| 68f5482b51dc991b6bc2e7c3f269a0d752c3f765 |
15-Jan-2015 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to lmp-mr1-dev
|
| e406d08e77f79a66caba291dcf2b3c929f39457c |
14-Jan-2015 |
Eric Laurent <elaurent@google.com> |
fix ACDB ID for headset mic with AEC
Bug: 18987396.
Change-Id: I3df6949f2ad145c1fa8cb55b77656deb209d761d
al/msm8974/platform.h
|
| f89368bde237b4d74802d9a9c11055b8a5a55b43 |
08-Jan-2015 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to lmp-mr1-dev
|
| 1c7637efea6eb40a796031685381816711231e80 |
08-Jan-2015 |
Eric Laurent <elaurent@google.com> |
add line out to device sharing codec backend
AUDIO_DEVICE_OUT_LINE was not listed in
AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND causing a loss of audio when
music was playing over line out and a ringtone playing over lineout and speaker
was stoppped.
Bug: 18903885.
Change-Id: I8c132db9cbfb17842a463600a2d0ced214244b72
al/msm8974/platform.h
|
| 0acf84469fcf19b9b3b73ce47a580623ac16dc1f |
04-Dec-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to lmp-mr1-dev
|
| cc4f6bfbf96f1df97a3ed0882446e5157810a7df |
03-Dec-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix progress bar issue during compress offload playback
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.
Bug: 17405549
Change-Id: If71c145c5a02c99ff55f528522f1f36e20ec8871
al/audio_hw.c
|
| b8667d91b80a599bc573432e32339e9b8b006913 |
02-Dec-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to lmp-mr1-dev
|
| 8652b46b27d7c7d70afed7be5ad8fc2d7b1e49b2 |
15-Nov-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
visualizer: do not apply calibration on audio routed to afe proxy
When compress offload playback is active, if the Visualizer effect
is enabled, decoded PCM audio is routed to AFE Proxy port and
read from it by the Visualizer wrapper library. When audio is routed
to proxy port, current output device specific calibration is also
being applied which is not desired.
Avoid this by sending default audio calibration i.e. no post-processing
to be applied the data.
Bug: 18390493
Change-Id: Id576c4ed7bbb482683074e3e33aa5760b7597d37
isualizer/Android.mk
isualizer/offload_visualizer.c
|
| 6b99a88bee33366e289b60a97d1ccb09e095994d |
27-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to lmp-mr1-dev
|
| f28290178598c01d29f152a46483e5e8bd67b0e0 |
13-Nov-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix EC not working in VoIP calls
Ensure the EC reference path is reset while switching device
for VoIP calls.
Bug: 17986908
Change-Id: I7eca6842ee2ba298493cb1cd479f90318a437e12
al/audio_hw.c
al/msm8974/platform.c
|
| fbbf48c430baad8683a3f53c1ad42455e87da8a3 |
08-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| 4053150c214c05f0cf9d515781e3b23474d1c115 |
06-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| 937723ef765f34b91654eed61da7ba8b90b0cc94 |
05-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| aa9091d45894b1d7f15b84abaa881f358334244f |
04-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| b6287fb76974d35a60ac18600ddf80ce1c72f8f6 |
03-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| e61b4dd90349563e029fc808bef447dbcdfbbffa |
02-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| 5e7012ceca6d4a7a12fdc635590a261dd2f63c19 |
01-Nov-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| 852a3f4816576fc4bb2fbbace866cd51b2844369 |
30-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 5f113addd92350a8b2a48b9adfb5804158278d24
|
| 5f113addd92350a8b2a48b9adfb5804158278d24 |
29-Oct-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "compress offload: use new sample rate representation." into lmp-mr1-dev
|
| ddc8faea5471e6d30265a268216a3468614faa09 |
29-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to d2305287be8947b4f3059795f5aab8d206ef48e4
|
| d2305287be8947b4f3059795f5aab8d206ef48e4 |
28-Oct-2014 |
Eric Laurent <elaurent@google.com> |
audio: use pcm_ioctl() from tinyalsa
Bug: 18137488.
Change-Id: Ic0887837edaf2135ec195cbf863d0840ffdbc521
al/audio_hw.c
al/audio_hw.h
|
| 34d2c396af179a5f80eec77563d78947462a5e46 |
28-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to b09e4a04c7a8ca770affbf48f154222ccd083f4b
|
| e8543b886a1d0d263c92a07de68dd0a194afb9cc |
27-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to b09e4a04c7a8ca770affbf48f154222ccd083f4b
|
| d8e977456a0c7f5e41479cc08569d4b4331998a2 |
26-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to b09e4a04c7a8ca770affbf48f154222ccd083f4b
|
| 4fe7dfbc6e443c65cd03d6100f5704359a1c19a2 |
25-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to b09e4a04c7a8ca770affbf48f154222ccd083f4b
|
| 1ccf397856de169071b9290bf2f6a4a5bb7c6371 |
23-Oct-2014 |
Eric Laurent <elaurent@google.com> |
compress offload: use new sample rate representation.
Pass directly the sample rate value to struct snd_codec
instead of the ALSA enum.
Bug: 17398311.
Change-Id: I79483773807ce3b0b146fde28d6498444c69fe89
al/audio_hw.c
|
| 99620ac4176efd83bc8d2dc4e233ffd7cd73b6d0 |
23-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to b09e4a04c7a8ca770affbf48f154222ccd083f4b
|
| 8feaa22cbd324cd00c4d837bdfed5855ebb7c9ad |
22-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to cefbbac40220d4e7690d204eecff5f6ae3f63069
|
| b09e4a04c7a8ca770affbf48f154222ccd083f4b |
21-Oct-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: use consistent in call indication
Use voice_is_in_call() instead of mode == MODE_IN_CALL
as a valid indication that we are in call to choose sound
devices.
Bug: 18058600
Change-Id: Iefa968ee463d4ade6c7d09626be667faab6eda98
al/msm8960/platform.c
al/msm8974/platform.c
al/voice.c
|
| fb3101dbf077a771102effa048610bd187aa5d38 |
21-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to cefbbac40220d4e7690d204eecff5f6ae3f63069
|
| 9f7813eedc72ef53a2d6b049315060a4cbe65dbd |
20-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to cefbbac40220d4e7690d204eecff5f6ae3f63069
|
| a60dc802302d7925a585a3358e2586f2a3cb95dc |
19-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to cefbbac40220d4e7690d204eecff5f6ae3f63069
|
| 0756f9c31f6d37319ce9f49401f5974dfb7618c9 |
18-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to cefbbac40220d4e7690d204eecff5f6ae3f63069
|
| 077d2653a1fbdb99b9e2dcaec0905f21e5b72bda |
16-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to a7657196b09d78ffc5d211af5771c66d416354ee
|
| cefbbac40220d4e7690d204eecff5f6ae3f63069 |
04-Sep-2014 |
Eric Laurent <elaurent@google.com> |
hal: update EC reference handling
Change-Id: I745e28c14902f810754887f9db195cf4f5261713
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.c
al/msm8974/platform.c
al/msm8974/platform.h
al/platform_api.h
|
| e4359333269d37e18e2a56cef4036777a25a0a61 |
15-Oct-2014 |
The Android Automerger <android-build@google.com> |
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|
| ba97d2461dbbd28145d3d2d29665f3bed7132f55 |
14-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to a7657196b09d78ffc5d211af5771c66d416354ee
|
| fcdfa3c3245223b139696b74f5e2b9b08f69aeb1 |
13-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to a7657196b09d78ffc5d211af5771c66d416354ee
|
| 8ac447f0d4960d3ac34b7e85f78065518526b2b3 |
12-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to a4fc90203ca4c60175b016c4d48438b985a71462
|
| 0e262f52348e36573568f1f755b5b607c14ae0fc |
11-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to a4fc90203ca4c60175b016c4d48438b985a71462
|
| a7657196b09d78ffc5d211af5771c66d416354ee |
10-Oct-2014 |
Eric Laurent <elaurent@google.com> |
audio: use consistent in call indication
Use voice_is_in_call() instead of mode == MODE_IN_CALL
as a valid indication that we are in call to apply
a new route via voice extention.
Bug: 17940126.
Change-Id: I672f2420968e7ffe8c21c9994c5996b15b93ea07
al/audio_hw.c
|
| a4fc90203ca4c60175b016c4d48438b985a71462 |
09-Oct-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: update handling voice call driving output
For device switch from USB headset to speaker or vice-versa,
it is not required to completely tear down the call and restart
on the new device.
Bug: 17573788
Change-Id: I5ccdc225e19a7036dd8c6a028f69a505c9a3634d
al/audio_hw.c
al/voice.c
|
| b607d789345372316e68b97b6f7b15226f961539 |
09-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 744996b561b92473cc8ba23275811eb1a6b44d5e
|
| 1708e22d1f8c8280b389818f7f746d3b0724bfc1 |
08-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 744996b561b92473cc8ba23275811eb1a6b44d5e
|
| 225f897a48a145e9fd0c842fdd42db86f1fb6183 |
07-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 744996b561b92473cc8ba23275811eb1a6b44d5e
|
| 87ceeb16c058196047a08d8305fd0882c3f1a199 |
06-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 744996b561b92473cc8ba23275811eb1a6b44d5e
|
| a57cd58be651b503e1e5d2d273440930b0c42cae |
05-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 744996b561b92473cc8ba23275811eb1a6b44d5e
|
| 84e7575b00f6bf9407d5b1918ce7dd7b0305e4ba |
04-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 744996b561b92473cc8ba23275811eb1a6b44d5e
|
| 8f1415f852dbf3c387e4aace523e0a72ef235354 |
02-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 36886fcdc57683b8a3d08edc59fa5a8e8f5f461a
|
| 744996b561b92473cc8ba23275811eb1a6b44d5e |
01-Oct-2014 |
Eric Laurent <elaurent@google.com> |
hal: updates for DEVICE_OUT_LINE
- support dual-route
- change ACDB ID to 77 for gain tuning independent of headset
Bug 17722311
Change-Id: I5c574dc08e26fa053f60337acb17fb5b73ebbaa1
al/msm8974/platform.c
al/msm8974/platform.h
|
| fbf14367ecfdfdd05216336ea941bd78245eb2c6 |
01-Oct-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 01fd3fbe9be158410b983d75ebc54b8c5cd0515f
|
| 36886fcdc57683b8a3d08edc59fa5a8e8f5f461a |
29-Sep-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: stop voice call when mode is set to AUDIO_MODE_NORMAL
Do not wait for a routing command after mode is set to NORMAL to
disconnect voice calls.
Start voice call use case only when mode is set to IN_CALL.
Bug: 17687327
Change-Id: Ida33fd6b215fcc70c0afe510b8f0a0d90496385e
al/audio_hw.c
al/voice_extn/voice_extn.c
|
| ccefe8a63f5e4d1bf51238132eac4e3d40599532 |
30-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 01fd3fbe9be158410b983d75ebc54b8c5cd0515f
|
| 4913be4d808fd5a6df9d181166f2299f0afea4f5 |
29-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 01fd3fbe9be158410b983d75ebc54b8c5cd0515f
|
| a6d6de5607f78fafda75963d246642aa67eb8229 |
28-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 01fd3fbe9be158410b983d75ebc54b8c5cd0515f
|
| f416bb175f0fc361481ee7c7e5e3f0bdd11751ff |
27-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 01fd3fbe9be158410b983d75ebc54b8c5cd0515f
|
| 01fd3fbe9be158410b983d75ebc54b8c5cd0515f |
23-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix audio mode transitions
Allow direct transition from IN_CALL to IN_COMMUNICATION
modes by exiting in call state when transitioning to NORMAL or
IN_COMMUNICATION modes instead of only when transitioning to NORMAL.
Bug: 17591576.
Change-Id: I2a915df0b283b311b8cbec0fa9cd8573f76d4686
al/audio_hw.c
|
| 40e64b5ab4238a0ac2a72ecdc41513e866f73933 |
25-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| 4a7f0a79294acfe6f8e51fbdf326864d44f1415d |
24-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| 8807c5f8a02e924a1e45e783eb5f305dcf03965d |
24-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| d7e77e82c09b06830b801312eb9d94cec2222635 |
23-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| be1096a2770a62c5756e2c4d2c3d376f56bd6929 |
22-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| 7e407d4242b2e8f36104e8b8fc66eacb20f44428 |
22-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| 0789deddb1fdb2c70ee8ff8a9914d4329162f887 |
22-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-mr1-release history after reset to 4920d3856ef7e8d678329d88276b5dcc78a6aa23
|
| 4920d3856ef7e8d678329d88276b5dcc78a6aa23 |
15-Sep-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audio: fix mic mute for SIP calls" into lmp-dev
|
| 6402646dbd962f62c37abbb1d435907d54da77dd |
15-Sep-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: enable more default logs
Enable minimal default logging to know audio routes and sound
devices being enabled or disabled.
Change-Id: Ia5c67a654fc2f36bcb7cf722706a573b9e6c0a04
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
|
| 539cfc3e68672528973cc6cb2bb7c96c9629c9d1 |
15-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 4cc4ce1a92a47fa1d98c884a33e5e93d3d468b15
|
| 7b2b5ab2921c4679455faa3bfbeb1457e5eb5f0d |
14-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix mic mute for SIP calls
Do not allow mic mute only when in PSTN calls.
Bug: 17321604.
