| Commit message (Collapse) | Author | Age | Files | Lines |
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* For the case where CALL/IN_COMMUNICATION, new_bit_width and
new_sample_rate were unconditionally reset to 0, resulting
in a sample rate/bit width change always occurring.
Don't do this for CALL/IN_COMMUNICATION as the original
logic intended.
Change-Id: Ib647cec084f4af9a044f90935dd57257bba0ec06
(cherry picked from commit cb739e9aa15cfae45d49fed57a7a5ed4cf779204)
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- Fixes FM recording where the app chooses the
AUDIO_SOURCE_FM_RX source. In this case, the
HAL was being set to configure for the
AUDIO_SOURCE_DEFAULT source, which selects
the wrong input device in platform_get_input_snd_device.
Change-Id: Ic52679404664b5a2ce3e3433cd6296df78803f30
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* Not all devices will have this code in the DSP, let alone the kernel.
* Featurize it, but leave the codec bit width and sample rate controls
in place so that 24-bit PCM offload still works.
* Enable by default.
Change-Id: I6e156048906a2b638e9ec4a083755536a14e161f
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- Add support to return error for compress time stamp query
during SSR
CRs-Fixed: 683288
Change-Id: Ie6849bbd3de9474fa556bfe4b183a10a44e4b3e8
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- Specific AAC clips are skipped to next clip on triggering SSR
- When SSR is triggered while track is waiting for stream end event for
low bit rate clips, drain is unblocked and playback is switched to
next track
- Fix is not to post drain ready event during SSR as actual stream end
is not reached with more compressed data buffered to DSP
CRs-Fixed: 766541
Change-Id: Id1573160001a2a252dc6613b58f70233e77cc3b4
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Ported from msm8974 original patch
Change-Id: I6be3870667daf5273db5e8fbb1aa39e5e14c1530
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Change-Id: I02b2f742c60e78d2342297917724af132047cf83
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Change-Id: Idc3e77017f54167001e497c7e1a2ea86f8f6233f
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* Supported sections now include device names
* This allows for aliasing device names to a custom name
Change-Id: I880e90a7e887f020517d89ba276199c700c0eeae
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- SA+ topology is selected for for incall recording use case.
- HAL selects default app type for incall recording use case,
which results SA plus topology selection
- Add support for app type selection based on use case type
instead of use case id
Change-Id: I0ce017ea4dca04eb25055e669e72bda0c281b22c
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-For all the rx/tx sessions hal sends the default app type 'general playback'
with acdb cal data if output policy conf file is not added.So it sets the SA+
topo id for POPP.
-With output policy conf file it would create the
multiple instance of COPP for different output with different app type,
which is not recommended for resource constrained target.So we can not
add the output policy conf file with different app types to avoid SA+.
-Overwrite the app type for PCM Rx and Tx path to avoid setting SA+ for POPP.
Change-Id: I1a3e0cbb01360e4d2e7b0ba0ecf8a1bee7fecda7
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Enable low latency use-case for Bear family
CRs-Fixed: 752390
Change-Id: I7cac5a0166ee9e19ab8753f567bde712ca36ebb3
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* Offload enhancements, 24-bit, etc
Change-Id: I728154c2c01650ea88b994b0ad73711694c64070
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Currently the deep buffer path uses 4 * 20msec buffer.
During device switch the AudioPolicyManager delays routing
command by 2 * output latency (80msec) to ensure that all
the data written to driver/DSP is played out on current
device itself.
The stream side buffering in the DSP in legacy PCM mode is
95msec and hence the depth of pipeline is 80 + 95 = 175msec
which exceeds the device switch delay 160msec (2 * 80). So
the tail (data written to driver before headset plugin) is
heard on headset and perceived as glitch.
Ensure that the buffering in the kernel is greater than or
equal to the buffering in the DSP to fix the issue.
Change-Id: I01a3862d63ce4c258056620693dee08761c7e83f
CRs-Fixed: 771446
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When pcm offload is done, override the buffer size
that was calculated and use the value from the system property
Make write call blocking if small buffers are used in offload
Update latency value for pcm offload with small buffer hint based
on period size and period count.
Change-Id: Ic74caa6bd172c8e4554384e9fa98a5137117f07c
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During MP3 (with no gapless meta data) playback in 'repeat one song'
mode observed that from second iteration onwards the progress
bar continues to increase beyond the clip duration.
At the end of second iteration the drain command is not reaching the
driver as the compress_set_gapless_metadata() is not called.
Fix the issue by ensuring that it gets called for every iteration.
