configure.ac: build: Run platform check for platform specific configuration. 2011-10-12 11:28:10 +0200 Pascal Buhler * 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 * 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 * 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 * common: Automatic update of common submodule From 63d592e to cb5da59 2012-01-03 11:04:23 +0100 Mark Nauwelaerts * 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 * gst/playback/gststreamsynchronizer.c: streamsynchronizer: avoid crashing when operating on released pad 2011-12-27 14:37:26 -0300 Thiago Santos * 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 * ext/theora/gsttheoraenc.c: theoraenc: fix template caps creation on big endian systems 2011-12-23 22:24:44 +0000 Tim-Philipp Müller * 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 * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: add a few more debug statements 2011-12-22 16:53:49 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstaudiodecoder.c: * gst-libs/gst/audio/gstaudiodecoder.h: audiodecoder: tweak documentation 2011-12-22 07:53:39 -0300 Thiago Santos * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: fix late buffer leak 2011-12-12 11:54:56 +0100 Sebastian Dröge * gst-libs/gst/glib-compat-private.h: glib-compat: Add license boilerplate for LGPL 2011-12-10 02:08:49 +0000 Tim-Philipp Müller * 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 * 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 * gst-plugins-base.spec.in: Add latest header file to spec file 2011-12-09 01:31:20 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * tests/examples/Makefile.am: examples: dist fft sub-directory 2011-11-28 10:05:50 -0300 Thibault Saunier * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/tag/xmpwriter.c: xmpwriter: update for thread API deprecations in glib master 2011-12-04 13:43:06 +0100 Stefan Sauer * tests/examples/fft/Makefile.am: fft-example: re-add Makefile.am 2011-12-02 23:35:50 +0100 Stefan Sauer * configure.ac: configure: trim trailing whitespace 2011-12-02 23:34:47 +0100 Stefan Sauer * 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 * 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 * 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 * tests/check/elements/audioresample.c: tests: fix caps leak in audioresample tests 2011-12-01 01:07:26 +0000 Tim-Philipp Müller * tests/check/pipelines/basetime.c: tests: fix memory leak in basetime test 2011-11-30 23:58:19 +0000 Tim-Philipp Müller * 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 * 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 * ext/alsa/gstalsasrc.c: Revert "alsasrc: Fix some compilation errors" This reverts commit 2b84f5bd74ddb50f7832917ea8b4dd38d005631b. 2011-11-30 23:15:12 +0000 Tim-Philipp Müller * 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 * 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 * 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 * ext/alsa/gstalsasrc.c: Revert "alsasrc: style fix" This reverts commit f70ca6d4cbfd2b672dcc7215814bf6b39ce2c3f8. 2011-11-30 14:25:11 +0100 Sebastian Dröge * 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 * 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 * ext/ogg/gstoggdemux.c: oggdemux: fix compiler warning 2011-11-29 08:49:53 +0100 Alessandro Decina * 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 * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Fix compiler warning 2011-11-28 19:03:50 +0100 Alexey Fisher * 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 2011-11-28 19:06:57 +0100 Sebastian Dröge * gst/playback/gstplaysinkconvertbin.c: playsinkconvertbin: Fix stupid mistake in last commit 2011-11-28 19:03:54 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gstplaysinkconvertbin.c: playsinkconvertbin: make identiy silent 2011-11-25 13:11:54 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggdemux.c: oggdemux: minor cleanup 2011-09-27 16:45:26 +0100 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggstream.c: oggstream: extract opus comments if available 2011-11-22 13:15:33 +0000 Vincent Penquerc'h * 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 * ext/ogg/gstoggdemux.c: oggdemux: add some more debug info when determining start time 2011-11-22 12:55:56 +0000 Vincent Penquerc'h * ext/ogg/gstoggstream.c: oggstream: fix opus duration calculation 2011-11-22 12:00:58 +0000 Vincent Penquerc'h * ext/ogg/gstoggstream.c: oggstream: early out on headers when determining packet duration 2011-11-21 17:03:21 +0000 Vincent Penquerc'h * 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 * 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 * ext/ogg/gstoggmux.c: * ext/ogg/gstoggstream.c: ogg: add opus support 2011-11-16 19:00:44 +0100 Mark Nauwelaerts * ext/vorbis/gstvorbisenc.c: vorbisenc: reset tag setter interface when appropriate 2011-11-16 19:00:30 +0100 Mark Nauwelaerts * 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 * 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 * gst-libs/gst/audio/gstaudiodecoder.c: audio: Remove some unused variables 2011-08-30 18:27:09 -0400 Olivier Crête * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: make unsigned properties unsigned, not signed 2011-11-09 00:36:51 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 2011-11-07 11:31:47 +0100 Felipe Contreras * 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 * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: rename some variables 2011-05-21 16:16:42 +0300 Felipe Contreras * 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 * 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 * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: trivial comment fixes Some found by Havard Graff. Signed-off-by: Felipe Contreras 2011-11-04 10:37:12 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * gst/playback/gstdecodebin2.c: decodebin2: Use existing "caps" quark for one of the structure sets 2011-11-03 10:07:27 +0100 Sebastian Dröge * gst/playback/gstplaysinkconvertbin.c: playsinkconvertbin: Don't add identity multiple times 2011-10-19 14:13:39 +0100 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gstplaysinkvideoconvert.c: playsink-videoconvert: Fix warning on build Remove unused variable 2011-10-17 21:05:30 +0000 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggmux.c: oggmux: Remove obsolete #include 2011-11-02 23:33:18 +0000 Tim-Philipp Müller * 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 * common: * configure.ac: configure: Allow setting GLIB_EXTRA_CFLAGS Similar to gstreamer commit bb2020b1e794210cf7d44c6626122f611016a620 2011-10-30 20:00:47 +0000 Tim-Philipp Müller * gst/playback/gstsubtitleoverlay.c: subtitleoverlay: don't use soon-to-be-deprecated gst_filter_run() 2011-10-28 13:58:47 +0200 Mersad Jelacic * 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 * 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 * ext/ogg/gstoggmux.c: * ext/ogg/gstoggmux.h: oggmux: port to gstcollectpads2 2011-10-27 23:39:31 +1100 Jan Schmidt * tests/examples/Makefile.am: build: Fix build for moved volume subdir 2011-10-27 09:51:46 +0200 Stefan Sauer * 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 * tests/examples/audio/Makefile.am: audio examples. fix the makefile 2011-10-27 09:33:55 +0200 Stefan Sauer * tests/examples/volume/volume.c: volume: make global vars static 2011-10-27 09:33:01 +0200 Stefan Sauer * 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 * gst/playback/gstplaysinkaudioconvert.c: playsinkaudioconvert: Fix warning when there is no target pad yet 2011-10-13 11:34:49 -0400 Nicolas Dufresne * 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 * gst/playback/gstplaybasebin.c: playbasebin: remove avoidable call to gst_object_set_name 2011-10-21 21:41:03 +0200 René Stadler * ext/ogg/gstoggdemux.c: oggdemux: remove avoidable call to gst_object_set_name 2011-10-21 21:39:01 +0200 René Stadler * gst/audioconvert/Makefile.am: * gst/audioconvert/channelmixtest.c: audioconvert: bury dead test program 2011-10-20 10:13:46 -0300 Reynaldo H. Verdejo Pinochet * 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 * gst/playback/gstsubtitleoverlay.c: subtitleoverlay: fix leaks of pad templates and internal proxy pads 2011-10-19 19:37:07 +0200 René Stadler * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gstsubtitleoverlay.c: subtitleoverlay: fix event unref in (rare) error case 2011-10-07 17:41:32 +0100 Vincent Penquerc'h * 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 * 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 * gst/videotestsrc/Makefile.am: videotestsrc: fix LDADD missing GST_LIBS 2011-10-09 21:19:32 +0200 Mark Nauwelaerts * 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 * gst-libs/gst/audio/gstaudiodecoder.c: audioencoder: fix compile warning 2011-10-08 20:17:43 +0200 Mark Nauwelaerts * 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 * tests/check/pipelines/vorbisdec.c: tests: vorbisdec: properly configure audiodecoder when requiring perfect ts 2011-10-08 20:14:27 +0200 Mark Nauwelaerts * 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 * 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 * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: fix timestamp tolerance handling 2011-10-08 20:09:09 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: handle empty input by