| Commit message (Collapse) | Author | Age | Files | Lines |
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The cause is that the thread executing setMicMute() can wait for
several seconds before acquiring the input thread mutex when audio
capture is active.
The fix consists in forcing a sleep of the capture thread when
standby() (called by setMicMute()) wants the mutex.
Applied the same fix to output stream and to setParameters() function.
Change-Id: I3e55670d2aad16b67d44ca8582ed16398143ff6e
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modem side doesn't store volume setting when device is rebooted.
When audio mode is incall, set current voice volume to modem.
Current incall volume is stored when upper layer sets or AudioService is started.
Change-Id: Icfeb0273dda55354e207d464884ef99f41c5fdec
Signed-off-by: UK KIM <w0806.kim@samsung.com>
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Change-Id: I0e410d7b641c19a773c3bc57cb78e8aa3e27a3ff
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The problem is that the AudioPolicyManager refreshes the device attached to the
hardware output stream every time another output stops (in this case A2DP output).
As there is no active stream on the HW output, the selected device is 0.
As the shutter sound is short and there is some additional delay to setup the A2DP output,
the sound is not yet out of the AudioFlinger mixer when the device is set to 0.
On Crespo, device 0 is a valid audio path (means audio off) which is not the case
on other platforms and therefore the output is disabled while the shutter sound
is actually playing.
The fix consists in ignoring requests to set the output device to 0 in the audio HAL.
Change-Id: I7366e359a7d3a0f8207e7a5c879ced9078224002
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Change-Id: I60e8ae30da4c3879bdd6a272dd65f6add4d1f520
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Rely on the audio input source parameter for mixer configuration.
The generic APM implementation now passes the audio recording
source as a configuration parameter, and uses the enum defined
in mediarecorder.h. But the driver uses a string to define
the "input source state". This change maps the input source
to the string used by the driver.
Change-Id: I5fce44579a3cda01ed73f67fb8c3091ef05cce76
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The problem is that when the voice search tone is started, audio capture
is still active and the output stream write function needs to place
the input stream in standby to reconfigure the kernel driver.
To do so it has to acquire the input stream mutex but as the input stream thread
holds the lock most of the time while sleeping in the driver waiting for more data,
this is very difficult and can take several seconds.
The fix consists in forcing a sleep in the next read() when another function needs
to acquire the input stream lock.
The same change is done for output stream write() function.
Also removed the workaround for issue 3201189 in setMode() (thread priority bump) as this
change addresses the same problem.
Change-Id: I3a5e672717752f83dfedce822a18748b165b0a5a
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Use only 15 bits for fractional part of the increment in resample_441_320()
to avoid overflow when multiplying by the difference between previous and next
samples in interpolated value computation.
Change-Id: I9f5a726d11f6b051db390df3d13312f1ee782d3a
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Change-Id: I0a6492f7c834ea572531e77f75486bcc385e345b
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There is no call audio uplink when headphones w/o mic are connected.
This change together with a corresponding change in the kernel driver adds
a separate device for headphonjes without mic: output is routed to headphones
and input is routed from built-in mic.
Change-Id: I19955f76ece19f661ae25d6a42bbcbe235a9e652
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Anton Rogozin <ant.rogozin@samsung.com>
- Move RIL clock sync to starting of ringtone mode as requested by modem team
- BT noise reduction turning off support
Change-Id: I95a8157ca0da7a4432fe0d5bc3a4adba94cdb19a
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Fix problem in audio HAL AudioStreamInALSA::read() function
when the down sampler is used and a read error from kernel
driver occurs: the requested frame count should be reset
before calling the resampler again otherwise we loop
for ever requesting 0 frames.
Change-Id: Ie85f7a1db4e417f5c1d97c0f0e0f5a28a62ee92a
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Latest audio HAL downsampler implementation had a problem that
caused one sample to be dropped if the input buffer size was odd
while downsampling by a factor of 2.
This explains why record timing was correct for 22 and 16kHz but
not for 11 or 8kHz: in the later case, the second stage of resampling
(22 to 11) receives 503 frames for each buffer.
Change-Id: Ib8fcba3ddfbbab50a908e6b0a6cdc2b398acd862
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Previous input stream downsampler implementation was very cheap
and for functional tests only. The quality was not suitable to
voice recognition.
Integrated a higher quality resampler handling conversions
from 44100Hz down to 22020,16000,11025 and 8000 Hz.