Change-Id: Ie82d4ea4de767c38e5d476d5827ebb9ae0c45c63
al/audio_hw.c
al/audio_hw.h
|
| 37d05b26c5a23ed82dea782ecb9003f449a7bf05 |
14-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 4cc4ce1a92a47fa1d98c884a33e5e93d3d468b15
|
| 2c6c2deb4d838206d3cd94a2779881823f7c9180 |
13-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to dfce498c011e29d3bd1255ddfd6915a9c2dca799
|
| 540542dbcc69d0b798b0db5d1cfdf42d05b9b11e |
13-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 4cc4ce1a92a47fa1d98c884a33e5e93d3d468b15
|
| dfce498c011e29d3bd1255ddfd6915a9c2dca799 |
10-Sep-2014 |
Eric Laurent <elaurent@google.com> |
hal: pass voice volumes to ext speaker driver
Bug: 17203285
Change-Id: I1b9bdc3a49fa162ac85b7b1c1b8de027a20983d2
al/audio_extn/audio_extn.h
al/audio_extn/ext_speaker.c
al/audio_hw.c
|
| 4cc4ce1a92a47fa1d98c884a33e5e93d3d468b15 |
10-Sep-2014 |
Eric Laurent <elaurent@google.com> |
hal: pass voice volumes to ext speaker driver
Bug: 17203285
Change-Id: I1b9bdc3a49fa162ac85b7b1c1b8de027a20983d2
al/audio_extn/audio_extn.h
al/audio_extn/ext_speaker.c
al/audio_hw.c
|
| 506ab0aff292f3caab502438a67bfc84a6bc87a2 |
23-Jun-2014 |
Jon Eklund <jeklund@motorola.com> |
hal: Support "safe speaker" on msm8974 platform
Used for limited-loudness use cases
Bug: 17319721
Change-Id: I408b32428b4af08c7a78682f51e0563167a1d3ef
al/msm8974/platform.c
al/msm8974/platform.h
|
| 1b0d8ce4ec97a57b38a401dbd1e16ac5cf4c4671 |
11-Sep-2014 |
Eric Laurent <elaurent@google.com> |
hal: Support "safe speaker" on msm8974 platform
Used for limited-loudness use cases
Bug: 17319721
Change-Id: I31c4b21e593b9d69bc0e1a81bb46f6201ffe0d7d
al/msm8974/platform.c
al/msm8974/platform.h
|
| 7aec9d7b2159c9caa599340e8b5996187ba8cb92 |
11-Sep-2014 |
Eric Laurent <elaurent@google.com> |
Revert "hal: Support "safe speaker" on msm8974 platform"
This reverts commit 4933ff67b2a41e6cefce7cfa3f9aa07389af092e.
Change-Id: Icd2a2414ccfa7c47072bc00583472792ebe34e33
al/msm8974/platform.c
al/msm8974/platform.h
|
| 7053b15741e4306fffc0bbbe7de087028984e8a3 |
11-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| 4933ff67b2a41e6cefce7cfa3f9aa07389af092e |
23-Jun-2014 |
Jon Eklund <jeklund@motorola.com> |
hal: Support "safe speaker" on msm8974 platform
Used for limited-loudness use cases
Bug: 17319721
Change-Id: I408b32428b4af08c7a78682f51e0563167a1d3ef
al/msm8974/platform.c
al/msm8974/platform.h
|
| cfd3a389fef051d73f2ee5173abe2995fc6fb02b |
10-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| 416372240aa6d964860456f5edd3c09d9c0475be |
09-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| e11432233a2bca075184e73ef89dd8b16f0cda02 |
08-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| 89c0713564d98ed55aa9f9a60c5f72a3a1ac1c74 |
07-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| 0cc6d4cada53b630d086d22aeab70aad083bb0c0 |
06-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 22c825ad2c212c8e25218dd39b90e228753e8680
|
| 984cbc0c9563a3ddb43731843d2bf75355ea1035 |
06-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| 22c825ad2c212c8e25218dd39b90e228753e8680 |
04-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699
|
| 0499d4fc40cafb9fbb0e8eace1657cfaf79c5699 |
26-Aug-2014 |
Eric Laurent <elaurent@google.com> |
hal: add support for external speaker driver library
Change-Id: I599c96793097ab6202412cbd55c8e9165c226eed
Bug: 17319721
al/Android.mk
al/audio_extn/audio_extn.h
al/audio_extn/ext_speaker.c
al/audio_hw.c
al/audio_hw.h
|
| 39ace3da692ef1f05ed14b4a5f2dbd7538276119 |
03-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audio: stop call before changing playback path route
When an output stream receives a routing command after the
audio mode was changed back to normal, we must first stop the call
and then apply the new routing to playback path. If the playback
path is applied while still in call, the logic will not change
current call use case routing.
Bug: 17016230.
Change-Id: Icea84c017e492e6524dbd4431f8c2a4ef8828c72
al/audio_hw.c
|
| 742b05d69befbe99cf0071e63810af8c9a4d9ec1 |
03-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to b9df188a8938144b72e11f6b57fab21ed2373c67
|
| d9a873922fc4f8886fb72bbb48500a63d0e34e8b |
02-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to b9df188a8938144b72e11f6b57fab21ed2373c67
|
| 02ea33a751c3f14f1fa42b67d17723419ed93338 |
01-Sep-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to b9df188a8938144b72e11f6b57fab21ed2373c67
|
| 44cf39a97d0c6c331b197619ee71fa2987e4769e |
31-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to b9df188a8938144b72e11f6b57fab21ed2373c67
|
| b9df188a8938144b72e11f6b57fab21ed2373c67 |
30-Aug-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audio: fix voice call crash on 8084 platform" into lmp-dev
|
| 9ef1bd649de7ae64047de29c389fb8a1a9b1dd71 |
30-Aug-2014 |
Eric Laurent <elaurent@google.com> |
Merge "Revert "Revert "audio: add support for routing to/from voice TX/RX paths""" into lmp-dev
|
| 54e0a442df07288b6e09a8b7839b215aa2fd24cd |
29-Aug-2014 |
Eric Laurent <elaurent@google.com> |
hal: fix headset path
Inserting HS without mic should select speakerphone mic
instead of handset mic
Bug: 17307386
Change-Id: I66d3b9037bbf7da5ff4bfebf2567efa3a0866074
al/msm8974/platform.c
|
| b991fb007d870cbd682273e5fc7413b9d238a6d9 |
29-Aug-2014 |
Eric Laurent <elaurent@google.com> |
hal: fix headset path
Inserting HS without mic should select speakerphone mic
instead of handset mic
Bug: 17307386
Change-Id: I66d3b9037bbf7da5ff4bfebf2567efa3a0866074
al/msm8974/platform.c
|
| 75a2e1d2119c21c043362d53208f4c46d8c556a6 |
29-Aug-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix voice call crash on 8084 platform
Fix voice API incompatibility introduced by
commit 8edfd66a.
Bug: 16949514.
Change-Id: I6ef5d12e6a3d54e7cf084c802b54b07f6bccce5c
al/audio_hw.c
al/voice.c
al/voice.h
|
| 99c752d87eb818fc3cfb2e5c6790b1ea0bc88da5 |
21-Aug-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
Revert "Revert "audio: add support for routing to/from voice TX/RX paths""
This reverts commit 55a1293b422d181281cf0f7d37c6c15c5d921ef3.
Bug: 15520724.
Change-Id: I46c2402bedd513c148b2c309c6f18a7ef3aa4d2a
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/voice.c
al/voice.h
|
| 3ad4e1b9949d04ad90d053458b10fa4dfbfa088e |
03-Jun-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add support for dualmic fluence for voice communication
Add support for noise suppression preprocessing effect.
Add support for dualmic fluence for voice communication.
Reorganize input sound devices for more readability.
Remove invalid and unused legacy sound devices.
Bug: 14088317
Change-Id: Id9859de56780a8952d6e9acac84faa9b8ef1fdde
al/audio_hw.c
al/audio_hw.h
al/msm8974/platform.c
al/msm8974/platform.h
|
| f94bf89186551ee654817f654bc9b28ebe5e4b60 |
27-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 08dbcfc116e62f5b4007cade7311430ea235a09e
|
| 1524fa12ad87740d12293fdb655e44d1110c1370 |
26-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 08dbcfc116e62f5b4007cade7311430ea235a09e
|
| b422818f59dd0996e14abf2402d3dc21089cc2b8 |
25-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 08dbcfc116e62f5b4007cade7311430ea235a09e
|
| a0e69aaed8c25ba1e5d26fbe090b5ca7548ed5c6 |
24-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 08dbcfc116e62f5b4007cade7311430ea235a09e
|
| 3e20a01e983fcf32ae0c0ca49a4cc1ebf9ae4940 |
23-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 08dbcfc116e62f5b4007cade7311430ea235a09e
|
| 08dbcfc116e62f5b4007cade7311430ea235a09e |
21-Aug-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: avoid usage of extern functions and tables
If there is any difference in the signature of a function declared
as extern, it will not be reported by the compiler and may result
in unexpected results when executed.
All the API functions should be declared in a header file.
Change-Id: I89662e23da8118c3a9eac728b389498ed52e19c2
al/audio_hw.h
al/voice.c
al/voice.h
al/voice_extn/voice_extn.c
|
| 235c34827f7b3b8977fe76dea1fb8d818fd74312 |
22-Aug-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: fix incorrect acdb ids of tmus specific devices
Bug: 17187894
Change-Id: I40db5fa395dda984e5536cd4ddbed3ed23c3ccb2
al/msm8974/platform.c
al/msm8974/platform.h
|
| 7bb58bda373c4960dc2887f6be25df7676200d05 |
21-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to f1819247a21f755757a28ea313678faff73ef349
|
| f1819247a21f755757a28ea313678faff73ef349 |
06-Aug-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: fix voice call device routing issue
voice_device_set flag is to indicate voice call device routing
update from policymanager to HAL. It is set to true in voice_extn_start_call
and reset in update_calls() which causes mismatch in flag update during back
to back voice calls scenario. Update adev->voice_device_set flag in
voice_extn_stop_call instead of update_calls().
Bug: 17149385
Cherry-pick of CAF commit: e0085d06416a8fb6d0b4503900508c16fbc98c58
Change-Id: Ie07105671f254899890bdb4c0635c7dc1f55dbff
al/voice.c
al/voice.h
al/voice_extn/voice_extn.c
al/voice_extn/voice_extn.h
|
| db5b99c8159d56f6a85e5bf94576d3a252535d8b |
20-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 8b33679691b04f8b14cc512a012359f78fadc723
|
| a1b4cc5c28f6eaaaa4c53a31512bb12e34ef57aa |
19-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 8b33679691b04f8b14cc512a012359f78fadc723
|
| ecc3bf735859a883745fe31019d56c6e5739e257 |
18-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 8b33679691b04f8b14cc512a012359f78fadc723
|
| f39b26d80f5493d73c2fe047c46f0cef26e72ebb |
17-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 8b33679691b04f8b14cc512a012359f78fadc723
|
| 2fc22936fdbb749b00c65a86b0fbf863c9ecbe5c |
14-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 8b33679691b04f8b14cc512a012359f78fadc723
|
| e59b4619efe88a8d4abbced5385bd5acc9f6e51a |
13-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 8b33679691b04f8b14cc512a012359f78fadc723
|
| 8b33679691b04f8b14cc512a012359f78fadc723 |
12-Aug-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix initial audio loss in voice calls
Remove redundant CSD client API call which is taking more than
600msec due to socket realted failures. The csd_get_sample_rate
API is only meant for I2S based external modem.
Bug: 16550747
Change-Id: I696846188b0b3815efdb7bc0905ac68c5195899b
al/voice.c
|
| 5df2f6440736e0f1de2da253d5602b6319ebba89 |
12-Aug-2014 |
The Android Automerger <android-build@google.com> |
merge in lmp-release history after reset to 55a1293b422d181281cf0f7d37c6c15c5d921ef3
|
| 60503b74ba99bbe503c84251a96f07e96d5d41c2 |
11-Aug-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Revert "audio: add support for routing to/from voice TX/RX paths"
This reverts commit 8edfd66a7b1d033e65e5621d25ef3cbf1f40316e.
Bug: 16949514
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/voice.c
al/voice.h
|
| 55a1293b422d181281cf0f7d37c6c15c5d921ef3 |
11-Aug-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Revert "audio: add support for routing to/from voice TX/RX paths"
This reverts commit 8edfd66a7b1d033e65e5621d25ef3cbf1f40316e.
Bug: 16949514
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/voice.c
al/voice.h
|
| ae9a10c15cf960e9b2b1159ade6414f80689db5f |
09-Aug-2014 |
Iliyan Malchev <malchev@google.com> |
audio: rework dependency on libmdmdetect
libmdmdetect is a binary-only library, so we cannot change its API. Instead,
make the audio HAL depend on libdetectmodem, which exports a single function,
which returns the number of modems, or -1 in the event of error. If the library
is not present, or we fail to look up the symbol, or in the event of an error,
we assume that there is no modem.
extern int32_t count_modems(void);
b/16859052 aosp-shamu has dependencies on vendor/
Change-Id: I197dc5386b13fc2cce69fd273a47298517bd8b04
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/Android.mk
al/msm8974/platform.c
|
| f7c646835daca18cbfc0e060141e4deec8e65e3c |
07-Aug-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix razor checkbuild
Do not build post processing wrapper library on msm8960 platform.
Change-Id: I9e08f4e0a994a1605a81109c2d498ad58f2606f6
ost_proc/Android.mk
|
| b37e36b918e1ead5cc13faaed9b21ca1403a9653 |
07-Aug-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audio: add support for routing to/from voice TX/RX paths" into lmp-dev
|
| 8edfd66a7b1d033e65e5621d25ef3cbf1f40316e |
10-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio: add support for routing to/from voice TX/RX paths
Add support for routing voice calls to devices in other audio
HALs by allowing playback and captuer to/from AFE proxy.
Bug: 15520724.
Change-Id: Ia4a4428001ea06f7c7b213db861ec281ebd25174
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/voice.c
al/voice.h
|
| 97a1059da4d3aa8bfb0883d5f932f86b95876512 |
17-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: compilation & warning fixes for offload effects
Fix compilation errors & unused param warnings
in offload effect modules.
Change-Id: I58fa250c413e07702cf4a4f96aa85f985883b343
ost_proc/Android.mk
ost_proc/bass_boost.c
ost_proc/bundle.c
ost_proc/equalizer.c
ost_proc/reverb.c
ost_proc/virtualizer.c
|
| 41f86651e362abc62d9d03f5c612c986bf15298f |
17-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
post_proc: Offload effects combined patchset
* Support effects in DSP for tunnel mode playback
Add interface to support configuring equlaizer, bass boost,
virtualizer and reverb effects in DSP for tunnel mode of
playback.