Change-Id: If71c145c5a02c99ff55f528522f1f36e20ec8871
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- Initialize default return value of out_get_render_position to
0(NO_ERROR).
- Default value(NO_ERROR) will be used if timestamp query happens
before compress driver is opened. Return 0(NO_ERROR) in this case
to avoid playback failures.
Change-Id: I0f8b2e0f19cfe736a19934ddef18016599ec582c
CRs-Fixed: 756508
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* Don't try to send the app type config during verification as the
streams aren't set up yet and will crash due to null pointers.
Change-Id: Ic4cdd5374b46b3e3c2042b158820af5ec020e754
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Support bit width selection for playback on speakers based on
platform info.
Change-Id: I8dd2cc7049186e7468615a43db550d7d580a81d1
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Support all compressed formats as mentioned in
audio_policy.conf for app type selection in compress_offload
section of audio_output_policy.conf.
CRs-fixed: 767955
Change-Id: Ifea1643429f1f9f5fdb587d4dfc4303e9af9ff40
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* Always include these unless explicitly negated
Change-Id: I3931ef089dddcd9ac9f61d8b7e870cf916a4e47d
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* Remove unnecessary ifdefs for PCM offload
* Remove unnecessary property for 24-bit support (use audio policy)
* Clean up warnings
* Also includes these patches from Google:
audio: fix set_parameters return value.
xxx_set_parameters functions were returning the status
returned by str_parms_create_str() which is incorrect.
These functions should return 0 when no error occurs.
Change-Id: Ib4a7ac427e49f5500c99902f86d2d69d5843eda0
Scan and verify audio device parameters on open
Scanning is default disabled at this time.
Verbose logs will display device params found.
Change-Id: Id188d096ec68d2058c66ae3a51fe57d9645d03ef
Signed-off-by: Andy Hung <hunga@google.com>
audio: deprecate audio_stream_frame_size()
Bug: 15000850.
Change-Id: I1bbe614c241befa24513a2b583594680e32fd954
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Change-Id: I048c15e2a0e0c417f61f4001a0b7ccc6d64df0f1
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When SSR happens the callbacks between sthal and ahal
are called within each context aquiring the locks resulting
in race condition. Fixed by removing lock before calling
the sthal callback.
Change-Id: I6cd88d25758f9c0a6ca39141df6629e2f8b6c0c5
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-Fix combo device configuration for 24 bit playback on speaker
-Move exception hadndling code to audio_extn
Change-Id: Ia0985a284042a5ac5e3de64aaf5e4d57462ceb85
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Check for 24 bit in HAL is not needed as the check for 24 bit
platform support is achieved through flags in frameworks.
Change-Id: Icc590dfc1c4b831399435c19dd505ddebf6503cd
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new system property is used to store FFSP enable/disable in persist memory.
user has to enable FFSP in both acbd file and system property.
Change-Id: Ia1a95515bac2e85cd33b944dba2edcba0712c5bd
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Add support for routing voice calls to devices in other audio
HALs by allowing playback and capture to/from AFE proxy
Change-Id: I2c6a1ddec072e1d5f1a8b7ded874e9c082a7b810
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- add support for low-latency capture
Change-Id: Ic8a82854799adfa4eb1fcd323e0177eeadc7e319
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Push the codec driver calibration using hwdep nodes.
Change-Id: I08875c543be5b69c6cf0d0bbc248806ae2f871c2
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thermal_client_unregister_callback pointer should be checked
for NULL before it is dereferenced.
Change-Id: I3b3a46c93c445a723a7db8a64c727c08f6ca5caf
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Sound Trigger App uses setParameters() API to pass the number of
sessions to be configured to STHAL.
Change-Id: I14abff49863d82178fb633c9a93af0cd02c83e2e
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Added new ACDB ID(131) for playback usecases to differentiate
with voice calls, so that FFSP can be enabled selectively
for playback usecases.
Change-Id: Iffae085862b422d99a49c3e774749c38cc0a7b82
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Headset speaker output device and Speaker phone mic input device
is an invalid device pair combination. Instead, select Handset mic
input device for Headphones.
Change-Id: I1bce425bd73d2f79b7dd78a7d5c701c43390b167
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- fix warnings for speaker protection in audio_extn.h
Change-Id: I4034b68e80b5378dd2a8d8ebbfd9a9ef89804024
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- use PCM_OUTPUT_BIT_WIDTH instead of out->bit_width to fix
compilation errors while it is not defined in audio.h
Change-Id: I5c494c622652226f5fd468b073bc29bdcc32d9b3
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