discarding 2011-10-07 14:52:33 +0200 Mark Nauwelaerts * 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 * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: make upstream queries MT-safe 2011-10-07 14:32:33 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: make upstream queries and events MT-safe 2011-10-05 15:43:35 +0200 Mark Nauwelaerts * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gstplaysink.c: playsink: Add audio- and text-sink props 2011-10-04 23:09:42 +0200 Stefan Sauer * gst/audiotestsrc/gstaudiotestsrc.c: auditestsrc: indent fix 2011-10-04 16:22:55 +0200 Robert Swain * 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 * gst/playback/gstplaybin2.c: playbin2: Minor cleanup of decoder-sink compatibility checking code 2011-09-30 12:29:34 -0300 Thibault Saunier * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: really push pending events 2011-09-28 14:32:20 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: really push pending events 2011-09-27 16:16:54 +0200 Mark Nauwelaerts * 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 * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: protect pending_events with proper lock 2011-09-27 15:31:20 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: clean up some documentation 2011-09-27 00:32:41 +0100 Tim-Philipp Müller * docs/libs/gst-plugins-base-libs-sections.txt: docs: minor docs fix 2011-09-26 16:36:56 +0200 Sebastian Dröge * docs/libs/gst-plugins-base-libs-sections.txt: docs: Adjust for GstAudioEncoder API changes 2011-09-26 16:36:22 +0200 Sebastian Dröge * win32/common/libgstaudio.def: win32: Adjust for GstAudioEncoder API changes 2011-09-26 16:35:55 +0200 Sebastian Dröge * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: Improve set_frame_sample_{min,max} documentation 2011-09-26 16:22:00 +0200 Sebastian Dröge * 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 * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: Delay sending of serialized events to finish_frame() 2011-09-26 16:02:51 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: provide CODEC/AUDIO_CODEC handling 2011-09-26 13:42:38 +0200 Mark Nauwelaerts * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefindfunctions: fix indentation 2011-09-23 17:50:31 +0200 Robert Swain * gst/encoding/gstencodebin.c: encodebin: Avoid unnecessary read only caps copy 2011-09-22 15:38:51 +0200 Mark Nauwelaerts * 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 * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: changed is verily the opposite of equal 2011-09-22 15:37:07 +0200 Mark Nauwelaerts * 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 * gst-libs/gst/audio/audio.h: audio: some more accessor macros for GstAudioInfo 2011-09-22 15:34:41 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: fix documentation typo 2011-09-19 18:32:26 +0100 Sjoerd Simons * tests/check/elements/videorate.c: videorate: Add tests for the max-rate case 2011-09-19 18:31:07 +0100 Sjoerd Simons * tests/check/elements/videorate.c: videorate: Print which caps didn't match up 2011-09-19 18:26:04 +0100 Sjoerd Simons * 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 * tests/check/elements/videorate.c: videorate: Add test for caps negotiation 2011-09-01 16:47:49 +0100 Sjoerd Simons * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggdemux.c: oggdemux: minor printf format fix 2011-09-14 12:23:19 +0100 Vincent Penquerc'h * 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 * 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 * gst/playback/gstplaybin2.c: playbin2: fix compiler warning Remove a check for gchar >= 128 2011-09-15 16:47:26 +0200 Stefan Sauer * gst/adder/gstadder.c: adder: don't access the event after pushing Fixes valgrind warnings. 2011-09-15 14:27:35 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/theora/gsttheoraenc.c: theoraenc: Fix descriptions of properties 2011-09-10 18:30:55 +0100 Tim-Philipp Müller * 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 * docs/design/design-decodebin.txt: docs: fix some typos in the decodebin design document 2011-09-09 13:07:57 +0100 Tim-Philipp Müller * 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 * 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 * 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 * docs/design/design-decodebin.txt: decodebin: Update design documentation about how Parser/Converter are handled 2011-09-08 13:25:27 +0200 Sebastian Dröge * 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 * 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 * gst/playback/gstdecodebin2.c: decodebin2: Make sure to fixate Parser/Converter caps before continuing autoplugging 2011-09-07 16:04:43 +0200 Josep Torra * 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 * 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 * 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 * gst/audiotestsrc/gstaudiotestsrc.h: docs: add two mising enum docs 2011-09-07 14:10:46 +0200 Sebastian Dröge * tests/check/elements/audiorate.c: audiorate: Use complete audio caps, including the endianness field 2011-09-07 12:32:01 +0100 Tim-Philipp Müller * 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 * 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 * common: Automatic update of common submodule From a39eb83 to 11f0cd5 2011-09-06 19:18:27 +0100 Christian Fredrik Kalager Schaller * gst-plugins-base.spec.in: Add latest files to spec file 2011-09-06 20:13:30 +0200 Stefan Sauer * docs/libs/Makefile.am: docs: activate overrides file to fix make distcheck 2011-09-06 16:46:02 +0200 Wim Taymans * 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 * 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 * 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 * common: Automatic update of common submodule From 605cd9a to a39eb83 2011-09-06 14:16:10 +0200 Sebastian Dröge * 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 * 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 * 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 * gst/playback/gstdecodebin2.c: decodebin2: Fix memory leak 2011-09-06 12:14:33 +0200 Sebastian Dröge * tests/check/elements/decodebin2.c: decodebin2: Add unit test for correct parser/converter negotiation 2011-06-26 15:40:17 +0200 Sebastian Dröge * 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 * 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 * tests/check/elements/decodebin2.c: decodebin2: Add Tim as author for the parser test 2011-09-06 10:07:33 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/videorate/gstvideorate.c: Correct added versions 2011-08-31 14:45:08 +0200 Sebastian Dröge * gst/playback/gstplaysink.c: playsink: Only unref ts_offset elements if they're not NULL 2011-08-31 12:39:18 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: remove leftover experimental code 2011-08-17 18:32:54 +0200 Mark Nauwelaerts * 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 * 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 * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: rename property 2011-08-16 20:39:07 +0200 Mark Nauwelaerts * 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 * 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 * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: rename some properties 2011-08-16 18:23:14 +0200 Mark Nauwelaerts * 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 * 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 * 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 * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: debug format fixes 2011-04-28 12:01:30 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: debug format fix 2011-03-31 14:03:11 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: fixup documentation 2011-03-29 15:51:40 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: fix FLUSH_STOP actions 2011-03-28 13:16:27 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: preserve upstream seek event seqnum 2011-03-22 11:09:56 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: use buffer running time for granule calculation 2011-03-22 10:45:47 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: minor fix in ts resync 2011-03-21 11:40:31 +0100 Mark Nauwelaerts * 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 * 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 * gst-libs/gst/audio/gstbaseaudiodecoder.c: baseaudiodecoder: cater for audio-codec tag 2011-03-10 16:01:05 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudiodecoder.c: * gst-libs/gst/audio/gstbaseaudiodecoder.h: baseaudiodecoder: initial version 2011-03-16 18:41:03 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: misc fixes 2011-03-15 17:27:42 +0100 Mark Nauwelaerts * 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 * gst-libs/gst/audio/gstbaseaudioencoder.h: baseaudioencoder: mark unstable API 2011-03-10 15:12:54 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: fix clearing context 2011-03-10 15:12:19 +0100 Mark Nauwelaerts * 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 * 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 * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: additional documentation on granule semantics and configuration 2011-03-09 12:24:34 +0100 Mark Nauwelaerts * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: elaborate property names 