(cherry-picked from CAF commit 3eedc004e9adf5967f393d65c22b7806d0c63e6c)
* post_proc: Enable effects param in DSP to start effect
-During switch to tunnel playback, offload effects-flags remain
disabled. This stops effects being applied on tunnel playback.
-Send effects enable params to DSP to start effects during switch.
(cherry-picked from CAF commit d02f2cd710334cc2be6a866da537a595256ae623)
* post_proc: disable adsp reverb when preset is 'None'
Audio HAL set corresponding preset id into adsp, but doesn't disabe
reverb explicitly when 'None' preset is chosen. However, preset=0
means custom preset in adsp, so adsp won't stop reverb processing.
(cherry-picked from CAF commit b27e25f062edfeebb6860078013c7b413b8c4301)
* post_proc: send ENABLE_FLAG directly to offload effect driver
Enable flag should be sent directly through offload effect driver to
adsp whenever bundle receives EFFECT_CMD_ENABLE. Otherwise enablement
will only take effect in adsp after subsequent parameters being set, and
that's not expected.
(cherry-picked from CAF commit d45948e2153c03b41f55577debd943408f9c8280)
* post_proc: reverb preset id is not mapped correctly
Array index is not correct and causes uninitialized value being used
when 'Plate' is chosen. Correct index to retrieve the right preset id.
(cherry-picked from CAF commit ca2685b2a0a695a7c4ac78883e750a92dcad0515)
* post_proc: disable Bassboost and Virtualizer for certain devices
WFD, Hdmi and usb audio are not intended to be applied with
SA+ bassboost and virtualizer, so add into invalid device list.
(cherry-picked from CAF ee2ff9621d25d83151ccb24a416f21533fdfbd31)
* post_proc: disable effect immediately when routed to invalid devices
Offload effect is still perceived when invalid output device is
connected. We should send disable command immediately once phone is
routed to unexpected device and forbid effect enablement during
temporary disabled state.
(cherry-picked from CAF commit 95d74c2232721631f3a04f6ccf35760b37b33fea)
* post_proc: Enable reverb in DSP to start effect
During switch to tunnel playback, reverb enable command is not sent to
DSP and causes reverb effect not applied continuously.
Send reverb enable params to DSP to start effects during switch.
(cherry-picked from CAF commit c37c260fa174453f4822261bcd531ddb2706ab2c)
* post_proc: include audio_effects.h from kernel exported headers
Currently, audio_effects.h header file is included with reference
to the absolute path. Instead, reference it from the kernel
exported headers.
(cherry-picked from CAF commit 090a2aa3f38298196ad9f47e3b6578535e1f7e10)
* hal: Add support for audio effects in DSP for tunnel mode playback
Add support to enable or disable audio post processing effects
in DSP for tunnel mode playback.
(cherry-picked from CAF commit 1d0891672175d431e8872dd7dff21e0ce507361a)
Change-Id: I7ead6da4c216fd87e8ca1884811c4e0155053f49
ndroid.mk
al/audio_hw.c
al/audio_hw.h
ost_proc/Android.mk
ost_proc/bass_boost.c
ost_proc/bass_boost.h
ost_proc/bundle.c
ost_proc/bundle.h
ost_proc/effect_api.c
ost_proc/effect_api.h
ost_proc/equalizer.c
ost_proc/equalizer.h
ost_proc/reverb.c
ost_proc/reverb.h
ost_proc/virtualizer.c
ost_proc/virtualizer.h
isualizer/offload_visualizer.c
|
| 09f2e0e6bf8accf1728ca89e780702d53f2c5b6d |
29-Jul-2014 |
Eric Laurent <elaurent@google.com> |
hal: Add support for AUDIO_DEVICE_OUT_LINE
Change-Id: I0d1163d7f7716f9a0366f2be245d50adad55b0cc
al/msm8974/platform.c
al/msm8974/platform.h
|
| 9d0d3f1537b5d157a2c31e0744303524b846446e |
25-Jul-2014 |
Eric Laurent <elaurent@google.com> |
hal: Add support for Hearing Aid Compatibility (HAC) mode
Change-Id: Id4e7eb98336fd3dd3569a31b61a3ccf529d081a3
al/msm8974/platform.c
al/msm8974/platform.h
al/voice.c
al/voice.h
|
| 91975a6766f1403c785e1834e2d3e99d51a13f4a |
28-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio HAL: add parameters to open stream functions
Pass device address (and audio source for inputs) to
open_output_stream() and open_input_stream() audio HAL functions.
Bug: 14815883.
Change-Id: I8019263435721f4f6ee3144897e1398946a843a7
al/audio_hw.c
egacy/alsa_sound/audio_hw_hal.cpp
|
| 7fa48c272e2aba35dcef005a19148b27826e0171 |
25-Jul-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "hal: use correct voice call use case id" into lmp-dev
|
| a237ecc29c7a2f22d01bc202a3daffe5856d73e4 |
25-Jul-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: use correct voice call use case id
The use case ID is different for different voice call types
such as CS call, IMS call etc.
Change-Id: I6e12730ebb81c2c7a5ea11a9b4e4881e5df1261b
al/audio_hw.c
|
| 8251ac85bf0c688d6043df3cb3f45d4082e62bc4 |
23-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix aac offload
Fix AAC format checks broken when
AAC sub types were added.
Change-Id: Ifef2ee24e0375eb92f758f034995f3d496baf2c6
al/audio_hw.c
|
| ff5d6518fa0e7e4d405fd11f71670551b7728969 |
19-Jul-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: Ignore ENOSYS error from voice_extn_set_parameter
Ignore this error as voice extn is unsupported by some targets.
Bug: 16373768
Change-Id: I6e3f51948aa21f679aca75a8986cfab09a72e320
al/voice.c
|
| 68e79ce63f768175d095d6c2b7c185a99ee4ddef |
15-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Add audio_input_flags_t to open_input_stream for low latency
Change-Id: Ibbb8350d1c5350e2f61499f8a081415cb1d4e19f
al/audio_hw.c
|
| 98c95622da1e906d32dde6b6651ed5b270b9b5f1 |
21-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: extend platform parser
Extend platform parser with support for additional sections.
Supported sections now include acdb ids, pcm device ids and backend names.
Change-Id: Idfbc8a8bb490606686436c107db5b0c7d636ccbe
al/msm8960/platform.c
al/msm8974/platform.c
al/platform_api.h
al/platform_info.c
|
| 5bc188456348ebdfc5d3c86414952503ec41bd44 |
17-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
hal: Add XML parser for platform info
change 1
Add XML parser which parses the platform_info.xml
on the device. That xml contains ACDB ID information
and is populated from the device project folder to
the /etc folder on the device. It is used to overwrite
hardcoded ACDB ID's in platform.c.
change 2
Move platform_parser to root hal directory. Rename
platform_parser to platform_info. Change name of
XML file read from platform_info.xml to
audio_platform_info.xml. The xml now only needs
information for ACDB ID's that you want overwritten.
Names in the XML now match sound device enums in
platform.c.
(cherry-picked from CAF commits
61764e3b8069b819c3da19a6bb38b37ad173bf50, 5588688cbdd065a3572afb032e48a265790dfea2)
Change-Id: Ie5978f609bbe9d60a64e20a0906d6bd7a8c48e1b
al/Android.mk
al/msm8960/platform.c
al/msm8974/platform.c
al/platform_api.h
al/platform_info.c
|
| fdf296a35ba3deb8490522c834037e5e977b05cf |
04-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio: deprecate audio_stream_frame_size()
Bug: 15000850.
Change-Id: I1bbe614c241befa24513a2b583594680e32fd954
al/audio_hw.c
|
| 0de8d1f80ff3cf452e9eb867f780b22bf8c54115 |
02-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio: fixed channel count determination from channel mask
Do not use popcount() to derive channel count from channel mask.
Bug: 15000850.
Change-Id: Idaf241be22f85040c6461834bad60ae1d9244f32
al/audio_hw.c
isualizer/offload_visualizer.c
|
| 20bcfa8b451941843e8eabb5308f1f04f07d347a |
25-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: Fix volume control for offload playback
Update the mixer control name used for volume control during
offload playback. Add an option to try the newer control
if the first (default) one isn't present.
Change-Id: I02dee627cc97bfb454b0e5dec2558f693593bb85
al/audio_hw.c
|
| b4d368e0fe6006657ebc4e1f9ba01a072c4ca2c7 |
25-Jun-2014 |
Eric Laurent <elaurent@google.com> |
hal: msm8974: fix VOICE_HANDSET selection for non-TMUS operators
Change-Id: I5033faeaa568b1c31c3ef4b7386cc5c1b3127f69
al/msm8974/platform.c
|
| eb772b7725bda68b203b8e4469202db8b96162cc |
20-Jun-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "hal: Fix no voice call audio issue on handset"
|
| 1de6e5aac3120408a003dc8b5f7fdd68c40f436d |
20-Jun-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix no voice call audio issue on handset
- Update the ACDB ids for voice handset sound device
- Fix incorrect condition check to add the backend name to
mixer path.
Change-Id: Ie70ea20191c5563456cf5733f66847e6100e71ff
al/msm8974/platform.c
al/msm8974/platform.h
|
| 24ca9ade8ff9550a1f30e28c6008e3b0832ab15f |
19-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: compilation fixes
Fix compilation errors introduced by the IMS change.
Change-Id: I24a3d11dd1f0619a96ab4dc4ef6afb3d324205ab
al/msm8960/platform.c
|
| 4b89e37ad290ef955abf8ac1d151728303311345 |
19-Jun-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Revert "Revert "hal: Add support for IMS calls""
This reverts commit a609e8ebdfeca875b6d35ccfb3fb8b87710f3499.
al/Android.mk
al/audio_extn/hfp.c
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.c
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/platform_api.h
al/voice.c
al/voice.h
al/voice_extn/voice_extn.c
al/voice_extn/voice_extn.h
|
| 299760a41231bd0f6d9991fb189977347365c72b |
01-Nov-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add support for external sound devices
Update audio HAL to detect and set the relevant mixer paths for
the sound devices such as speaker, earpiece etc. connected
externally through MI2S backend.
Change-Id: I1e9337a89cb022bce5271b6bde710f633ca2ac29
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
al/platform_api.h
|
| a609e8ebdfeca875b6d35ccfb3fb8b87710f3499 |
18-Jun-2014 |
Eric Laurent <elaurent@google.com> |
Revert "hal: Add support for IMS calls"
This reverts commit cedf1ac3c00e331b5f51b077f26c1367544ddd65.
Change-Id: I5f92f28c8b97265263a0bce5b38ff60d4655b68b
al/Android.mk
al/audio_extn/hfp.c
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.c
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/platform_api.h
al/voice.c
al/voice.h
al/voice_extn/voice_extn.c
al/voice_extn/voice_extn.h
|
| cedf1ac3c00e331b5f51b077f26c1367544ddd65 |
11-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
hal: Add support for IMS calls
Move all the voice call specific code to a new file.
Add voice extension files to support IMS calls.
Change-Id: I1b7235500c8e3c2285b726b351d996dc3e5ebdf6
al/Android.mk
al/audio_extn/hfp.c
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.c
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/platform_api.h
al/voice.c
al/voice.h
al/voice_extn/voice_extn.c
al/voice_extn/voice_extn.h
|
| cc9649b4f1e7d109ed005819971c3d9657a19711 |
11-Jun-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: fix unused param warnings
Change-Id: Ibd6b6cc1c76030997ec33a3a038ef664393b7aeb
al/audio_extn/hfp.c
al/audio_hw.c
isualizer/offload_visualizer.c
|
| 6e4a58a0f547f62c8f3b506c787af5969e1b5f33 |
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
am 88acf968: Revert "Disable fast capture by default for 8974"
* commit '88acf968e90f37c4e218c997b4476986f229aaf0':
Revert "Disable fast capture by default for 8974"
|
| 88acf968e90f37c4e218c997b4476986f229aaf0 |
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Revert "Disable fast capture by default for 8974"
This reverts commit 98fd45339a56253ea5fe117f2c91daaf586adeae.
Bug: 15591052
Change-Id: Ic07cd368bfb354f843d1dc3e4466beb877f3e452
al/msm8974/platform.h
|
| 517fcc613bfe1cb8e1d28a5c86acfb8fb5da22ab |
13-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
am 98fd4533: Disable fast capture by default for 8974
* commit '98fd45339a56253ea5fe117f2c91daaf586adeae':
Disable fast capture by default for 8974
|
| 98fd45339a56253ea5fe117f2c91daaf586adeae |
12-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Disable fast capture by default for 8974
Fast capture interacts poorly with AEC on 8974.
Frame count is changed from 240 to 480 (10 ms), which means
fast capture thread will not be started by default.
Fast capture thread is only used if frame count is strictly < 10 ms.
Fast capture feature can be re-enabled for specific app testing
by setting property audio_hal.in_period_size=240
in /data/local.prop
Bug: 15591052
Change-Id: I678bf233597c63dc414cab9f64b5ee494eb8701d
al/msm8974/platform.h
|
| ddd5f65fa348fd541391bab42fba1622db312613 |
12-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of 4f993391 to master
Change-Id: I5a1fd2107a613f2dd2c5c59c4c3e3e279f85585b
|
| 4f99339176510e3a9995fb84797efb6e77de3572 |
14-May-2014 |
Glenn Kasten <gkasten@google.com> |
Runtime configuration of period size
Bug: 14938247
Change-Id: I0b846e7b27155aba26c86d9232aa3fcd4aa9b8e1
al/audio_hw.c
al/msm8960/platform.h
al/msm8974/platform.h
|
| 8e6e98fc5af6d6f79bc71eb37df470380ae82fad |
06-Nov-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add support for HFP feature
- Add set_parameters to be set from hfp app to enable
and disable hfp session.
- Implement start and stop hfp session which takes
care of switching device and setting the session
Change-Id: Ie8697328ccbfee09d0d162f6fad01ddb552e4f83
al/Android.mk
al/audio_extn/audio_extn.h
al/audio_extn/hfp.c
al/audio_hw.c
al/audio_hw.h
al/msm8974/platform.c
al/msm8974/platform.h
|
| 83281a951af159ca00517f6132fab39727b293f5 |
20-May-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add voice call support for msm8084 target
Update platform file to load CSD libraries to enable voice calls.