Change-Id: I5d6de5c137717e02ca6024c852c9a67285fd2df5
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Crespo libaudio fails to use the Voice Recognition configuration
when doing a recording on the VOICE_RECOGNITION input because
the associated setParameter always returns before the value is
set due to a cast error.
Change-Id: Icc7c7edb5f680de82140d6ece4e536c0d9cb2419
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Change-Id: Ia714d2f16205773c7e30db1d9e03aba6b8f08cf7
Signed-off-by: DooHyunHwang <dh0421.hwang@samsung.com>
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Lakkyung Jung <lakkyung.jung@samsung.com>:
S5PC11X: SOUND: Change h/w parameters setting in setDevice().
Audio input/output stream will be used snd_pcm_hw_params_set_period_size()
and snd_pcm_hw_params_get_period_size() when setting h/w params.
because we already know period values by bufferSize / periodSize.
So we don't need to set period values.
audio input stream will be used period value 4.
Anton Rogozin<ant.rogozin@samsung.com>
libaudio: RIL clock functionality
RIL clock functionality was added.
Change-Id: I9f8f8d8b2851562e3c1bed3e4c7ae896d7865331
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Audio input stream can not be opened because function
snd_pcm_hw_params_get_period_size() now returns an error.
Probably due to one of the recent kernel commits.
The result of this call was not used anyway with current patch
so removed it.
Change-Id: I36f2dbdc94e6ec726d9c0df063d0ec3702c1b94a
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Change-Id: I6819ea28e0d16f163a00f8d25ba77506b144a886
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- 1: make sure that we restore normal mode in codec when the
call is ended.
- 2: make sure that we set in call mode in codec when the call
is started even if output stream is playing.
Merged support for special mic gain setting for voice recognition.
Some more cleanup.
Change-Id: I00ab773ec294f7b186b463177372ece5b520a386
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Anton Rogozin <ant.rogozin@samsung.com>:
Unused code removed, dynamic loading of RIL lib, alsa lib resampler turning on
UK KIM <w0806.kim@samsung.com>:
1. deleted unneeded standby_l() and set_wakelock func() in each stream class.
2. use StreamOPS:close() to close stream.
3. reference new libaudio.
Eric Laurent <elaurent@google.com>:
Some more cleanup
Fixed clicks when playing call ended tone
Change-Id: Ieea1319262576b2f6680c675957643eacbab9c11
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Patched audio HAL to allow minimal audio input functionality
until real fixes are submitted.
This will allow features like voice search, VoIP, camcorder and
mms audio record to work (with poor audio quality).
Change-Id: I18acb120c5c398ccaeac11c462d629d41e33eef4
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AudioPolicyManager implementation must use AudioPolicyManagerBase for all
default behavior. Crespo was using an implementation of AudioPolicyManager
based on a pre Eclair version that did not derive from AudioPolicyManagerBase
and thus was missing a lot of improvements and bug fixes.
Change-Id: I0dd8de15047109d1534e532c924e9beb46fd23e6
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Use a different name for the stub than for the real
library, so that the real library still appears in the list
of proprietary files.
Also, use the real library instead of the stub when building
an open-source build with the proprietary files.
Change-Id: I570e7b86be5be5c8e18326c9438b5736f6019d44
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This allows to build libaudio even if we don't have
the true libsecril-client
Change-Id: I3f8c8aa975a96a144157505272cf9a4497bef196
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getInputBufferSize() function was added to AudioHardwareALSA for supporting of other sample rates in addition to 8 kHz
Change-Id: I0bb7aa333f7c51135a3bc8b65c5e5528946c48d3
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fit latest code to enable A2DP feature
Change-Id: I9e11a6e611efd9443fdecd496238efe575358c75
Signed-off-by: UK KIM <w0806.kim@samsung.com>
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Two benefits:
-this follows the convention that all Android.mk files are included,
i.e. that reading an Android.mk file shows the entire picture.
-this allows some of the leaves to become non-conditional, to increase
the number of device-specific modules that are compiled in each build,
which allows catching build breakages earlier.
Change-Id: I060a4aecc4b5d53b9ca56f13b8599b8fa560f896
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disable BOARD_USES_GENERIC_AUDIO feature
modified libaudio, alsa-lib, alsa-utils to build according to crespo's source tree.
Change-Id: I3f8b57e60c1ddd691ab9d85a161705a833155e4e
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Change-Id: I4a6ee248b407c67682eb8884d1176f4807288c7c
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