2011-03-09 12:22:04 +0100 Mark Nauwelaerts * 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 * gst-libs/gst/audio/gstbaseaudioencoder.c: baseaudioencoder: gtk-doc syntax fixes 2011-03-09 12:17:18 +0100 Mark Nauwelaerts * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * win32/common/libgstpbutils.def: win32: Add new discoverer API 2011-08-26 10:03:17 +0200 Sebastian Dröge * docs/libs/gst-plugins-base-libs-sections.txt: docs: Add new discoverer API 2011-08-24 16:29:08 +0100 Vincent Penquerc'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-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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/video/gstvideosink.h: docs: fix xref for the property 2011-08-20 19:16:42 +0200 Stefan Kost * 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 * 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 * 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 * 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 * gst-libs/gst/tag/Makefile.am: tag: fix distcheck issue Dist licenses dict. 2011-08-18 16:20:57 +0100 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: configure: try pkg-config first when looking for zlib 2011-08-14 20:44:19 +0100 Tim-Philipp Müller * 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 * gst-libs/gst/tag/id3v2.c: tag: id3v2: avoid some relocations, make table static 2011-08-14 01:47:41 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/tag/id3v2.c: tag: id3v2: cast pointer math results to glong 2009-10-09 13:38:17 +0300 Stefan Kost * 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 * gst-libs/gst/tag/id3v2.c: tag: id3v2: fix printf warnings on macosx 2009-10-07 14:03:20 +0300 Stefan Kost * gst-libs/gst/tag/id3v2frames.c: tag: id3v2: fprintf, sprintf, sscanf need stdio.h 2009-09-22 15:03:20 +0200 Alessandro Decina * gst-libs/gst/tag/id3v2frames.c: tag: id3v2: Fix compile warnings with gcc 4.0.1. 2009-08-09 12:52:17 +0200 LoneStar * 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 * 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 * gst-libs/gst/tag/id3v2.c: tag: id3v2: fix typo in debug message 2009-08-07 16:02:23 +0100 Tim-Philipp Müller * 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 * 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 * 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 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 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 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 * 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 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 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 tag: id3v2: Extract license/copyright URIs from ID3v2 WCOP frames (Fixes #447000). Original commit message from CVS: Based on patch by: Jason Kivlighn * 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 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 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 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 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 2006-11-01 13:59:49 +0000 Tim-Philipp Müller 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 * 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 * 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 * 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 * 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 * 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 * 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 * win32/common/libgsttag.def: win32: update libgsttag.def for new API 2011-08-10 15:21:41 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * .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 * 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 * 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 * 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 * 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 * gst/volume/gstvolumeorc-dist.c: volume: Update disted ORC files 2011-08-03 14:14:55 -0300 Thiago Santos * 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 * 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 * ext/ogg/gstoggstream.c: oggstream: Fix crashes with 0-byte vorbis packets Fixes bug #655574. 2011-07-28 14:43:53 +0200 Jens Georg * 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 * ext/theora/gsttheoradec.c: theoradec: segfault on 0-byte ogg_packet in _chain_reverse 2011-07-29 10:23:02 +0100 Tim-Philipp Müller * 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 * 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 * 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 * gst-libs/gst/tag/gsttagmux.h: tagmux: const-ify GstTagList argument of render vfuncs 2011-07-15 20:39:20 +0100 Tim-Philipp Müller * 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 * 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 * 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 Fix build on Mac OS X 10.5 Original commit message from CVS: 2007-11-20 Julien MOUTTE * 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 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 Add apev2mux element (#343122). Original commit message from CVS: Patch by: Sebastian Dröge * 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 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 * 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 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 * 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 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 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 * 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 * 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 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 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 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 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 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 ext/taglib/gsttaglib.cc: and add support for TCOP (copyright) Original commit message from CVS: 2006-03-11 Christophe Fergeau 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 new id3v2 muxer based on TagLib Original commit message from CVS: 2006-03-09 Christophe Fergeau reviewed by: Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/adder/gstadder.c: adder: more debug logging for events 2011-07-26 12:33:56 +0200 Edward Hervey * 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 * 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 * 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 * 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 * ext/pango/gsttextoverlay.c: textoverlay: add example for feeding from stdin 2011-07-23 13:46:31 +0200 Stefan Sauer * tests/check/pipelines/basetime.c: test: print actual timestamp on failure 2011-07-20 13:46:31 +0200 Stefan Sauer * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggmux.c: oggmux: check for EOS on current pad, not best Fixes #654270. 2011-07-09 11:59:42 +0200 Piotr Fusik * gst/typefind/gsttypefindfunctions.c: typefind: fixed detection of audio/x-sap Fixes: #654295. Signed-off-by: David Schleef 2011-06-30 20:33:36 +0200 Luis de Bethencourt * gst/encoding/gstencodebin.c: encodebin: fix compiler warning cspace and cspace2 may run uninitialized. 2011-06-29 13:12:49 +0200 Robert Swain * 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 * 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 * 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 * 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 * docs/design/design-decodebin.txt: docs: minor addition to decodebin2 design doc 2011-06-26 01:06:19 +0100 Tim-Philipp Müller * tests/check/libs/navigation.c: tests: the navigation interface isn't GstImplementsInterface-wrapped 2011-06-26 00:49:46 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 69b981f to 605cd9a 2011-06-18 13:32:17 +0100 Tim-Philipp Müller 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 * 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 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 * gst-plugins-base.spec.in: Add gobject introspection files to spec 2011-06-15 14:53:56 +0100 Christian Fredrik Kalager Schaller * gst-plugins-base.spec.in: remove old v4l plugin from spec file 2011-06-15 14:49:41 +0100 Christian Fredrik Kalager Schaller * tests/examples/Makefile.am: Add missing dist subdir 2011-06-15 14:21:30 +0100 Tim-Philipp Müller * gst-libs/gst/audio/Makefile.am: audio: link test program against libgstaudio 2011-06-14 10:31:18 +0530 Debarshi Ray * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: fix c99-ism 2011-05-23 16:02:34 +0200 Sebastian Dröge * 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 * 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 * gst-libs/gst/tag/lang.c: lang: fix possible array overrun We where checking for i * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * docs/design/design-decodebin.txt: docs: add some text about parser/decoder autoplugging issues 2011-06-06 14:41:41 +0200 Mark Nauwelaerts * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: fix c99-ism 2011-06-03 16:29:00 +0200 Luis de Bethencourt * 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 * ext/theora/gsttheoraenc.c: theora: use fixed src cap pads 2011-06-02 18:57:05 +0200 Luis de Bethencourt * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * win32/common/libgstvideo.def: win32: update .def file for new API 2011-05-28 12:39:06 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/pango/gsttextrender.c: textrender: Add bound checks to not write outside the image area 2011-05-26 10:42:46 +0200 Jindrich Makovicka * 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 * ext/gnomevfs/gstgnomevfssrc.c: gnomevfssrc: Keep track of interruptions during read with a flag 2010-09-03 09:11:30 -0400 American Dynamics * 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 * sys/ximage/ximagesink.c: ximagesink: Remove g_assert from interface query 2011-05-25 14:08:43 +0300 Sreerenj Balachandran * sys/xvimage/xvimagesink.c: xvimagesink: Remove the g_assert from interface query 2011-05-26 00:17:40 +0300 Stefan Kost * 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 * 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 * 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 * tests/examples/seek/seek.c: seek: add --audiosink and --videosink command line options 2011-05-25 18:50:34 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * sys/xvimage/xvimagesink.c: xvimagesink: run gst-indent 2011-05-25 12:29:21 +0300 Stefan Kost * 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 * 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 * gst-libs/gst/video/video.c: docs: add missing parameter docs 2011-05-23 23:53:38 +0300 Stefan Kost * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/subparse/gstsubparse.c: subparse: Compile the typefind regex with optimization to speed up matching 2011-05-23 15:51:14 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/tag/gstxmptag.c: xmptag: have the default branch as the last one 2011-05-23 14:00:04 +0300 Stefan Kost * gst-libs/gst/tag/gstxmptag.c: xmptag: an uint value can't be <0 2011-05-23 13:53:06 +0300 Stefan Kost * gst-libs/gst/video/video.c: whitespace: trim trailing whitespace 2011-05-23 13:50:59 +0300 Stefan Kost * 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 * 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 * gst-libs/gst/tag/lang.c: lang: fix possible array overrun We where checking for i * 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 * 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 * common: Automatic update of common submodule From 9e5bbd5 to 69b981f 2011-05-19 13:40:29 +0100 Christian Fredrik Kalager Schaller * gst-plugins-base.spec.in: Add new header file 2011-05-19 08:30:14 -0300 Thiago Santos * 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 * gst/encoding/gstencodebin.c: encodebin: fix typos 2011-05-18 22:07:58 +0200 Aleix Conchillo Flaque * ext/vorbis/gstvorbisdec.c: vorbisdec: Handle headers in caps 2011-05-18 16:09:47 +0300 Stefan Kost * common: Automatic update of common submodule From fd35073 to 9e5bbd5 2011-05-18 13:18:15 +0200 Robert Swain * 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 * common: Automatic update of common submodule From 46dfcea to fd35073 2011-05-18 09:34:52 +0200 Robert Swain * 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 * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Fix typo which broke the build 2011-05-16 15:35:50 +0200 Miguel Angel Cabrera Moya * 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 * 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 * 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 * 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 * 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 * 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 * tests/examples/encoding/Makefile.am: encodebin: examples: Add missing base libs to makefile 2011-04-28 10:58:15 -0300 Thiago Santos * 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 * 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 * tests/check/libs/tag.c: tests: xmp: New tests for the Iptc4xmpExt tags 2011-04-18 23:28:13 -0300 Thiago Santos * 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 * 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 * 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 * 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 * 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 * win32/common/libgstaudio.def: win32: Update libgstaudio.def for new symbols 2011-05-14 17:27:30 +0530 Arun Raghavan * 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 * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: Fix trivial indentation problems 2011-03-07 20:49:16 +0530 Arun Raghavan * 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 * 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 * 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 * gst-libs/gst/audio/gstringbuffer.c: ringbuffer: Add support for MPEG audio buffers 2011-03-14 15:49:57 +0530 Arun Raghavan * 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 * gst-libs/gst/audio/gstringbuffer.c: ringbuffer: Add support for DTS buffers 2011-05-14 11:42:50 +0200 Sebastian Dröge * configure.ac: configure: Require core 0.10.34.1 for the new ghostpad API 2011-05-09 22:20:23 +0200 Andoni Morales Alastruey * 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 * gst/playback/gstplaysink.c: playsink: Use new ghostpad/proxypad API to get the internal pad 2011-05-09 12:59:22 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * ext/alsa/gstalsasrc.c: alsasrc: Fix some compilation errors 2011-05-09 11:50:05 +0200 Pontus Oldberg * ext/alsa/gstalsasrc.c: * ext/alsa/gstalsasrc.h: alsasrc: Improve timestamp accuracy Fixes bug #635256. 2011-05-06 17:01:53 +0200 Sebastian Dröge * gst/playback/gstsubtitleoverlay.c: * gst/playback/gstsubtitleoverlay.h: subtitleoverlay: Use new, public ghostpad functions 2011-05-03 11:26:32 +0300 Sreerenj Balachandran * sys/xvimage/xvimagesink.c: xvimagesink: Use GST_BOILERPLATE 2011-05-14 09:41:58 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From c3cafe1 to 46dfcea 2011-04-23 12:44:50 -0700 David Schleef * 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 * 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 * 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 * 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 * 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 * 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 * 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 * tests/check/pipelines/oggmux.c: oggmux: Remove bus GSource to prevent a valgrind warning 2011-04-18 09:16:35 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gstplaybin2.c: playbin2: avoid foregoing READY_TO_NULL when appropriate 2011-04-14 22:13:21 +0200 Mark Nauwelaerts * 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 * 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 * tests/check/elements/encodebin.c: encodebin: Unref encoding profiles after usage in the test 2011-04-14 12:55:00 +0200 Sebastian Dröge * 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 * 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 * 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 * tests/check/libs/tag.c: tests: tag: Fix typo 2011-04-13 23:17:14 -0300 Thiago Santos * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/video/video.h: docs: fix typo in video format docs 2011-04-12 12:41:06 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * gst/videorate/gstvideorate.c: videorate: empty caps have no structure to pick 2011-04-11 10:06:53 +0100 Tim-Philipp Müller * 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 * 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 * gst-plugins-base.spec.in: Add new header file to spec file 2011-04-08 15:10:02 +0200 Sebastian Dröge * 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 * 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 * 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 * ext/vorbis/gstvorbisdec.c: vorbisdec: Unref events if the parent element disappeared 2011-03-21 16:03:16 +0100 Havard Graff * ext/vorbis/gstvorbisdec.c: vorbisdec: make upstream queries and events MT-safe 2011-04-06 16:25:37 +0100 Bastien Nocera * 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 * 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 * 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 * ext/theora/gsttheoraenc.c: theoraenc: refactor multipass file writing 2011-02-08 14:02:20 +0100 Mark Nauwelaerts * 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 * 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 * 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 * 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 * 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 * 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 * tests/check/libs/.gitignore: tests: ignore xmpwriter unit test binary 2011-04-04 17:21:45 +0200 Haakon Sporsheim * 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 * 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 * 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 * common: Automatic update of common submodule From 1ccbe09 to c3cafe1 2011-03-11 10:41:11 +0100 Trond Andersen * 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 * 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 * 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 * 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 * gst/encoding/Makefile.am: * gst/playback/Makefile.am: Remove setting of plugindir from Makefiles 2011-03-23 