Change-Id: Ie2aa194a2addc82a5121f92eb39c8d434cc42f26
al/Android.mk
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
al/msm8974/platform.h
al/platform_api.h
|
| aeac395b9fef46d9e80c3838d42f9b6c75d38618 |
06-May-2014 |
Vineeta Srivastava <vsrivastava@google.com> |
Audio HAL support for 8084 board
Change-Id: If5834e3a74543db91fd808d1c9c96b88d77117df
ndroid.mk
|
| 80e2d861c927f1eb7c26853dadd324ddf8442090 |
06-May-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Correct audio HAL library name
- Ensure that the platform name is reflected in the audio HAL
library name.
Change-Id: I942d7c5b428cd1e73cdf598c74f5203c881a060a
al/Android.mk
|
| d0660b7cf04d740a908c8ecf79ae63841c8b4498 |
30-Apr-2014 |
Eric Laurent <elaurent@google.com> |
am ac84e81d: am d47ff224: am 34fa769a: audio msm8974: new path for speaker phone with AEC
* commit 'ac84e81d06572586ba27db4b8b38e93bdf0408e6':
|
| ac84e81d06572586ba27db4b8b38e93bdf0408e6 |
30-Apr-2014 |
Eric Laurent <elaurent@google.com> |
am d47ff224: am 34fa769a: audio msm8974: new path for speaker phone with AEC
* commit 'd47ff224c7b24933c701acae8d5e4c98a1bc80af':
|
| 9f3065480c81bf01d3af65bfd3da09e1fb74b520 |
03-Apr-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
audio: support for wide-band speech audio over BT
Adding support for WB audio with BT devices
Change-Id: Ibaab69a8eb52f42f214c9c1f8f26ad3494728695
Bug: 13763881
al/audio_hw.c
al/audio_hw.h
al/msm8960/platform.c
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
|
| c38e452918fd27b410a40be44132db32090dfced |
14-Apr-2014 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
audio: use of audio_route_{apply/reset}_and_update_path APIs
enable/disable device and route using the new APIs to update
controls in the order listed in mixer paths file.
Change-Id: Ic0a8874e4a2080347cfa0c2e66af606a08a207a7
al/audio_hw.c
|
| e713730fd37f2a7f0cac765975e3b0c522fcf808 |
29-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Fix compile errors
Change-Id: Ic5d12a62bfc71b6f7dd3743ec8b65b6393771089
al/audio_hw.c
|
| 6fcba9c9bb5faa7ecf047447ddccc899dfbc43f3 |
18-Mar-2014 |
Andy Hung <hunga@google.com> |
Add extended bit-depth capability to audio hal
Extended bit-depth capability is disabled for now,
until mixer paths are fully updated.
Known issue - at least one device reports S24_LE
but operates as S32_LE, so when enabling extended precision,
PCM playback may be quiet.
Change-Id: Id76e6cb86d14be0bec69a6b4b01d780573eff6be
Signed-off-by: Andy Hung <hunga@google.com>
al/audio_hw.c
|
| e005e9e7480887ab04fdd8c77815ff775e99809a |
29-Mar-2014 |
Andy Hung <hunga@google.com> |
Merge "Scan and verify audio device parameters on open"
|
| 31aca91506be3beb30ae598cf22f6f940c2aac0b |
21-Mar-2014 |
Andy Hung <hunga@google.com> |
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>
al/audio_hw.c
al/audio_hw.h
|
| 03f09436bb527c26750659d702713ee16bbe75bf |
26-Mar-2014 |
Eric Laurent <elaurent@google.com> |
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
al/audio_hw.c
|
| fc3907054ff6dab21a07654e8bf0bd9a2b46fddc |
14-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 1e95cf3e: am 34fa769a: audio msm8974: new path for speaker phone with AEC
* commit '1e95cf3eb9c3433cd9f66899797b71114a6857a6':
audio msm8974: new path for speaker phone with AEC
|
| 1e95cf3eb9c3433cd9f66899797b71114a6857a6 |
13-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 34fa769a: audio msm8974: new path for speaker phone with AEC
* commit '34fa769ab5835cefb9a6b842590f5f690c0a52db':
audio msm8974: new path for speaker phone with AEC
|
| d47ff224c7b24933c701acae8d5e4c98a1bc80af |
12-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 34fa769a: audio msm8974: new path for speaker phone with AEC
* commit '34fa769ab5835cefb9a6b842590f5f690c0a52db':
audio msm8974: new path for speaker phone with AEC
|
| 34fa769ab5835cefb9a6b842590f5f690c0a52db |
07-Mar-2014 |
Eric Laurent <elaurent@google.com> |
audio msm8974: new path for speaker phone with AEC
Added new path for voice/video chat mic when AEC is on
in speakerphone mode to allow different gain settings from
speakerphone in telephony.
Bug: 13279002.
Change-Id: If8f76c243f2bf8b5defae35ecf871510cc6fe41d
al/msm8974/platform.c
|
| daa827d53a1ee952188b90c0dc8ea51bc87cb27d |
18-Dec-2013 |
Nick Kralevich <nnk@google.com> |
am 48177a7b: am 7bc597c3: Merge "Visualizer: do not use GNU old-style field designators"
* commit '48177a7b470da78543b66a618cfbf0c43a41fd74':
Visualizer: do not use GNU old-style field designators
|
| 48177a7b470da78543b66a618cfbf0c43a41fd74 |
18-Dec-2013 |
Nick Kralevich <nnk@google.com> |
am 7bc597c3: Merge "Visualizer: do not use GNU old-style field designators"
* commit '7bc597c30d8a5170a254d7a512e9eff11a46dc2b':
Visualizer: do not use GNU old-style field designators
|
| 7bc597c30d8a5170a254d7a512e9eff11a46dc2b |
18-Dec-2013 |
Nick Kralevich <nnk@google.com> |
Merge "Visualizer: do not use GNU old-style field designators"
|
| 19af91c3fe388314c0f6dec17af7c6b6e468ac68 |
18-Dec-2013 |
synergy dev <synergye@codefi.re> |
Visualizer: do not use GNU old-style field designators
Avoiding the use of GCC extensions improves code
portability
Change-Id: Iabe9fc84d135160367922e6d026e8608475fe8c8
isualizer/offload_visualizer.c
|
| defe30caceea40c6ef2bb864bef7d4fb6254f348 |
06-Dec-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 3f6b86a5: am e1d13d9e: Merge commit \'1ed341977485618189efe926120b291f0df3d33a\' into HEAD
* commit '3f6b86a5baf26c621e65bb25fb2c671764ddc479':
|
| 3f6b86a5baf26c621e65bb25fb2c671764ddc479 |
06-Dec-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am e1d13d9e: Merge commit \'1ed341977485618189efe926120b291f0df3d33a\' into HEAD
* commit 'e1d13d9ec7bc47e29253a4448c361d84b52ecd88':
|
| e1d13d9ec7bc47e29253a4448c361d84b52ecd88 |
05-Dec-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Merge commit '1ed341977485618189efe926120b291f0df3d33a' into HEAD
|
| c58bb27d9228dd1993523769b73001e935b78441 |
27-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audio: remove unused struct member
Removed unused member out_device from struct audio_device.
Change-Id: Id5ae06aed7d38aa85aefe09b9ff733bde1e52f57
al/audio_hw.c
al/audio_hw.h
|
| 4114c19f32caeea50d0ae34f6d19618f19c1dc4d |
23-Nov-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am c777e8df: Merge commit \'2c69188d9160b5e71ccffa6e42759c12fdb5965c\' into HEAD
* commit 'c777e8df408ea2bac6e283d0e9986725e8b41330':
|
| c777e8df408ea2bac6e283d0e9986725e8b41330 |
22-Nov-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Merge commit '2c69188d9160b5e71ccffa6e42759c12fdb5965c' into HEAD
|
| 1ed341977485618189efe926120b291f0df3d33a |
12-Nov-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 8bba9e95: hal: Fix the audio loss issue on codec back end
* commit '8bba9e957f56100d4e1464d576121273ffa434eb':
hal: Fix the audio loss issue on codec back end
|
| 8bba9e957f56100d4e1464d576121273ffa434eb |
12-Nov-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix the audio loss issue on codec back end
- Start music playback, plug in and plug out headset and press
'Next' button in the Music app immediately. Repeating these steps
result complete loss of audio on HW codec.
- When headset is pluged out and Next is pressed immediately, the
audio HAL triggers audio routing change from Headset to Speaker,
and closure of compress playback driver. The later is not lock
protected which result un protected access of back end information
in the ALSA framework. This leads to incorrect routing and hence
loss of audio. It is also observed that sometimes it could lead
to crash in kernel and phone reboots.
- Fix by ensuring that the kernel driver close is also lock protected
along with other routing events.
Bug: 11088400
Change-Id: I785effb09e5cef7ba20ee43e0ef91dc296d4e58a
al/audio_hw.c
|
| 023c5c93fbd2ae51b98df91a57e22cc657bf5694 |
30-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am a16217ce: Merge "audio: fix output flag test in open_output_stream" into klp-dev
* commit 'a16217ce0f296f69110ebd2cd773a2751c883c7d':
audio: fix output flag test in open_output_stream
|
| a16217ce0f296f69110ebd2cd773a2751c883c7d |
30-Oct-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audio: fix output flag test in open_output_stream" into klp-dev
|
| b1a7d2f9abe9f39d74fa52bcd8032542b17e018c |
24-Oct-2013 |
sangwon.jeon <sangwon.jeon@lge.com> |
am 866d5ff1: hal: Fix for Audio Route issue when sound path changes
* commit '866d5ff1463591e276b7f138c42466d9832c47d4':
hal: Fix for Audio Route issue when sound path changes
|
| 866d5ff1463591e276b7f138c42466d9832c47d4 |
17-Oct-2013 |
sangwon.jeon <sangwon.jeon@lge.com> |
hal: Fix for Audio Route issue when sound path changes
- if you make an outgoing call as mp3 playback is runnign with headset,
Right side of headset is not functional when keep changing sound path
from headset to speaker
- Fix the issue by separating loop related of disable/enable_snd_device
Bug: 11232052
Change-Id: Id02a6d7221c77cf4003d97749d75a062d8575d02
al/audio_hw.c
|
| a331bf737441ac8d59c2c741a7af06f4743608c2 |
20-Oct-2013 |
Nick Kralevich <nnk@google.com> |
am 02290055: Merge "voice_processing: do not use GNU old-style field designators"
* commit '022900556d6c3512cc841ea5e93e0597b6dab696':
voice_processing: do not use GNU old-style field designators
|
| 022900556d6c3512cc841ea5e93e0597b6dab696 |
20-Oct-2013 |
Nick Kralevich <nnk@google.com> |
Merge "voice_processing: do not use GNU old-style field designators"
|
| 30337434288e9abda1c6b59e88c17d76e84755e1 |
20-Oct-2013 |
synergydev <synergye@codefi.re> |
voice_processing: do not use GNU old-style field designators
Not using a GNU extension improves portability
Change-Id: I732753d2efa19efb48312664c7a2155f78ceee35
oice_processing/voice_processing.c
|
| 2c69188d9160b5e71ccffa6e42759c12fdb5965c |
17-Oct-2013 |
Ed Heyl <edheyl@google.com> |
merge in klp-release (no-op)
|
| c75149615da1f80fc4cf0988548270aa524d7984 |
13-Oct-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 07eeafd9390a85c5b9ad1642e89d3973615584cb |
06-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audio/hal: Configure HDMI channels based on output stream channels
- Current HAL configures HDMI channels based on sink capabilities,
even when the output content is stereo. DSP upmixes the content
if the HDMI backend is configured for 5.1 channels.
- This change ensures that HDMI backend is configured based on
output stream channels.
Bug: 7290997.
Change-Id: I42b2773b8f4ccc62203c13ff9ac6a6511df0705f
al/audio_hw.c
al/audio_hw.h
|
| 4e22546db0b489df9b805948de417b28a7f170e9 |
09-Oct-2013 |
Ed Heyl <edheyl@google.com> |
merge in KFS78N (no-op)
|
| a2b49c8236932f859de4ec1c163d0859fb9bd61a |
09-Oct-2013 |
Ed Heyl <edheyl@google.com> |
merge in KQS81M
|
| 5269691432805027b4ac9418e269b5fee98c9350 |
03-Oct-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| ed9c56ad071e5d0fb09ac7f310dc83a4553c1ad7 |
02-Oct-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix for Tx Mute issue when a new call accepted while in call
- While in a voice call, if an incoming call is accepted, Tx path
is muted on some specific networks.
- If there is an incoming call, Telephony sets the mode to RINGTONE,
and the condition check in audio HAL leads to voice call tear
down. When the call is accepted, the mode is set back to
IN_CALL and voice call path is re-enabled but the voice drivers
will be operating with default settings as there is no notification
from Modem side in this case. For modem, there is no change
in the call state i.e. just switched from caller 1 to caller2.
- Fix the issue by ensuring that the voice call is teared down
only if mode is set to NORMAL while call is progress.
Bug: 10733490
Change-Id: I2f205bb055807bb8d25f81e2907f78cd98bb77ad
al/audio_hw.c
|
| 7f24504d395380420bb4143bdec3e701aef428aa |
01-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audio: fix output flag test in open_output_stream
If flag AUDIO_OUTPUT_FLAG_DIRECT is set, we should only
try to open a multi channel HDMI PCM output if flag
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD is not set.
Bug: 8174410.
Change-Id: I63fe6cabca590e70c077dea13c86c6948700e606
al/audio_hw.c
|
| de517ba3e19fbf621b3d4d486d121bea25692e1a |
25-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| a6c11c11e2e7aee28b544674f1158b7b057c0c52 |
25-Sep-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
Level measurement in offloaded visualizer
Add Peak and RMS measurement capabilities in the "offloaded" version
of the visualizer effect.
