| Commit message (Collapse) | Author | Age | Files | Lines |
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Flush the mono pipes used during voice call to avoid rendering
old data from the last time the pipe was used (e.g. when the
paired phone enables, disables and re-enables Bluetooth during
the voice call).
Change-Id: I43756abed06143522e5630ca0276ca5fbc02bb29
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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Add support for the auxiliary input available in the JAMR3
board in the Jacinto6 EVMs. The auxiliary input is a stereo
analog port.
Change-Id: I815dbcad3ea3ec47c57580a3243a631abdcbf4b9
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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JAMR3 board may not always be present in the system, the port
usage is being changed dynamically to:
- JAMR3 is present
o Cabin : port 1, slots 0 & 1
o Mic : port 1, slot 2
o Back Mic: port 1, slot 3
- JAMR3 is not present
o Cabin : port 0, slots 0 & 1
o Mic : port 0, slots 0 & 1
Above port allocation allows:
- Relinquish McASP3 (port 0) when JAMR3 is connected. This allows
DSP Radio to own that McASP instance for its rendering.
- Hardware setups that don't have JAMR3 fall back to the port 0
that is present in the main Jacinto6 EVM board (CPU board).
The default hardware setup is assumed to have JAMR3 connected.
When it's not connected, the property "persist.audio.use_jamr"
must be set to "0" or "false" and system has to be rebooted
(or media server restarted). This is meant to be a development
option.
Change-Id: Icc5399b370e2247a1a0d5a967dcd47e0ef4947ac
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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The input buffer size returned from AudioStreamIn::getBufferSize()
and AudioHwDevice::getInputBufferSize() was being rounded to a
multiple of 16 with a note that it is required by AudioFlinger,
but AudioFlinger has no such requirement for input buffer size
(only output buffer size). Rounding to 16 caused an issue for the
multizone audio when certain sample rates were used. So, the
rounding is removed as it is not needed.
Change-Id: I8d5010f5bab7eb780868b47899ea0115fa8c1c4b
Signed-off-by: Angela Stegmaier <a0866189@ti.com>
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The PCM ports used for Bluetooth SCO are in slave mode in Jacinto6
EVM board. So the BT side masters the BCLK and FSYNC. The BT side
could cut the BCLK and FSYNC while the ALSA PCM ports are not closed,
which leads to blocking read() and write() calls. They would eventually
timeout and cause an I/O error.
The workaround consists of switching the BT DAI link to slave mode
(McASP is master, BT is slave) while leaving the call, it guarantees
that the BCLK and FSYNC are present as they are generated from an
internal source, the actual rate of the clock is irrelevant.
For the new source to take effect, the ALSA PCM ports used for BT are
explicitly stopped.
Change-Id: Ifadb8b1348529d404d06d4b043c5fe686f85a49a
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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Unregister the voice call downlink stream before letting any
ongoing stream on cabin switch back its original writer.
Otherwise, it may end up trying to register two streams with
the same slot masks.
An incorrect log message is also fixed.
Change-Id: Ic5cb4d2ed0cc4de4232151df9ed8d45b2de9d5c2
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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Voice call is supported using two data pipes:
Uplink:
PcmWriter <-- MonoPipe <-- PcmReader
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McASP7 McASP3
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Bluetooth Mic
Downlink:
PcmReader --> MonoPipe --> PcmWriter
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McASP7 McASP3
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Bluetooth Speaker
Voice call reuses the PcmWriter associated with Cabin/Speaker that
is otherwise used for media. So, when the voice call starts any
ongoing media stream active on that writer has to be re-routed to
a null writer and routed back to the actual writer when the voice
call ends.
The null writer is a PcmWriter that writes data to a null output port.
The null port consumes the data at the same rate that the actual port
does. IOW, media streams going to the Cabin output (primary output)
during voice call will not be rendered on the actual hardware output.
Change-Id: I7856b67ab1cf788ea2f2cb251d9097e57e487cb5
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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Refactor the resume (leaving standby on first read/write) and
idle (entering standby) in preparation for voice call support.
JAMR3 outputs have mixer controls that do not need to change
dynamically, so they are moved to the default routes section
of the dra7evm paths.
Change-Id: I432a69ca67eff0e276ce90170dff4a5201ac181a
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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Add the multizone AudioHAL support to jacinto6evm. The
multizone audio is built if the multizone flag is set,
otherwise the legacy AudioHAL is built.
Change-Id: Ib0b401f963cd9327caeb73623bba163fe8129870
Signed-off-by: Angela Stegmaier <a0866189@ti.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
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