23:10:51 -0700 David Schleef * 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 * ext/ogg/gstoggmux.c: oggmux: fix warning building in mac os x 2011-04-01 15:33:42 +0200 Josep Torra * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/pango/gsttextoverlay.c: textoverlay: Added support for ARGB and other RGB alpha variants 2011-01-11 10:34:33 -0700 Lane Brooks * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefind: Use the DataScanCtx for the m3u8 typefinder 2011-02-14 19:05:09 +0100 Andoni Morales Alastruey * gst/typefind/gsttypefindfunctions.c: typefind: add m3u8 playlists 2011-03-21 15:34:09 -0300 Thiago Santos * 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 * 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 * 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 * 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 * 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 * tests/check/libs/.gitignore: tests: ignore new libsabi test binary 2011-03-26 11:59:54 +0000 Tim-Philipp Müller * 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 * common: Automatic update of common submodule From 193b717 to 1ccbe09 2011-03-25 14:55:52 +0200 Stefan Kost * common: Automatic update of common submodule From b77e2bf to 193b717 2011-03-25 11:06:35 +0200 Stefan Kost * 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 * docs/design/part-interlaced-video.txt: docs: Add an interlaced video design document 2011-03-25 09:29:38 +0100 Sebastian Dröge * common: Automatic update of common submodule From d8814b6 to b77e2bf 2011-03-25 09:03:13 +0100 Sebastian Dröge * common: Automatic update of common submodule From 6aaa286 to d8814b6 2011-03-24 18:48:59 +0200 Stefan Kost * common: Automatic update of common submodule From 6aec6b9 to 6aaa286 2011-03-24 14:22:00 +0100 Sebastian Dröge * gst/playback/gstplaysink.c: playsink: Update comment about why an audio queue is needed 2011-03-24 14:21:01 +0100 Sebastian Dröge * 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 * 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 * 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 * win32/common/libgstvideo.def: video: adds missing function to win32 def 2011-03-23 12:02:35 -0300 Thiago Santos * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * autogen.sh: autogen: wingo signed comment 2011-03-21 19:22:30 +0100 Fraxinas * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/pango/gsttextoverlay.c: textoverlay: drop trailing whitespaces 2011-03-04 14:52:28 +0200 Stefan Kost * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: configure.ac: export plugin description more platform independent Fixes #642504. 2011-02-28 18:32:33 +0100 Mark Nauwelaerts * common: Automatic update of common submodule From 1de7f6a to 6aec6b9 2011-02-28 10:10:22 +0200 Stefan Kost * 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 * 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 * 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 * 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 * 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 * 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 * 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 * tests/check/elements/encodebin.c: encodebin: Fix double unref in unit test 2011-02-22 14:54:55 +0000 Tim-Philipp Müller * 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 * 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 * ext/ogg/gstoggmux.c: oggmux: Don't handle GstCollectData as GstObject, use the pad instead 2011-02-24 16:02:50 +0100 Sebastian Dröge * 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 * 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 * 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 * gst/playback/gstplaysink.c: playsink: release all chains when going to NULL Also fixes #642466. 2011-02-23 14:29:03 +0100 Mark Nauwelaerts * 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 * 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 * win32/common/libgstvideo.def: win32: update .def file for new libgstvideo API 2011-02-22 16:41:54 +0200 Stefan Kost * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: Chain dispose() up to parent class 2011-02-07 13:04:55 +0530 Arun Raghavan * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/videotestsrc/videotestsrc.c: videotestsrc: Add 16-bit-per-channel formats 2011-02-19 12:03:17 -0800 David Schleef * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gsturidecodebin.c: uridecodebin: Update autoplug-* signal docs from decodebin2 uridecodebin proxies these signals. 2011-02-18 11:58:44 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefind: Fix mpeg TS detection 2011-02-04 17:36:40 -0800 David Schleef * ext/theora/gsttheoraenc.c: theoraenc: move debug category init earlier 2011-02-03 22:41:23 -0800 David Schleef * ext/ogg/gstoggparse.c: * ext/ogg/gstoggstream.h: oggparse: better detection of delta unit flag 2011-01-15 18:21:28 -0800 David Schleef * ext/theora/gsttheoraenc.c: theoraenc: Set speed level while running 2011-01-13 15:12:53 -0800 Ralph Giles * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: don't leak parent tags 2011-02-16 11:56:16 +0200 Stefan Kost * 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 * 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 * 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 * 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 * 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 * gst/gdp/gstgdppay.c: gdppay: ensure buffer's metadata is writable before setting caps 2011-02-14 12:52:59 +0200 Stefan Kost * common: Automatic update of common submodule From f94d739 to 1de7f6a 2011-02-10 23:44:43 +0000 Tim-Philipp Müller * gst-plugins-base.doap: doap: update mailing list location 2011-02-08 23:58:56 +0530 Arun Raghavan * 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 * 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 * 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 * 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 * ext/ogg/gstoggmux.c: oggmux: free stream map caps when done 2011-02-02 17:23:43 +0000 Tim-Philipp Müller * 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 * 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 * gst/playback/gsturidecodebin.c: uridecodebin: fix copy-and-paste typo in property docs 2011-01-21 10:56:00 +0000 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gsturidecodebin.c: uridecodebin: also add https to buffer protocols HTTPS also needs buffering. 2011-01-30 15:40:53 +0200 Felipe Contreras * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: trivial cleanups It seems these stuff was neglected from commmit d8942e2. Signed-off-by: Felipe Contreras 2011-01-27 15:26:25 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * gst-libs/gst/pbutils/descriptions.c: pbutils: add description for degas images 2011-01-26 09:06:10 +0000 Tim-Philipp Müller * 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 * 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 * tests/examples/snapshot/snapshot.c: snapshot: add a newline to the usage and error output 2011-01-25 18:03:23 +0200 Stefan Kost * gst/playback/gstdecodebin2.c: decodebin2: add comment and whitespace trimming 2011-01-12 14:03:12 +0200 Stefan Kost * gst-libs/gst/pbutils/descriptions.c: pbutils: add description for x-annodex 2011-01-25 13:39:25 +0000 Vincent Penquerc'h * 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 * 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 * 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 * ext/ogg/gstoggdemux.c: oggdemux: Remove dead code 2011-01-11 15:10:42 +0800 Yang Xichuan * 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 * 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 * 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 * gst-libs/gst/riff/riff-media.c: riff: Add support for video/x-camstudio 2011-01-24 00:00:27 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/app/gstappbuffer.c: app: make GstAppBuffer get_type() function thread-safe 2011-01-18 01:09:53 +0530 Arun Raghavan * 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 * 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 * 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 * gst/gdp/gstgdppay.c: gdppay: make newsegment buffer metadata writable 2011-01-16 16:46:22 +0000 Tim-Philipp Müller * 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 * gst/typefind/gsttypefindfunctions.c: typefinding: set framed=false on DTS caps 2011-01-12 17:51:43 +0000 Tim-Philipp Müller * 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 * 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 * 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 * po/da.po: * po/gl.po: * po/pt_BR.po: po: update translations 2011-01-11 14:41:53 +0000 Bastien Nocera * 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 * 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 * 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 * common: Automatic update of common submodule From e572c87 to f94d739 2011-01-10 16:35:44 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From ccbaa85 to e572c87 2011-01-10 14:53:04 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 46445ad to ccbaa85 2011-01-10 15:55:26 +0800 Yang Xichuan * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * win32/common/libgstpbutils.def: win32: udpate pbutils .def file for API change 2011-01-06 23:13:53 +0000 Tim-Philipp Müller * 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 * po/fi.po: * po/ru.po: po: update translations 2011-01-06 23:08:34 +0000 Tim-Philipp Müller * 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 * 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 * tests/check/libs/.gitignore: tests: ignore new rtsp test binary 2011-01-05 15:54:15 -0800 David Schleef * 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 * 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 * gst-plugins-base.spec.in: Update spec file with discoverer and encodebinchanges 2011-01-05 15:53:09 +0530 Arun Raghavan * 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 * 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 * 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 * tools/gst-discoverer.c: tools: don't leak the GMainLoop 2011-01-06 00:28:39 +0000 Tim-Philipp Müller * 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 * 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 * docs/libs/gst-plugins-base-libs-sections.txt: docs: Add various new symbols 2011-01-05 01:50:34 +0530 Arun Raghavan * 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 * gst-libs/gst/pbutils/encoding-profile.c: encoding-profile: Give a better usage example 2011-01-03 18:52:00 +0100 Edward Hervey * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/encoding/gstencodebin.c: encodebin: Convert to new GstElementClass::request_new_pad_full vmethod 2011-01-05 15:31:09 +0100 Edward Hervey * gst-libs/gst/pbutils/pbutils.h: pbutils: Don't forget to include the encoding headers 2011-01-05 12:02:02 +0100 Edward Hervey * gst-libs/gst/video/video.c: video: Fix uninitialized variables reported by macosx gcc 2010-12-07 14:59:46 +0530 Arun Raghavan * gst-libs/gst/pbutils/codec-utils.c: codec-utils: Minor documentation changes 2011-01-02 15:48:47 -0800 David Schleef * gst/typefind/gsttypefindfunctions.c: typefind: Add stream-format to h264 caps 2011-01-02 17:21:54 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * tests/icles/output-selector-test.c: tests: Remove executable bits from non-executable files. 2009-02-24 16:33:51 +0100 Sebastian Dröge * 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 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 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 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 * 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 * 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 * gst/tcp/gstmultifdsink.c: multifdsink: only keep last valid timestamp Fixes #634397 2010-10-13 17:09:13 +0200 Andoni Morales Alastruey * 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 * ext/ogg/gstoggstream.c: ogg: fix typo in comment 2010-12-28 17:39:58 +0000 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/pbutils/descriptions.c: pbutils: add description for DVB subtitle caps 2010-12-23 17:18:17 +0000 Vincent Penquerc'h * 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 * ext/ogg/gstoggstream.c: oggstream: Fix parsing of theora size 2010-12-22 19:06:56 +0000 Vincent Penquerc'h * 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 * gst/encoding/gstencodebin.c: encodebin: Also use "Formatter"s for container formats 2010-12-22 18:19:48 +0100 Edward Hervey * gst-libs/gst/pbutils/encoding-target.c: encoding-target: Fix typo 2010-12-22 10:32:03 -0300 Thiago Santos * 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 * gst/encoding/gstencodebin.c: encodebin: minor fix in error handling. Don't call gst_bin_remove (bin, ). 2010-12-21 18:51:29 +0100 Edward Hervey * 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 * 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 * 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 * gst-libs/gst/rtp/gstbasertpdepayload.h: depay: update some docs 2010-12-21 15:02:18 +0100 Wim Taymans * 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 * 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 * ext/vorbis/gstvorbisdec.c: vorbisdec: avoid using invalid timestamps 2010-12-21 10:41:27 +0100 Wim Taymans * tests/examples/seek/seek.c: seek: don't pause for live buffering messages 2010-12-20 18:29:15 +0100 Wim Taymans * 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 * 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 * 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 * gst-libs/gst/pbutils/encoding-profile.c: encoding-profile: Add guard against profiles without format 2010-12-21 13:07:27 +0100 Edward Hervey * gst/encoding/gstencodebin.c: encodebin: Fix usage of non-container profiles 2010-12-17 16:10:53 +0100 Edward Hervey * docs/plugins/inspect/plugin-videoscale.xml: docs: Update for videoscale class changes 2010-12-20 17:46:48 +0100 Edward Hervey * common: Automatic update of common submodule From 169462a to 46445ad 2010-12-19 13:41:22 +0100 Edward Hervey * gst-libs/gst/pbutils/gstdiscoverer.c: gstdiscoverer: Don't leak tags 2010-12-19 13:22:23 +0100 Edward Hervey * tools/gst-discoverer.c: gst-discoverer: show global tags by default 2010-12-19 09:53:08 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/audioresample/gstaudioresample.c: audioresample: replaced void* with gpointer 2010-12-16 20:30:24 -0800 Leo Singer * gst/audioresample/gstaudioresample.c: audioresample: initial filter transient discarded; unit tests passing 2010-12-16 20:09:58 -0800 Leo Singer * 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 * 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 * gst/videoscale/gstvideoscale.c: videoscale: Change classification to Filter/Converter/Video/Scaler 2010-12-15 23:47:29 +0200 Stefan Kost * win32/common/libgstrtsp.def: win32: update the def file with the new rtsp api 2010-12-15 17:51:36 +0100 Andy Wingo 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 * win32/common/libgstrtp.def: win32: update defs file 2010-12-15 16:30:55 +0100 Wim Taymans * 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 * common: Automatic update of common submodule From 20742ae to 169462a 2010-12-15 12:58:47 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefind: Add check for yuv4mpeg 2010-12-13 18:05:41 +0200 Stefan Kost * gst-libs/gst/pbutils/descriptions.c: pbutils: spell out two more container formats 2010-12-13 16:20:23 +0200 Stefan Kost * 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 * common: Automatic update of common submodule From 011bcc8 to 20742ae 2010-12-13 13:04:40 +0100 Mark Nauwelaerts * tests/check/elements/audioresample.c: tests: audioresample: adjust unit test to relaxed discont checking 2010-12-13 12:34:58 +0200 Stefan Kost * 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 * gst/audioresample/gstaudioresample.c: audioresample: relax discont checking slightly 2010-12-13 09:56:04 +0100 Mark Nauwelaerts * 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 * 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 * 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) * gst/videotestsrc/generate_sine_table.c: videotestsrc: Add a missing return statement 2010-12-11 17:18:49 +0100 Sebastian Dröge * 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 * 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 * 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 * gst-libs/gst/pbutils/descriptions.c: pbutils: Add/Fix some media descriptions Fixes #623413 2010-12-09 08:40:25 +0100 Gavin Stark * 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 * 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 * tests/check/libs/pbutils.c: tests: remove superflous ';' and reindent 2010-12-08 12:09:45 +0200 Stefan Kost * 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 * tests/check/gst/typefindfunctions.c: tests: add AC-3, E-AC-3 typefind tests 2010-12-03 17:33:40 +0200 René Stadler * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefind: remove useless masking in (E-)AC-3 typefinders 2010-12-03 16:14:15 +0200 René Stadler * gst/typefind/gsttypefindfunctions.c: typefind: stop scanning after suggesting E-AC-3 caps 2010-12-03 18:08:58 +0200 René Stadler * 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 * 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 * 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 * 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 * 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 * win32/common/libgstvideo.def: defs: add new symbol 2010-10-27 13:49:41 +0200 Mark Nauwelaerts * ext/ogg/gstoggstream.c: oggstream: additional tag extraction ... supporting theora, flac, speex, celt. Fixes #629349. 