Bug 8413913
Change-Id: I09a88f4cc791db6c68f0769dc23ced0d3aac955c
isualizer/offload_visualizer.c
|
| cfe66ae99db0b196788cc92a5e9b33223ee38efc |
22-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 7ff216f80f6e53235b4239c6fb7da9b0d5127738 |
12-Sep-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: Presentation time enhancements
1) Add API to query platform render latency. This API is only
valid for deep-buffer and low-latency streams.
2) Adjust frames rendered for deep-buffer and low-latency streams
with the platform render latency
3) Use tinycompress APIs to query presentation time in case of
offload streams.
Bug: 10551158
Change-Id: If94e0994bfc0b757f29aa4b48be6fc63dc17bca0
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
al/platform_api.h
|
| 949a089132db31b652d937a949f7bd425f2e42f9 |
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
implement get_presentation_position() for offloaded outputs
Bug: 9587132.
Change-Id: Idf40259b59552c29671830f30ccca3bef6ef0edd
al/audio_hw.c
|
| ed0b99ceb452a371e9f7707e7beef5002e9418c9 |
14-Sep-2013 |
sangwon.jeon <sangwon.jeon@lge.com> |
audio : add new TMUS MCC and MNC list
Add new TMUS MCC and MNC to use TMUS audio tuning data
Bug: 10751789
Change-Id: I1082cda69f50e722164cb5066eeef0273943a93c
al/msm8974/platform.c
|
| 9f2265da42169d740ec65b67e231d7435f544b20 |
19-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-factoryrom-release history after reset to klp-dev
|
| 51b475d4ed8b2945bb12dd37ca942a48c8897854 |
19-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 8d8f4d50ebc4caa2c886fcb2ab8cf4729ec84878 |
14-Sep-2013 |
sangwon.jeon <sangwon.jeon@lge.com> |
audio : add new TMUS MCC and MNC list
Add new TMUS MCC and MNC to use TMUS audio tuning data
Bug: 10751789
Change-Id: I1082cda69f50e722164cb5066eeef0273943a93c
al/msm8974/platform.c
|
| 957b4382a1fdd8b7bf3cb32db32041887863fa37 |
18-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "add offloaded audio visualizer" into klp-dev
|
| 63bbf66aa650ea75538b8668a2b1fe5d3f8b8878 |
18-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| b891db5473ab23a0cbc52d566a97c3d3529f8ddd |
14-Sep-2013 |
sangwon.jeon <sangwon.jeon@lge.com> |
audio : add new TMUS MCC and MNC list
Add new TMUS MCC and MNC to use TMUS audio tuning data
Bug: 10751789
Change-Id: I1082cda69f50e722164cb5066eeef0273943a93c
al/msm8974/platform.c
|
| c4aef75c2c5a0d49cac941d22235ac0b9e435ca0 |
13-Sep-2013 |
Eric Laurent <elaurent@google.com> |
add offloaded audio visualizer
Add library for visualizer effect used when
audio decompression is offloaded to QCOM audio DSP.
The implementation reads PCM back from the proxy port
in the audio DSP.
The audio HAL dynamically loads the effect library if present
and indicates offloaded output activity.
The PCM capture is only active when an offloaded output
is active and at least one effect is enabled on this output.
Bug: 8174410.
Change-Id: Ic78de932f9116e246494f9171c1cc7c3e35a0ea1
ndroid.mk
al/audio_hw.c
al/audio_hw.h
isualizer/Android.mk
isualizer/MODULE_LICENSE_APACHE2
isualizer/NOTICE
isualizer/offload_visualizer.c
|
| 02111e922558f75006e6340382367c5b987bf1a3 |
16-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-factoryrom-release history after reset to klp-release
|
| c8e25eee84af304e57c62718a29afeae604320a6 |
16-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 352f27bea3ea82b64234485de7a0f87a1991ab06 |
26-Jul-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: compress offload enhancements
1) Parse and set encoder metadata (delay, padding)
2) Call compress_next_track to allow partial drain
3) Do not flush output on returning succesfully from partial drain
Change-Id: I0fa1a2c968a5590dff9b6c58bd52bb111dcf3e9b
Bug: 8174410
al/audio_hw.c
al/audio_hw.h
|
| 37f761768f624be68495bf48cc5192695666b8f0 |
13-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audio: enable AAC offload" into klp-dev
|
| 86e1713c4eb1378823c1f544f0fea2339c08f393 |
13-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audio: enable AAC offload
Bug: 8174410.
Change-Id: I343a35b90a2b21ea7954856ac7f73b9f5a07e0f2
al/audio_hw.c
|
| 4d20e58c4389ae81ba9b3a1fe7357c281aad1b9e |
11-Sep-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: set cached volume before starting voice call
Apply cached voice volume before starting voice call.
Change-Id: I6a86d5f104b6a19c13bffbdeafa7a7f325d5ad3b
Bug: 10516515
al/audio_hw.c
|
| 5191a856311c5bd5a1b48810032ccdbc35a7fdc7 |
11-Sep-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audio: set cached volume before starting voice call
Apply cached voice volume before starting voice call.
Change-Id: I6a86d5f104b6a19c13bffbdeafa7a7f325d5ad3b
Bug: 10516515
al/audio_hw.c
|
| 96454108bcb4606e734277f30f61a9c7a9bd68fe |
12-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 2ccd7babcde54a8073945dec69660cb60e66a931 |
10-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Implement HAL API get_presentation_position
This does _not_ address bug 10551158 (to include DSP buffering)
Change-Id: Ifbc5ca21c46eced3f93a891200c763a062625dd9
al/audio_hw.c
al/audio_hw.h
|
| 10e79c660583f74269cd197bc5764be80964a3fc |
09-Sep-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 6e89524079e9c3a1037c57a99854820d88c9380b |
06-Sep-2013 |
Eric Laurent <elaurent@google.com> |
hal: force stop after drain.
Force playback state to IDLE and send compress_stop()
when drain completes to force reset driver and DSP pointers.
This ensures that even if last write was partial, next write
will be on a 32 byte boundary.
Also do not wait for write completion if compress_write()
returns an error.
Bug 8174410.
Change-Id: If144981c6396b24515d45b32a75ab61872a35ea2
al/audio_hw.c
|
| d604d86dd53680fb3106f4d5eb99afcec6797b41 |
29-Aug-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 4e02e5575f2eb440632a60fb8bed0a44ddae83af |
18-Jul-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add support for Tunnel mode audio playback
- Initial implementation to support audio playback in tunnel mode.
Change-Id: I4ffa660bd9beb855fdfe6a7572d8f6b7eade7bd9
al/Android.mk
al/audio_hw.c
al/audio_hw.h
al/msm8974/platform.c
|
| 7951576278e012351381ff75eb8d43b66796b798 |
22-Aug-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| be3335ab71f097449546112d9ae77f43aa967f47 |
20-Aug-2013 |
Ajay Dudani <adudani@codeaurora.org> |
hal: Add support for msm8226 platform
msm8226 uses common code base as msm8974.
Change-Id: Ibea589637c7a9464ba274156ab43820c82124bd3
al/Android.mk
|
| daf9f09e61e7160508c678fa9e79e526f3a7f26d |
19-Aug-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| 1e5f35378d14987df8f8641c2da6438999763536 |
09-Aug-2013 |
Devin Kim <dojip.kim@lge.com> |
audio: fix logging
After separate platform specific code, device table
is set to each platform specific code.
But, this debug message didn't changed correctly.
Change-Id: I4607907e182cf57525847aba1e189c6539c9a3be
Signed-off-by: sangwoo <sangwoo2.park@lge.com>
Signed-off-by: Devin Kim <dojip.kim@lge.com>
al/audio_hw.c
|
| 2d5c4b75ef979fecd0c662180808e1854d65d9cc |
13-Aug-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to klp-dev
|
| bc677243eb84cad20cdedd9909d44f71308620c3 |
08-Aug-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: change debug message level for debugging
For debugging, path changing debug message and
out_set_paramters debug message need.
Change-Id: Ibd203aea957f84381c631184a4c9303ec069dad5
al/audio_hw.c
|
| 83841b9f2bcb53bd05dc48ff3304543db1be797e |
19-Jul-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
hal: Fix EDID information
- Get EDID audio block info from a mixer control instead of a file.
- Fix a bug where the current sad is not updated in the for loop.
Bug: 9430906
Change-Id: I750e307ce1064eeb98d09ea8534a375252630841
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
al/platform_api.h
|
| 47cd4cbdb19543338d5c887e3d7bcd2513c5c3ad |
19-Jul-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
hal: Fix EDID information
- Get EDID audio block info from a mixer control instead of a file.
- Fix a bug where the current sad is not updated in the for loop.
Bug: 9430906
Change-Id: I750e307ce1064eeb98d09ea8534a375252630841
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
al/platform_api.h
|
| 787b92a13cd583250324636f577fefd8240d4993 |
30-Jul-2013 |
The Android Automerger <android-build@google.com> |
merge in klp-release history after reset to master
|
| 72cd4efa0628af6f6f0941a901a5715ec8ce0802 |
26-Jul-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: change mute/volume controls for using soft control api
Volume and Mute controls are changed to use specific session and
soft control api. So, HAL also changed to use new api.
bug: 10023401
Change-Id: I686d8495293242097817d870477f59ed154a6b31
Signed-off-by: hyunsub.na <hyunsub.na@lge.com>
Signed-off-by: Sungmin Choi <sungmin.choi@lge.com>
Signed-off-by: sangwoo <sangwoo2.park@lge.com>
al/msm8974/platform.c
al/msm8974/platform.h
|
| 0fecf5d5777a0fc8754f1f275d052bd9bd80ec81 |
29-Jul-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "audio: Change TMUS Voice call acdb id"
|
| 53b2cf0c72aa18a5848919e2309731af652e84f9 |
26-Jul-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: change mute/volume controls for using soft control api
Volume and Mute controls are changed to use specific session and
soft control api. So, HAL also changed to use new api.
bug: 10023401
Change-Id: I686d8495293242097817d870477f59ed154a6b31
Signed-off-by: hyunsub.na <hyunsub.na@lge.com>
Signed-off-by: Sungmin Choi <sungmin.choi@lge.com>
Signed-off-by: sangwoo <sangwoo2.park@lge.com>
al/msm8974/platform.c
al/msm8974/platform.h
|
| c69476f6abbd7f82c2df8cf51ba3f4fed6eb75fe |
27-Jul-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: Change TMUS Voice call acdb id
The acdb id is changed in Handset_cal.acdb file.
So, it needs to match with acdb file
Change-Id: Idb044426ad21fd5bc0a14a75c0e1ea4df8bbe6bb
al/msm8974/platform.c
|
| 994a693158202488516c48c22534ae2035b5c8fa |
17-Jul-2013 |
Eric Laurent <elaurent@google.com> |
audio: reduce audio HAL log spam.
Change-Id: I73a7ee40a32ccd4e6a85e49d08a6610351fedab7
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
|
| 1b9f4b3708d1ed1204bdb1dec370ad2e9db7a779 |
22-Jun-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio HAL: Add retry to get mixer in adev_open
If the sound card is not created in kernel before audio hal initialized
by audio flinger, the mixer open would be failed.
This is timing issue. So retry routine is need.
Change-Id: Icff3cd53763bfc483725849874fe27ff4de28890
al/audio_hw.c
al/msm8960/platform.c
al/msm8974/platform.c
|
| 33d330678f797b8796f72114bad42957b9ca204f |
11-Jun-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
audio/hal: Set playback buffer size to integral multiple of msec
- Set the deep-buffer and low-latency output buffer sizes to
integral multiple of msec. This reduces the variations in
the writes.
- Compute the input buffer size based on sample rate. Use 20msec
buffers in capture path.
Bug: 9283911, 9106434
Change-Id: Icbcb653f7f0fd3293dd4b514a54ac91d8311b308
al/audio_hw.c
al/msm8960/platform.h
al/msm8974/platform.h
|
| b23d5286490ad2dc0edf919d52428fa02dc2b2dc |
15-May-2013 |
Eric Laurent <elaurent@google.com> |
audio HAL: separate platform specific code
Separate platform specific code from generic audio HAL code.
Platform specific code is:
- platform initialization
- pcm device selection
- pcm stream configuration
- sound device selection
- acdb ID selection
- HDMI configuration
Change-Id: Iaf327943fa8674aad0c22a71e7cbf4288a138c7d
Conflicts:
hal/audio_hw.c
hal/audio_hw.h
al/Android.mk
al/audio_hw.c
al/audio_hw.h
al/edid.c
al/msm8960/platform.c
al/msm8960/platform.h
al/msm8974/platform.c
al/msm8974/platform.h
al/platform_api.h
|
| 7cf0f012bca68a05a97b2cb9a2dd1db1339537a9 |
10-Jun-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "audio: Change to perform voice calibration before update audio route"
|
| 170731f27d4be8575249d95c116560f82f5661a3 |
08-Jun-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: Change to perform voice calibration before update audio route
If audio route is set before voice calibration, the TX and RX device tap
is swapped in calibration tool. So, this sequence is changed to perform
voice calibration before update audio route.
Bug: 9363506
Change-Id: Ic7687a92f0d3c3faea0cf48f9d56e5877ba6255b
al/audio_hw.c
|
| 2b7f8554645121c0d0cd8066f17f831e12e58963 |
10-Jun-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 5b5e0aff: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit '5b5e0affac0abf011bbd0b4ea5bd5e6cebb5bc00':
hal: Fix Hangout and Voice call concurrency issue
|
| 5b5e0affac0abf011bbd0b4ea5bd5e6cebb5bc00 |
10-Jun-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: Ib89e9670cf7e277b9b5d0563cf3f9e72e9e59834
|
| ca414a0b591abf4178033481098233f2210cb4cb |
08-Jun-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "audio: msm8974: change the ACDB ID for voice call"
|
| ea6ef9d15574f1406940fae42914ffc49c85d659 |
03-Jun-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: msm8974: change the ACDB ID for voice call
msm8974 uses fluence version 5 in voice call.
So, ACDB ID for handset and speaker voice call should be changed to
41 and 43. It depends on ACDB mapping table.