2010-10-27 12:08:25 +0200 Mark Nauwelaerts * 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 * ext/ogg/gstoggstream.c: oggstream: refactor vorbis comment tag extraction 2010-10-27 11:16:15 +0200 Mark Nauwelaerts * ext/ogg/gstoggdemux.c: oggdemux: plug some oggstream leaks 2010-10-27 10:59:03 +0200 Mark Nauwelaerts * 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 * ext/ogg/gstoggdemux.c: oggdemux: send stream tags after newsegment and global tags 2010-09-14 23:08:51 +0300 Sreerenj Balachandran * 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 * 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 * ext/ogg/gstoggstream.c: oggstream: fix incorrect warning on skeleton headers 2010-11-20 19:02:50 -0800 David Schleef * 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 * 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 * 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 * 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 * gst/playback/gstplaybin2.c: playbin2: add some logging for failure case 2010-11-01 23:06:21 +0200 Stefan Kost * gst/playback/gstinputselector.c: inputselector: log times in human readable form 2010-11-01 22:44:16 +0200 Stefan Kost * gst/playback/gstinputselector.c: inputselector: more G_PARAM_STATIC_STRINGS use 2010-11-01 22:42:23 +0200 Stefan Kost * 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 * gst-libs/gst/rtsp/Makefile.am: libgstrtsp: Fix typo in .pc to use for GIR 2010-12-02 15:16:25 +0100 Edward Hervey * 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 * 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 * 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 * 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 * 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 * 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 * ext/gnomevfs/gstgnomevfssrc.c: gnomevfssrc: Remove dead assignment 2010-10-31 19:14:27 +0100 Sebastian Dröge * gst/tcp/gsttcp.c: tcp: Remove dead assignment 2010-10-31 19:11:53 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * sys/ximage/ximagesink.c: ximagesink: Add docs for the new property Including Since markers 2010-10-25 14:11:01 +0100 Sebastian Dröge * sys/xvimage/xvimagesink.c: xvimagesink: Add docs for the new property Including Since markers 2010-10-25 14:09:39 +0100 Sebastian Dröge * sys/xvimage/xvimagesink.c: xvimagesink: Use PROP_ instead of ARG_ for the property enums 2010-10-25 14:09:20 +0100 Andrea Sebastianutti * sys/xvimage/xvimagesink.c: xvimagesink: Add read-only properties window-width and window-height 2010-10-25 14:08:43 +0100 Andrea Sebastianutti * sys/ximage/ximagesink.c: ximagsink: Add read-only properties window-width and window-height 2010-10-17 14:26:23 +0200 Sebastian Dröge * gst-libs/gst/video/video.c: video: Return correct component width/height for A420 2010-12-02 00:15:25 +0000 Tim-Philipp Müller * 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 * 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 * 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 * gst/playback/gsturidecodebin.c: uridecodebin: disconnect signal handlers before disposing 2010-11-22 00:54:35 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * gst/playback/gstdecodebin.c: decodebin: fix one more pad template ref leak 2010-11-17 10:14:59 +0200 Harri Mähönen * 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 * 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 * 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 * 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 * 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 * 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 * 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 * tests/check/libs/tag.c: tests: fix leak in tag library unit test 2010-11-02 12:01:03 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggstream.c: ogg: add some more debug statements 2010-10-26 16:41:28 +0100 Jan Schmidt * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 7bbd708 to 011bcc8 2010-10-24 16:09:26 +0200 Edward Hervey * 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 * 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 * 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 * gst-libs/gst/pbutils/descriptions.c: pbutils: Description for RealAudio container format 2010-10-24 15:38:42 +0200 Edward Hervey * 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 * 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 * 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 * 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 * gst/playback/gsturidecodebin.c: uridecodebin: workaround internal decodebin2 failing state change Fixes #632656. 2010-10-21 13:38:01 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/pbutils/Makefile.am: pbutils: fix distcheck Apparently noinst implies dist. 2010-10-15 11:23:02 -0700 David Schleef * tests/check/Makefile.am: tests: Don't dist generated orc code 2010-10-15 11:22:45 -0700 David Schleef * gst/videoscale/gstvideoscaleorc-dist.c: Update generated orc code 2010-10-15 19:18:12 +0100 Tim-Philipp Müller * 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 * 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 * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: docs: improve gst_discoverer_new() docs a bit 2010-10-15 16:43:41 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: private structs need to padding 2010-10-15 11:26:50 +0200 Sebastian Dröge * gst-libs/gst/video/video.c: video: Fix stupid copy&paste error in last commit 2010-10-13 22:51:12 +0200 Sebastian Dröge * 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 * 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 * common: Automatic update of common submodule From 5a668bf to 7bbd708 2010-10-14 16:36:30 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/ogg/gstoggdemux.c: oggdemux: Fix check for last page 2010-10-10 15:22:04 -0700 David Schleef * ext/ogg/gstoggdemux.c: oggdemux: change checks from is_skeleton to is_sparse 2010-10-10 15:17:31 -0700 David Schleef * 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 * 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 * 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 * 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 * 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 https://bugzilla.gnome.org/show_bug.cgi?id=628488 2010-10-10 18:35:54 +0200 Sebastian Dröge * 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 * 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 * 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 * gst/tcp/gstmultifdsink.c: multifdsink: gdp protocol is deprecated. People should use gdppay instead. 2010-10-08 12:43:28 -0700 David Schleef * common: Automatic update of common submodule From c4a8adc to 5a668bf 2010-09-28 12:17:41 +0200 Edward Hervey * 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 * 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 * common: Automatic update of common submodule From 5e3c9bf to c4a8adc 2010-10-08 11:23:33 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * gst/volume/gstvolumeorc-dist.c: * gst/volume/gstvolumeorc-dist.h: volume: Update generated orc sources 2010-10-08 11:49:09 +0200 Sebastian Dröge * 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 * configure.ac: configure: Require orc 0.4.10 2010-10-07 23:54:57 +0200 Sebastian Dröge * gst/audioconvert/gstaudioconvertorc-dist.c: * gst/audioconvert/gstaudioconvertorc-dist.h: audioconvert: Update generated orc files 2010-10-07 23:54:25 +0200 Sebastian Dröge * 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 * gst/volume/gstvolumeorc-dist.c: * gst/volume/gstvolumeorc-dist.h: volume: Update generated orc sources 2010-10-03 12:00:42 +0200 Sebastian Dröge * gst/volume/gstvolumeorc.orc: volume: Fix controlled processing via orc 2010-10-03 11:24:29 +0200 Sebastian Dröge * gst/volume/gstvolume.c: volume: Actually enable usage of the orc optimized functions 2010-10-03 11:20:37 +0200 Sebastian Dröge * gst/volume/gstvolume.c: * gst/volume/gstvolumeorc.orc: volume: Implement int32 processing with orc 2010-10-01 12:21:52 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/vorbis/gstvorbisdec.c: vorbisdec: reverse playback; decode pending buffers upon EOS 2010-10-05 19:15:47 +0100 Tim-Philipp Müller * 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 * 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 * win32/common/libgstrtp.def: win32: update def file with new RTP methods 2010-10-05 17:13:09 +0200 Wim Taymans * 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 * tests/examples/seek/seek.c: seek: fix position reporting 2010-08-26 12:34:11 -0400 Olivier Crête * 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 * 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 * 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 * 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 * 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 * tests/check/libs/rtp.c: tests: Add test for RTP header extension functions 2010-08-20 17:13:06 -0400 Olivier Crête * 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 * 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 * 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 * 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 * 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 * 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 * tools/gst-discoverer.c: gst-discoverer: The 'ready' signal was renamed to 'finished' 2010-10-04 17:27:00 +0200 Wim Taymans * 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 * ext/theora/gsttheoraparse.c: theoraparse: set caps on streamheader too 2010-10-04 13:07:14 +0530 Arun Raghavan * 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 * 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 * gst/volume/gstvolumeorc-dist.c: * gst/volume/gstvolumeorc-dist.h: volume: Update generated orc files 2010-10-01 10:42:27 +0200 Sebastian Dröge * 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 * gst-libs/gst/pbutils/codec-utils.c: pbutils: include config.h in codec utils 2010-09-30 00:19:29 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/tag/tags.c: tags: fix unused function warning with debug disabled 2010-09-30 12:59:46 +0300 René Stadler * 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 * 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 * 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 * gst-libs/gst/rtp/gstbasertpdepayload.c: basertpdepay: ensure metadata is writable 2010-09-29 13:29:20 +0200 Edward Hervey * gst-libs/gst/pbutils/descriptions.c: pbutils: Add descriptions for more codecs 2010-09-29 12:33:44 +0200 Edward Hervey * tests/examples/seek/seek.c: seek: Fix debug statement argument type 2010-09-28 09:30:57 -0300 Thiago Santos * 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 * 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 * 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 * gst-libs/gst/interfaces/xoverlay.c: xoverlay: fix endless loop in deprecated method 2010-09-28 08:46:25 -0300 Thiago Santos * 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 * gst-libs/gst/tag/tags.c: tags: add debug category for tags utility functions 2010-09-27 14:36:17 +0100 Tim-Philipp Müller * 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 * 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 * ext/theora/gsttheoraenc.c: theoraenc: ptalarbvorm speed level goes to 3 2010-09-24 16:31:37 +0200 Vladimir * tests/examples/seek/seek.c: seek: Add #define for seekbar granularity Fixes #630496 2010-09-24 14:03:45 +0100 Tim-Philipp Müller * 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 * tests/check/elements/.gitignore: .gitignore: ignore new appsrc unit test 2010-09-24 13:09:28 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: add Since markers Fixes #630443 2010-07-30 13:54:42 +0200 Havard Graff * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From aa0d1d0 to 5e3c9bf 2010-09-20 21:04:48 +0300 Stefan Kost * 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 * gst-libs/gst/pbutils/gstdiscoverer.c: discoverer: Fix debug statement. Fixes build on macosx 2010-09-20 16:28:52 +0200 Edward Hervey * 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 * 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 * win32/common/libgstpbutils.def: win32: Update with symbols from GstDiscoverer Fixes #625944 2010-09-20 11:23:17 +0200 Edward Hervey * 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 * 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 * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Add mp3 to the apetag extensions 2010-09-18 13:15:08 -0700 David Schleef * 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 * 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 * 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 * 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 * 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 * gst/videoscale/gstvideoscaleorc.orc: videoscale: Don't use broken orc feature 2010-09-16 19:30:59 +0200 Wim Taymans * 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 * 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 * gst/videotestsrc/videotestsrc.h: videotestsrc: Fix indentation 2010-09-16 17:14:20 +0200 Thijs Vermeir * 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 * 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 * 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 * 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 * 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 * 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 * gst/videotestsrc/videotestsrc.c: videotestsrc: fix segfault on negative horizontal-speed 2010-09-15 14:15:13 +0200 Edward Hervey * gst/playback/gstdecodebin2.c: decodebin2: Actually dispose the unused ghostpads 2010-09-15 11:28:29 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * gst/videotestsrc/videotestsrc.c: videotestsrc: Add UYVP 2010-09-12 20:36:19 -0700 David Schleef * 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 * 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 * gst/videotestsrc/videotestsrc.c: videotestsrc: Clean up the RGB code 2010-09-10 14:40:44 -0700 David Schleef * 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 * 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 * gst/videotestsrc/videotestsrc.c: videotestsrc: Fix recent breakage of smpte75 pattern 2010-09-01 15:18:31 +0200 Thijs Vermeir * 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 * 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 * 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 * 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 * 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 * tests/check/libs/video.c: video: Add unit test for gst_video_convert_frame_async() 2010-09-14 08:39:20 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * gst/playback/gstdecodebin2.c: decodebin2: prevent another race with shutdown state change 2010-09-11 14:55:01 +0200 Sebastian Dröge * win32/common/libgstsdp.def: win32: Add new SDP symbols to the .def files 2010-09-10 18:42:16 +0200 Wim Taymans * gst-libs/gst/sdp/gstsdpmessage.c: sdp: remove leftover g_print 2010-09-10 17:55:46 +0200 Wim Taymans * 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 * tests/check/elements/videotestsrc.c: tests: videotestsrc change the pattern property for the tests 2010-09-10 08:42:37 +0200 Sebastian Dröge * 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 * 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 * 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 * gst/adder/gstadder.c: adder: Do debug category initialization in plugin_init again 2010-09-09 10:59:13 +0200 Sebastian Dröge * 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 * 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 * win32/common/libgstaudio.def: win32: Add new symbol to libgstaudio 2010-09-07 18:09:12 +0200 Wim Taymans * 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 * 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 * tests/check/Makefile.am: check: Fix linking order of libs/tag 2010-09-07 16:26:30 +0200 Edward Hervey * tests/check/gst-plugins-base.supp: check: Make fontconfig leak suppression more generic 2010-09-07 08:46:15 -0300 Thiago Santos * 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 * 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 * common: Automatic update of common submodule From c2e10bf to aa0d1d0 2010-09-06 18:17:10 +0100 Tim-Philipp Müller * 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 * 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 * gst/videotestsrc/.gitignore: videotestsrc: .gitignore new generate_sine_table 2010-09-06 11:44:17 +0300 Stefan Kost * 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 * 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 * 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 * common: Automatic update of common submodule From d3d9acf to c2e10bf 2010-09-05 17:04:31 -0700 David Schleef * gst/videotestsrc/gstvideotestsrc.c: videotestsrc: deprecate colorspec property Fixes: #616392. 2010-09-05 12:57:36 +0200 Sebastian Dröge * 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 * common: Automatic update of common submodule From ca1c867 to d3d9acf 2010-09-05 12:12:43 +0200 Sebastian Dröge * gst/audioconvert/gstaudioconvertorc-dist.c: * gst/audioconvert/gstaudioconvertorc-dist.h: audioconvert: Update disted orc files 2010-09-05 12:09:19 +0200 Sebastian Dröge * gst/volume/gstvolume.c: volume: Enable float processing with orc again 2010-09-05 12:08:44 +0200 Sebastian Dröge * configure.ac: configure: Require orc 0.4.8.1 for the volume test 2010-08-26 19:16:18 +0200 Sebastian Dröge * gst/audioconvert/audioconvert.c: * gst/audioconvert/gstaudioconvertorc.orc: audioconvert: Use the ORC double support 2010-09-04 09:06:08 +0200 Leo Singer * 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 * 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 * gst/videotestsrc/gstvideotestsrc.c: videotestsrc: fix typo in property description 2010-09-01 17:52:31 +0200 Sebastian Dröge * ext/pango/gsttextoverlay.c: textoverlay: Add support for AYUV 2010-09-01 11:37:37 +0200 Sebastian Dröge * gst/audiorate/gstaudiorate.c: audiorate: Fill segment until the end on EOS 2010-09-01 11:33:12 +0200 Sebastian Dröge * 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 * 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 * tests/examples/seek/seek.c: seek: allow seeking behind the currently downloaded position. 2010-09-01 10:06:09 +0300 Stefan Kost * gst/adder/gstadder.c: adder: use GST_BOILERPALTE macro 2010-08-31 10:09:51 +0200 Edward Hervey * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst-libs/gst/tag/Makefile.am: tag: Link to $(LIBM) for pow(), log2() and friends 2010-08-25 08:41:52 -0300 Thiago Santos * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ext/pango/gsttextoverlay.c: textoverlay: Support NV21 too and minor cleanups 2010-08-24 10:03:04 +0200 Sebastian Dröge * 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 * 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 * 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 * ext/pango/gsttextoverlay.c: textoverlay: add NV12 support Fixes bug #627768. 2010-08-20 12:03:44 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 3e8db1d to ec60217 2010-08-13 13:59:08 +0300 Stefan Kost * docs/plugins/gst-plugins-base-plugins-sections.txt: plugin-docs: the 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