Change-Id: I13990299230daa4cd0ade037d3c224f50013d9e9
al/audio_hw.c
|
| 518da37222759a15a9c6c7a564c6c2de4920aec2 |
05-Jun-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix Hangout and Voice call concurrency issue
- While a hangout is in progress, if the user accepts an incoming
voice call, there is no audio heard on the far-end.
- The tx path for hangout is using voice-speaker-mic device and
the voice call tx path tries to enable the voice-dmic-ef device.
But both these device use same physical mic and same codec backend.
When the Hangout is terminated it tries to disable the speaker-mic
which disables mixer controls that are common with voice-dmic-ef.
- Fix the issue by making sure all the capture usecases are routed
to same input sound device always.
Bug: 9228503
Change-Id: Iaf1b0e61d10437e2d9deeeffd7ca67770b6e00f6
al/audio_hw.c
|
| 935a0f9de560fd4b59ba011e97f3c6dffc9c686a |
06-Jun-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 9aaf2a65: (-s ours) Reconcile with jb-mr2-zeroday-release - do not merge
* commit '9aaf2a65e3cbc6866cfd50fe5f3341063c71f36f':
|
| 9aaf2a65e3cbc6866cfd50fe5f3341063c71f36f |
06-Jun-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-zeroday-release - do not merge
Change-Id: I2ff8807ad12569b93bbed8cf242c8fd1758d7bcb
|
| 69cde0e6efbbc98d412bc54c260e8241ed98248b |
06-Jun-2013 |
The Android Automerger <android-build@google.com> |
merge in jb-mr2-zeroday-release history after reset to jb-mr2-dev
|
| 144281282b1d8b8cb9e6c9b50ad6ccd66da42838 |
06-Jun-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 3d746369: am c4ba743c: hal: Fix Hangout and Voice call concurrency issue
* commit '3d74636915254e13957a59fc307dbf8fc396e7e9':
hal: Fix Hangout and Voice call concurrency issue
|
| 3d74636915254e13957a59fc307dbf8fc396e7e9 |
06-Jun-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am c4ba743c: hal: Fix Hangout and Voice call concurrency issue
* commit 'c4ba743cc4e48b9feabccf03959642d63cf7076e':
hal: Fix Hangout and Voice call concurrency issue
|
| c4ba743cc4e48b9feabccf03959642d63cf7076e |
05-Jun-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Fix Hangout and Voice call concurrency issue
- While a hangout is in progress, if the user accepts an incoming
voice call, there is no audio heard on the far-end.
- The tx path for hangout is using voice-speaker-mic device and
the voice call tx path tries to enable the voice-dmic-ef device.
But both these device use same physical mic and same codec backend.
When the Hangout is terminated it tries to disable the speaker-mic
which disables mixer controls that are common with voice-dmic-ef.
- Fix the issue by making sure all the capture usecases are routed
to same input sound device always.
Bug: 9228503
Change-Id: Iaf1b0e61d10437e2d9deeeffd7ca67770b6e00f6
al/audio_hw.c
|
| 159cd3f13ddd7e1dea418a6c03f3772ac32b2450 |
03-Jun-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 9c2d459e: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit '9c2d459e952b65029bbb7d280a1fcc971896cf22':
|
| 9c2d459e952b65029bbb7d280a1fcc971896cf22 |
03-Jun-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I3cb51e82df9bd0f75f5a00820a6f71ddb23734bf
|
| ed152bbbaa43332db58b7a78d445a2b326597f26 |
03-Jun-2013 |
The Android Automerger <android-build@google.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| c000485d469e952edfed0c953e75240efd6928ca |
02-Jun-2013 |
The Android Automerger <android-build@google.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| b3714b1830ce154f12a22c0d0683912140667ba7 |
01-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am d90c5495: am bdfd2924: add device specific voice processing descriptors
* commit 'd90c54958fe8239b2d4302141d3e8e81d18ef7ec':
add device specific voice processing descriptors
|
| d90c54958fe8239b2d4302141d3e8e81d18ef7ec |
01-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am bdfd2924: add device specific voice processing descriptors
* commit 'bdfd2924d535d8d3de1bd63c407568134641ef18':
add device specific voice processing descriptors
|
| bdfd2924d535d8d3de1bd63c407568134641ef18 |
29-May-2013 |
Eric Laurent <elaurent@google.com> |
add device specific voice processing descriptors
Add the possibility to define device specific effect descriptors
in libqcomvoiceprocessingdescriptors.so. This will allow
exposing different implementation UUISs according to device tuning
so that applications can distinguish one from the other.
Bug: 9126576.
Change-Id: I8e6ca00cbc6386498a5df99b514b1c7b7b1fd82c
oice_processing/voice_processing.c
|
| e402fdd61ea5f231a323de3a14a0dca3e31bc192 |
31-May-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
resolved conflicts for merge of 5b875cdd to master
Change-Id: Ic61cb0062d587851d2f0b4963a45cd3e4a73afa8
|
| 5b875cdd1f69f878e795f36fa213571bbf3d0de5 |
30-May-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
am c56336bf: QC audio HAL handles device rotation
* commit 'c56336bfad4661796b749fc4db7de3a1e6aba06f':
QC audio HAL handles device rotation
|
| c56336bfad4661796b749fc4db7de3a1e6aba06f |
25-May-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
QC audio HAL handles device rotation
Use the "speaker-reverse" configuration when the device rotation
requires it. Device rotation is received through a parameter to
parse.
Bug 9095903
Change-Id: Ie24a625a18e1fc1093f6f564ba0ff0f5cbb5cce0
al/audio_hw.c
al/audio_hw.h
|
| 75ebaa052d5cad1b3b46c7beb40c9022b5806f04 |
22-May-2013 |
Sungmin Choi <sungmin.choi@lge.com> |
audio: Remove specific library path
Some devices use /system/lib, others do /vendor/lib.
Both are located in LD_LIBRARY_PATH whicn is defined in init.rc.
So it doesn't need to use specific library path.
Change-Id: I97e3abf3f84e8fe2d280d0714a32dd6861b5b937
al/audio_hw.c
|
| fbf5d518e04ce4145db1c90965223cb64a40435e |
20-May-2013 |
sangwoo <sangwoo2.park@lge.com> |
audio: Fix to send voice calibration during voice call
To voice cal, use the acdb_send_voice_cal interface.
Bug: 8966887
Change-Id: I95361d90bd74e4b8096bd2eb109e3eba05951fd9
al/audio_hw.c
|
| 6961109b2f446e92d969fe093b940f71c01a8b7a |
16-May-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 20d28cb2: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit '20d28cb2dc069a17af843ce666420fc3702973e0':
|
| 20d28cb2dc069a17af843ce666420fc3702973e0 |
16-May-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I46e32f210b004047f9b3714595cbdea459eafedd
|
| 1958f10d1c31eaa3232cdf636056b9d129efba20 |
15-May-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 9ac4be47317acb19a4077de3723118d879374018 |
15-May-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 2c921376: am f70ffb40: hal: Add support for pre-processing effects
* commit '2c9213763f7b7423c26dacbc59c61635bf01d4f4':
hal: Add support for pre-processing effects
|
| 9ace006143041267662d8e3082c2ee98f334b4dd |
15-May-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 7c57a3a8: am 59d296d8: hal: select speaker mic for voice communication
* commit '7c57a3a8ab8d526c074bcc373e8cba2fa06b4f23':
hal: select speaker mic for voice communication
|
| 2c9213763f7b7423c26dacbc59c61635bf01d4f4 |
15-May-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am f70ffb40: hal: Add support for pre-processing effects
* commit 'f70ffb40ca0c4e8cce15c77fd9edff7f2b6980de':
hal: Add support for pre-processing effects
|
| 7c57a3a8ab8d526c074bcc373e8cba2fa06b4f23 |
15-May-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 59d296d8: hal: select speaker mic for voice communication
* commit '59d296d800f7bacd9c2b07a84b2db55489be9a09':
hal: select speaker mic for voice communication
|
| f70ffb40ca0c4e8cce15c77fd9edff7f2b6980de |
17-Apr-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add support for pre-processing effects
- Add support for AEC audio effect for voice communication.
Bug: 7241490
Bug: 8325112
Change-Id: Ic33d2f1f7be86484f748627d9afabbe10c369c21
Signed-off-by: Eric Laurent <elaurent@google.com>
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/Android.mk
al/audio_hw.c
al/audio_hw.h
|
| 59d296d800f7bacd9c2b07a84b2db55489be9a09 |
02-May-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: select speaker mic for voice communication
- Current implementation selects handset mic for voice
communication in speaker mode
- Check if the primary output (rx path) is active on
speaker and select speaker mic during input device
selection
- If the tx path is started first, ensure that tx device
is updated while starting the rx path.
Bug: 8325112
Change-Id: I1c556c0c9c92e599c8a1f68575b26ecdad155e7e
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
|
| 87f6ee0a0074149c8060a4c1672d2907d1a50a1a |
07-May-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
hal: Add support for voice call volume and mute
- Set mixer controls to support mute/unmute and volume
for voice call for non-fusion targets.
- Non-fusion targets do not need CSD library for voice
call functionality.
Change-Id: Ie232d89691c3ca91a9a366ece020588d82fba679
al/audio_hw.c
al/audio_hw.h
|
| b3baf83700d625926443be4c4fdfde820067dee2 |
03-May-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "audio: msm8974: Fix device number for voice call usecase."
|
| d2f3c3b46baa2caadbbecdeb3ff65472102ae6ba |
03-May-2013 |
Shashank Mittal <mittals@codeaurora.org> |
audio: msm8974: Fix device number for voice call usecase.
Change-Id: I55e69cb64cc061fabed08fcacb5145dcf4530c7b
al/audio_hw.c
|
| 91f61baacf0e176acd1d7b3a0a04d9335cc1da60 |
02-May-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| ec7ce2d5ce0c218688921c974bb54e89dcdc57c4 |
02-May-2013 |
Eric Laurent <elaurent@google.com> |
am af242070: am 73fb11d9: audio: add voice processing effect wrapper
* commit 'af242070effe2929eb8bc39fe4721729368523a2':
audio: add voice processing effect wrapper
|
| af242070effe2929eb8bc39fe4721729368523a2 |
02-May-2013 |
Eric Laurent <elaurent@google.com> |
am 73fb11d9: audio: add voice processing effect wrapper
* commit '73fb11d93274bc1c3675b24e910a4cb87571ffd0':
audio: add voice processing effect wrapper
|
| 73fb11d93274bc1c3675b24e910a4cb87571ffd0 |
09-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audio: add voice processing effect wrapper
Added wrapper library to expose Fluence AEC and NS to effect
framework.
Bug 7241490
Change-Id: I9cec4a5e7dde210e0eb9f4dd3de341b9c83b340d
ndroid.mk
oice_processing/Android.mk
oice_processing/voice_processing.c
|
| 28349072f3c6b0e1696b9636d23f03f2b4d70c56 |
15-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am a9d5c825: Reconcile with jb-mr2-release - do not merge
* commit 'a9d5c8259176af09e0d27d8941a5832069f862fc':
|
| a9d5c8259176af09e0d27d8941a5832069f862fc |
15-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I480c3dcdb8b1541a9d463ed11a6a90f34535aacb
|
| 8282fd33e6bd7ef24854018a2e52090998bc92b4 |
15-Apr-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| e8a5711ba90424af83d18e0666ac53595f03abb4 |
12-Apr-2013 |
Sungmin Choi <sungmin.choi@lge.com> |
am 59376cff: am 6336b0d0: audio: enable recording
* commit '59376cffcc4e85ce189c9d8d64f241704d38e2b9':
audio: enable recording
|
| 59376cffcc4e85ce189c9d8d64f241704d38e2b9 |
12-Apr-2013 |
Sungmin Choi <sungmin.choi@lge.com> |
am 6336b0d0: audio: enable recording
* commit '6336b0d02f1d8f136e2ab35d6222263ff54334bd':
audio: enable recording
|
| 6336b0d02f1d8f136e2ab35d6222263ff54334bd |
11-Apr-2013 |
Sungmin Choi <sungmin.choi@lge.com> |
audio: enable recording
Update audio-record config, align with kernel hw parameter
Change-Id: I428d98f5d28edc26de335be1ac4667dcc4ffa4ea
Signed-off-by: Cong Zhou <cong.zhou@lge.com>
Signed-off-by: Sungmin Choi <sungmin.choi@lge.com>
al/audio_hw.h
|
| 85e19fa4fa66b48d3b406015be4c9ea0767e3b33 |
11-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am e78d87df: Reconcile with jb-mr2-release - do not merge
* commit 'e78d87df3253516a93c1042beabe0edfe782a170':
|
| e78d87df3253516a93c1042beabe0edfe782a170 |
11-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I30773b20dfe3913483382ba0c9d0c75749e398bc
|
| 36a6eaeb2025dedca55cb70357e8196b3647e9c7 |
11-Apr-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 778608788d4c3fe5f87d08c772c05d5afd794fca |
10-Apr-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
am bbf251cd: am e06e66d9: Merge "audio: enable audio using tinyalsa on MSM8974" into jb-mr2-dev
* commit 'bbf251cded050b28824b1e8f9467ee96fbf1e3e6':
audio: enable audio using tinyalsa on MSM8974
|
| bbf251cded050b28824b1e8f9467ee96fbf1e3e6 |
10-Apr-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
am e06e66d9: Merge "audio: enable audio using tinyalsa on MSM8974" into jb-mr2-dev
* commit 'e06e66d912219ae6e83bddb8559f3264ac51f817':
audio: enable audio using tinyalsa on MSM8974
|
| e06e66d912219ae6e83bddb8559f3264ac51f817 |
10-Apr-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
Merge "audio: enable audio using tinyalsa on MSM8974" into jb-mr2-dev
|
| f2e1154f83ff0e02405544268b62fbad2ae25125 |
10-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 42683cc2: Reconcile with jb-mr2-release - do not merge
* commit '42683cc2cea897d4696b233c48bfd7d49f5dab46':
|
| 42683cc2cea897d4696b233c48bfd7d49f5dab46 |
10-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: Ide8e99ec794b6436156f491beed44ffc3440a7c3
|
| 5195a4b2f95ad704d2408b7cdcbb537c362a748b |
04-Apr-2013 |
Sungmin Choi <sungmin.choi@lge.com> |
audio: enable audio using tinyalsa on MSM8974
Change-Id: I003dedd9f29de5aec1b620442aa8b3c3c7b7a816
al/Android.mk
al/audio_hw.c
al/audio_hw.h
|
| 4d30b01e1d2048c120f7e1efbf7cfda3f623cc09 |
10-Apr-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| f389189dff0cfb4c449991a1385c23695894cc2e |
09-Apr-2013 |
Rom Lemarchand <romlem@google.com> |
am 045c5068: am 00681076: Add support for msm8974 and msm8226 targets
* commit '045c5068a08a2d36fafc369e2e0fc5ddc9df4957':
Add support for msm8974 and msm8226 targets
|
| 045c5068a08a2d36fafc369e2e0fc5ddc9df4957 |
09-Apr-2013 |
Rom Lemarchand <romlem@google.com> |
am 00681076: Add support for msm8974 and msm8226 targets
* commit '00681076451d323128fb46b6d891b525c5f9bb50':
Add support for msm8974 and msm8226 targets
|
| 00681076451d323128fb46b6d891b525c5f9bb50 |
09-Apr-2013 |
Rom Lemarchand <romlem@google.com> |
Add support for msm8974 and msm8226 targets
Adding support for msm8974 and msm8226 targets to make Qualcomm patch
integration easier.
Change-Id: Id3196c8c314ee1174580b22b11fc4068b0421504
ndroid.mk
|
| 1a903cbdde8e2a0103a1c8ba2598d22f98ef9f5a |
08-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 1cd0e202: Reconcile with jb-mr2-release - do not merge
* commit '1cd0e20269fb5faf6cd85e338b7370648a95f8ef':
|
| 1cd0e20269fb5faf6cd85e338b7370648a95f8ef |
08-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: Ic31e144dc752e8c259b9589e9f0804b2bf8d05b2
|
| 550d5f939e5f22f207cf23230d73ff09ecd32b9c |
08-Apr-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 2acc769200b56ca144dbff6f778c7369078b9a6f |
05-Apr-2013 |
Dmitry Shmidt <dimitrysh@google.com> |
am 4db6aef9: am 795d21f1: qcom: audio: Add 8x26 and 8x74 architecture
* commit '4db6aef993e36fb0f6af83e587380c1179a355cb':
qcom: audio: Add 8x26 and 8x74 architecture
|
| 4db6aef993e36fb0f6af83e587380c1179a355cb |
05-Apr-2013 |
Dmitry Shmidt <dimitrysh@google.com> |
am 795d21f1: qcom: audio: Add 8x26 and 8x74 architecture
* commit '795d21f14da4538ff4727b2df3c5cf4f70563af9':
qcom: audio: Add 8x26 and 8x74 architecture
|
| 795d21f14da4538ff4727b2df3c5cf4f70563af9 |
05-Apr-2013 |
Dmitry Shmidt <dimitrysh@google.com> |
qcom: audio: Add 8x26 and 8x74 architecture
Change-Id: Ib60be6a0377effe5a198100842c5bd916d77c0ca
Signed-off-by: Dmitry Shmidt <dimitrysh@google.com>
ndroid.mk
|
| 4ecbfa6672b64f6f11ce09103950d19367d32239 |
04-Apr-2013 |
Iliyan Malchev <malchev@google.com> |
am 01cbedaa: am 0819f6a5: Merge "qcom/audio: use TARGET_BOARD_PLATFORM to name audio.primary.xxx.so" into jb-mr2-dev
* commit '01cbedaae471e38f38ed47c544dee75d9447d75e':
qcom/audio: use TARGET_BOARD_PLATFORM to name audio.primary.xxx.so
|
| 01cbedaae471e38f38ed47c544dee75d9447d75e |
04-Apr-2013 |
Iliyan Malchev <malchev@google.com> |
am 0819f6a5: Merge "qcom/audio: use TARGET_BOARD_PLATFORM to name audio.primary.xxx.so" into jb-mr2-dev
* commit '0819f6a5f2bd682eced906ba54499a640d394fb8':
qcom/audio: use TARGET_BOARD_PLATFORM to name audio.primary.xxx.so
|
| 0819f6a5f2bd682eced906ba54499a640d394fb8 |
04-Apr-2013 |
Iliyan Malchev <malchev@google.com> |
Merge "qcom/audio: use TARGET_BOARD_PLATFORM to name audio.primary.xxx.so" into jb-mr2-dev
|
| 323fb9e55e0aafc213bd578c87469bed9caded30 |
04-Apr-2013 |
Iliyan Malchev <malchev@google.com> |
qcom/audio: use TARGET_BOARD_PLATFORM to name audio.primary.xxx.so
Change-Id: I945a37cdb11fe10e0d1c7a4b8d9e2f31b62ae521
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/Android.mk
|
| 9128884b42b65a17621a7e9b19f6580852355dc5 |
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
am 991c0925: am a9024def: audio: implement mute on hdmi multichannel
* commit '991c092525c6f2ca9f4c6cd2f3cee1713d33fc23':
audio: implement mute on hdmi multichannel
|
| 991c092525c6f2ca9f4c6cd2f3cee1713d33fc23 |
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
am a9024def: audio: implement mute on hdmi multichannel
* commit 'a9024defa11f6502ca55425a4803cd00441d51e7':
audio: implement mute on hdmi multichannel
|
| a9024defa11f6502ca55425a4803cd00441d51e7 |
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audio: implement mute on hdmi multichannel
On direct output streams the audio HAL must implement
the volume function. In the case of HDMI the only function
required is to mute audio when volume is 0 as volume
is defined as fixed on digital output streams.
Bug 8541062
Change-Id: Ia1342f6ffb7b7c95c7c386e3e2ee5243fe65051b
al/audio_hw.c
al/audio_hw.h
|
| f289cf0457559cbc14ff511f58ee454c33b64076 |
02-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 36451458: Reconcile with jb-mr2-release - do not merge
* commit '3645145898b9abc49a0b8296aeb46299dfc72e56':
|
| 3645145898b9abc49a0b8296aeb46299dfc72e56 |
02-Apr-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I476bbd02389437601ba40b8c2436d8468cb5f892
|
| 4aafb5c3f14fe4e5fd28d57cc933a60063b7b8ff |
01-Apr-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| fbde5ec436b28bdcd467a584f1d97d50b11ca5d9 |
29-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am ac0d2e9f: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit 'ac0d2e9fc0a6c5b86dcbcacbf9ac6ad8810cbdb1':
|
| ac0d2e9fc0a6c5b86dcbcacbf9ac6ad8810cbdb1 |
29-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: Ic8c24bd4533dfa8c5f12bcf866dc9ece242ad10a
|
| 3e0f97d28191bc39ee874a7e4e988b9b2bc3943a |
29-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am be32c65f: am c301186b: audio/hal: Set playback buffer sizes to integral multiple msec
* commit 'be32c65f1396a9fbf9869a2d950171c54facb0e2':
audio/hal: Set playback buffer sizes to integral multiple msec
|
| be32c65f1396a9fbf9869a2d950171c54facb0e2 |
29-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am c301186b: audio/hal: Set playback buffer sizes to integral multiple msec
* commit 'c301186b49201c8ebf1dc05b336ba0a5e3877408':
audio/hal: Set playback buffer sizes to integral multiple msec
|
| c301186b49201c8ebf1dc05b336ba0a5e3877408 |
20-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
audio/hal: Set playback buffer sizes to integral multiple msec
- The call to pcm_write was taking varying time to complete.
This was because DSP always expects the buffer duration to
be an integral multiple msec. When this is not the case,
DSP waits for the rest of the data to be filled too.
This accumalates the delay and causes the variation
in timing.
- Change the deep buffer playback buffer size to 960 samples(20msec)
and low latency to 240 samples (5msec) to fix the issue.
Change-Id: I9448920e89595a65cf92a5abd9187e02043b699a
al/audio_hw.h
|
| d0c74bc9dc0ac2fc9d9c29c336694ec9733127c9 |
27-Mar-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 77f14b75f719047021947412b86fbfc9c9531bbc |
26-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 81b9b396: am b1995069: qcom/audio/hal: Fix ringtone playback issue on Speaker
* commit '81b9b39666eaa27cc10ed3cd6d075691fde7289b':
qcom/audio/hal: Fix ringtone playback issue on Speaker
|
| 81b9b39666eaa27cc10ed3cd6d075691fde7289b |
26-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am b1995069: qcom/audio/hal: Fix ringtone playback issue on Speaker
* commit 'b199506991c9a93103ed149c6e1ab42c47bb8fc3':
qcom/audio/hal: Fix ringtone playback issue on Speaker
|
| b199506991c9a93103ed149c6e1ab42c47bb8fc3 |
22-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix ringtone playback issue on Speaker
- Start music playback on HDMI, go to settings-->sound-->ringtone
and select a ringtone for playback. The ringtone audio playback
starts only after 15sec.
- When ringtone is selected, the low latency path is switched from
HDMI to Speaker device. The low latency path uses only 2 buffers
of 10.3msec each. If the device switch takes more time, the data
filled kernel buffers meet the stop threshold and the ALSA
framework triggers auto stop on the stream. This results PCM
stream to be blocked for more than 10sec and hence no audio
heard until the write is unblocked.
- Fix the issue by setting the stop threshold to INT_MAX to avoid
auto stop.
- This change also ensures that open_output_stream fails if the
HDMI sink does not support 5.1 or 7.1 playback.
Bug: 8401042
Change-Id: I4c1e04be2c47d67087b1cdda87e2dce77bde58f1
al/audio_hw.c
al/audio_hw.h
|
| 68d7f8a28d173df038967f8b4d75110cfa1313f2 |
20-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 36b4980a: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit '36b4980ac6cbcabf25364122e67cb157b80298ff':
|
| 36b4980ac6cbcabf25364122e67cb157b80298ff |
20-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I3c7bd1f8cb10abd5fe288ad8a40c0db87f18dfb0
|
| 1b8813d1819b536428c9da494f4fca7bfd68e7a2 |
18-Mar-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| d24896cf3f4e91dd7bfa0abbd0677ec6edae0529 |
14-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 2bf04328: am 71c84b70: qcom/audio/hal: Fix the routing logic to route streams independently
* commit '2bf04328e721f2cddaa63ae765788dab6f29250f':
qcom/audio/hal: Fix the routing logic to route streams independently
|
| 2bf04328e721f2cddaa63ae765788dab6f29250f |
14-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 71c84b70: qcom/audio/hal: Fix the routing logic to route streams independently
* commit '71c84b70ff7c428e35ac187ca4a234acac558240':
qcom/audio/hal: Fix the routing logic to route streams independently
|
| 71c84b70ff7c428e35ac187ca4a234acac558240 |
11-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix the routing logic to route streams independently
- Current implementation assumes that the output devices for all
the output streams and voice call will be same. So it updates
the devices on all the output streams when out_set_parameters()
is called on any stream.
- Update the routing logic to support all the streams independently
based on the devices set by audio policy manager on each stream.
- However, on this target there is a limitation that earpiece,
speaker, and headset devices cannot be enabled concurrently as
they share the same backend. Updated routing logic takes care of
this limitation.
Bug: 8239898
Change-Id: I3091be6894210c77c479b872cec39d821d10bd90
al/audio_hw.c
al/audio_hw.h
|
| b547e749d0edcac72980630b6b79c97af0cd6e70 |
12-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am b10756cf: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit 'b10756cf6c650ead77de6cbeb8236fa61b4d9bb7':
|
| b10756cf6c650ead77de6cbeb8236fa61b4d9bb7 |
12-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: Id502924260eb89508aa4e926328ef6b6d97b4b5b
|
| 43044c21172e76e441ac10909af2ceb705a9d1af |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am ff9bdc8c: (-s ours) am 4f5f408b: am 62752611: am 1954242c: am 83e25988: (-s ours) am 294cefd4: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit 'ff9bdc8cd6330475e636ec988e0251e22b906635':
|
| ff9bdc8cd6330475e636ec988e0251e22b906635 |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am 4f5f408b: am 62752611: am 1954242c: am 83e25988: (-s ours) am 294cefd4: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '4f5f408b912ee3e39f31e7e2c558dd6bc5701204':
|
| 4f5f408b912ee3e39f31e7e2c558dd6bc5701204 |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am 62752611: am 1954242c: am 83e25988: (-s ours) am 294cefd4: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '62752611e864f14768615e0b22184feb66993e78':
|
| 62752611e864f14768615e0b22184feb66993e78 |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am 1954242c: am 83e25988: (-s ours) am 294cefd4: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '1954242cb27c68ef311abb3fa12c607145a10d7b':
|
| 1954242cb27c68ef311abb3fa12c607145a10d7b |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am 83e25988: (-s ours) am 294cefd4: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '83e25988078d0c80e07c2055e21f1dc5fcbca0bd':
|
| 83e25988078d0c80e07c2055e21f1dc5fcbca0bd |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am 294cefd4: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '294cefd490efdfc42bfc2a87a80b4d1a364713ee':
|
| 294cefd490efdfc42bfc2a87a80b4d1a364713ee |
11-Mar-2013 |
Jean-Baptiste Queru <jbq@google.com> |
Reconcile with jb-mr1-release - do not merge
|
| dfc5f352b6b2ddb339ae0ce033c6f7d754e1346f |
06-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 1b07c468: am 02317798: qcom/audio/hal: Do not enable Fluence in speaker mode by default
* commit '1b07c46800089bf5e2c0a5a2e4157fb79197ab00':
qcom/audio/hal: Do not enable Fluence in speaker mode by default
|
| 1b07c46800089bf5e2c0a5a2e4157fb79197ab00 |
06-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 02317798: qcom/audio/hal: Do not enable Fluence in speaker mode by default
* commit '02317798dec329868318e75a83c7c654cf5200b3':
qcom/audio/hal: Do not enable Fluence in speaker mode by default
|
| 9b99948079695b2b7e3724030035ec6427bd2e89 |
06-Mar-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 02317798dec329868318e75a83c7c654cf5200b3 |
05-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Do not enable Fluence in speaker mode by default
- Speaker volume is low during voice call.
- Fluence is enabled by default even in speaker mode during voice
call. Mako does not meet the spec for Fluence due to mics
placement issue.
- Fix the issue by not enabling Fluence in speaker mode.
- If the device supports it, set the property to enable fluence.
Bug: 8272345
Change-Id: I9c4726409c4eb8d39dfbbb2f47e3075a6f6d5cc3
al/audio_hw.c
al/audio_hw.h
|
| 37c2223e8c73adc4cfcd5316bd1e9e50da6a167a |
05-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am c3db47c1: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit 'c3db47c15529aaf9c88fd86d63dcc70e6d9c4e66':
|
| c3db47c15529aaf9c88fd86d63dcc70e6d9c4e66 |
05-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: If1ae235f07c2cfe4184e9a18d1fe110a54955e20
|
| 7e58a5862d3a8f048c1c7b1d4fff0ab78fc88456 |
05-Mar-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 74e59d49b1d2f59d64b653c3645cf82d767d6ea4 |
05-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am a1755750: am 096c87f8: qcom/audio/hal: Fix audio routing to wired headset
* commit 'a175575025477413bb6fc96372d8b47ba13e9aac':
qcom/audio/hal: Fix audio routing to wired headset
|
| a175575025477413bb6fc96372d8b47ba13e9aac |
05-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 096c87f8: qcom/audio/hal: Fix audio routing to wired headset
* commit '096c87f83ccc1439acb639dbab00faf5a393afa7':
qcom/audio/hal: Fix audio routing to wired headset
|
| 43d457cbfc4711a45aeac08284bb231535a8bd4e |
05-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am b40b5076: am 3b1816cd: qcom/audio/hal: Use linked list APIs from libcutils
* commit 'b40b50768039961410294593ccb5cc413f7c702f':
qcom/audio/hal: Use linked list APIs from libcutils
|
| b40b50768039961410294593ccb5cc413f7c702f |
04-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
am 3b1816cd: qcom/audio/hal: Use linked list APIs from libcutils
* commit '3b1816cd594eba53a9869d7b23af36daacf58fa1':
qcom/audio/hal: Use linked list APIs from libcutils
|
| 0301366456fa5fe6178936c0e264410c2e45ee8d |
04-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
am 967414c4: (-s ours) Reconcile with jb-mr2-release - do not merge
* commit '967414c4692df6988ea3ae7424199ebf1f89156c':
qcom/audio/hal: Fix the issue audio not routed to headset
qcom/audio/hal: Enable debug logs
Revert "Fix routing for wired headset"
qcom/audio/hal: Fix no voice call audio on bt-sco device
Fix routing for wired headset
|
| 967414c4692df6988ea3ae7424199ebf1f89156c |
04-Mar-2013 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr2-release - do not merge
Change-Id: I1b72e802b0ea0288d030d54011a9af97e3cdd5fd
|
| c6ee42a6d1b99273e7df54530a84ef262526673a |
04-Mar-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 096c87f83ccc1439acb639dbab00faf5a393afa7 |
01-Mar-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix audio routing to wired headset
- Start music playback and plug in headset within 3sec. The
audio is heard on both headset and speaker whereas it is
expected to play on headset.
- Fix the output device updation and selection logic to resolve
the issue.
Bug: 8239898
Change-Id: I476c9ede241e319c90cb960dd302384f41a6b52c
al/audio_hw.c
al/audio_hw.h
|
| 3b1816cd594eba53a9869d7b23af36daacf58fa1 |
28-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Use linked list APIs from libcutils
- Replace the linked list implementation with relevant APIs
available in libcutils
Bug: 8292602
Change-Id: I2db173b845cbf4f35e53738b272f7f4a79279f3b
al/audio_hw.c
al/audio_hw.h
|
| 150dbfe8b5b3ab634604d2a309d4ef9fb7602f4a |
27-Feb-2013 |
Eric Laurent <elaurent@google.com> |
audio: fix audio glitch when closing pcm stream
Holding the audio device mutex while calling pcm_close() is not necessary
and will cause writes on other output streams to be blocked until close
completes which can take several hundred milliseconds.
Not holding the audio device mutex during the whole standby sequence
forces to change the lock order between audio device and output stream mutex.
The result is that we do not acquire the audio device mutex systematically before
the stream mutex in out_write(). This is not a problem with this audio HAL
as set_mode() does not acquire the stream mutex and out_set_parameters() is always
called in the same thread (same priority) as out_write().
Same change done for input threads.
Bug 8267567.
Change-Id: I17bb187c0564200f6362586885e61500d52d5bc2
al/audio_hw.c
al/audio_hw.h
|
| a5dc76f6634d80337f90d3172482282e4d0d7300 |
26-Feb-2013 |
The Android Automerger <android-build@android.com> |
merge in jb-mr2-release history after reset to jb-mr2-dev
|
| 3771884b983f69b43b3000647cb436feb41dd92b |
23-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Move the check for network opeartor to proper place
- Check for the network operator at boot up does not return correct
value always
- For the T-Mobile, US the HANDSET and MIC devices need different
gain settings
- Do the check before enabling those devices
Bug: 8255423
Change-Id: I58011f9c239dce87507b581a62e0dcc09164d15a
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
al/audio_hw.h
|
| 6c811e8d5c62b62bdbdb85be8982ce5364b3189e |
22-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix the issue audio not routed to headset
- Boot the phone without headset plugged in, plug in headset and
play music. Audio is heard on speaker instead of headset.
- The devices of output stream corresponding to music playback
is not being updated correctly.
- Fix by correcting the output device uption logic in
out_set_parameters()
Bug: 8239898
Change-Id: Ie24de09847533660d2280744d33cba7d7fb7d535
al/audio_hw.c
|
| 616289a0082b4db447e59dec49a6d98068ad5312 |
21-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Enable debug logs
- Enable few debug logs to track audio use cases and
sound device activities
- Clean up some ToDos.
- Rename SND_DEVICE_INVALID to SND_DEVICE_NONE.
Bug: 8242117
Change-Id: I0510288334d6a1e71c0846f6d10ac8ba283965a6
al/audio_hw.c
al/audio_hw.h
|
| 7a1a6a96e0b8a022f4d2ee4261dde1801b6f00e9 |
23-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Fix routing for wired headset"
This reverts commit 8637747fab82a99325aae69755646e96fe2e62a5
Change-Id: I59073d3e47548a5697b61a5691b2506bf5876e5e
al/audio_hw.c
|
| 778bf121b16f3bd87d4c8b4deaa577f5d2061733 |
21-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix no voice call audio on bt-sco device
- No audio observed in the first voice call if the device is
booted with the BT-SCO device connected.
- csd_client_enable_device is not being called when the voice
call is enabled which results no audio.
- Fix the issue by making sure that API is called with proper
acdb ids while starting a voice call.
Bug: 8236957
Change-Id: I83a4c00e950b8311162b33087ed73a390c39ca7d
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
|
| a995a357b5da94ad30f155a53325810138dcd718 |
22-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix the issue audio not routed to headset
- Boot the phone without headset plugged in, plug in headset and
play music. Audio is heard on speaker instead of headset.
- The devices of output stream corresponding to music playback
is not being updated correctly.
- Fix by correcting the output device uption logic in
out_set_parameters()
Bug: 8239898
Change-Id: Ie24de09847533660d2280744d33cba7d7fb7d535
al/audio_hw.c
|
| 75d924d06336949440090c214af199fd05d5bb06 |
21-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Enable debug logs
- Enable few debug logs to track audio use cases and
sound device activities
- Clean up some ToDos.
- Rename SND_DEVICE_INVALID to SND_DEVICE_NONE.
Bug: 8242117
Change-Id: I0510288334d6a1e71c0846f6d10ac8ba283965a6
al/audio_hw.c
al/audio_hw.h
|
| 0e4949903cb6655211a95d3070f0951756fa05d1 |
23-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Revert "Fix routing for wired headset""
|
| 4b731445015b1be90ee280d5b2c4901e19589561 |
23-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Fix routing for wired headset"
This reverts commit 8637747fab82a99325aae69755646e96fe2e62a5
Change-Id: I59073d3e47548a5697b61a5691b2506bf5876e5e
al/audio_hw.c
|
| 8b9c5c8c26b4acce5870c48ed8aa76decd823a1e |
21-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix no voice call audio on bt-sco device
- No audio observed in the first voice call if the device is
booted with the BT-SCO device connected.
- csd_client_enable_device is not being called when the voice
call is enabled which results no audio.
- Fix the issue by making sure that API is called with proper
acdb ids while starting a voice call.
Bug: 8236957
Change-Id: I83a4c00e950b8311162b33087ed73a390c39ca7d
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
|
| 75436d5bebe6a7920c1e27f1da0c77dae33c0a46 |
21-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Fix routing for wired headset
Bug: 8239898
Change-Id: Id001206b2e5aa441340b38d62fbcee7449cf5cba
al/audio_hw.c
|
| 8637747fab82a99325aae69755646e96fe2e62a5 |
21-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Fix routing for wired headset
Bug: 8239898
Change-Id: Id001206b2e5aa441340b38d62fbcee7449cf5cba
al/audio_hw.c
|
| 8455fa7d7e51680e45b6a88d28cf48074280e2f9 |
19-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix no audio issue after A2DP disconnect
- No audio heard after A2DP headset is disconnected.
- During A2DP disconnect, the audio HAL receives routing
with device as 0. So the subsequence playback buffers are
dropped as there is no valid output device.
- Fix this issue by ignoring the routing to device 0.
- Update to use speaker mic for voice communication when
using speaker as output device.
Bug: 8214360
Bug: 8219514
Bug: 8230266
Change-Id: Ic19e8e512ae3c5e493014a1ba3c17bf0ddf35e36
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
|
| f99670408844a07cdfabf9a01078ed7ef4c71bbf |
15-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: add support for TTY feature
Bug: 8227215
Change-Id: I4617916b2b9830e7fae3675915939715eab3b9f8
Signed-off-by: Ravi Kumar Alamanda <ralama@codeaurora.org>
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
al/audio_hw.h
|
| c8400637beb896d2f5d7ae980682cd2d072a9da3 |
15-Feb-2013 |
Eric Laurent <elaurent@google.com> |
audio: restored front and back mic capture
Restored support for simultaneous capture of front and back mics
for voice recognition lost when switching to new audio HAL.
Also fixed problem in get_input_snd_device():
AUDIO_DEVICE_BIT_IN must be cleared in input_device before comparison
otherwise anding it with any input device value would return true.
Replaced adev->input_source and adev->in_device by adev->active_input:
this gives access to all info on the active input stream.
Change-Id: I03f02ffc120d6f69669618b4ab63cdcbd7b65877
al/audio_hw.c
al/audio_hw.h
|
| 72c411f8ef451934ababc209eef482b9cc7005a8 |
12-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Add support for Dual MIC feature
- Added support to select Dual mic devices for voice call and
voice recognition use cases.
Bug: 8175884
Change-Id: I7f8cb9e7bd614cfc6010b4cf1baa20ad234c4ddc
Signed-off-by: Ravi Kumar Alamanda <ralama@codeaurora.org>
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
al/audio_hw.h
|
| 610e8cc467a8aa21a3fe25f730793d6c6413d3e7 |
12-Feb-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
qcom/audio/hal: Fix the device switch delay issue during voice call
- Device switch from earpiece to speaker during voice call takes
around 3.5sec which is not acceptable
- Disabling the voice call mixer path consumes 3.5 sec
- Fixed by making sure that device path is deactivated on MDM through
CSD client before disabling voice call mixer path on APQ.
- Also remove incorrect calls to dlerror() and make the dlsym-error messages
more consistent.
Change-Id: Ib23c0a3c0341f41904ca06524bf9d2f4214ad92e
Signed-off-by: Iliyan Malchev <malchev@google.com>
al/audio_hw.c
|
| 006be586d2694724a658f415347bf60aeb17b60b |
08-Feb-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
Audio policy HAL exposes remte stream activity
Audio policy HAL implements audio_policy.is_stream_active_remotely()
Bug 7485803
Change-Id: I221d5891d2bdae8df3892c17f42d5a3972e4883f
egacy/alsa_sound/audio_policy_hal.cpp
|
| 2dfba2b9264a43951889e591260162a67894c0d0 |
18-Jan-2013 |
Ravi Kumar Alamanda <ralama@codeaurora.org> |
add new audio HAL (disabled)
-- build when BOARD_USES_LEGACY_ALSA_AUDIO is not defined
-- under hal/
-- uses audio_route library
Change-Id: Ibf2706ba55e5a2dbd69b5f4cfac8a5cc68220b86
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
al/Android.mk
al/audio_hw.c
al/audio_hw.h
al/edid.c
egacy/Android.mk
|
| 396ebb32c3a800e5c9a972af3fcec55e351996e3 |
09-Feb-2013 |
Iliyan Malchev <malchev@google.com> |
move old audio-HAL code under legacy
Change-Id: Ic70f5ed9690a80801865a60c807657e51a476be6
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAControl.cpp
lsa_sound/ALSAMixer.cpp
lsa_sound/ALSAStreamOps.cpp
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioPolicyManagerALSA.cpp
lsa_sound/AudioPolicyManagerALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/AudioUsbALSA.cpp
lsa_sound/AudioUsbALSA.h
lsa_sound/AudioUtil.cpp
lsa_sound/AudioUtil.h
lsa_sound/CleanSpec.mk
lsa_sound/MODULE_LICENSE_APACHE2
lsa_sound/NOTICE
lsa_sound/acoustics_default.cpp
lsa_sound/alsa_default.cpp
lsa_sound/audio_hw_hal.cpp
lsa_sound/audio_policy_hal.cpp
egacy/Android.mk
egacy/alsa_sound/ALSAControl.cpp
egacy/alsa_sound/ALSAMixer.cpp
egacy/alsa_sound/ALSAStreamOps.cpp
egacy/alsa_sound/Android.mk
egacy/alsa_sound/AudioHardwareALSA.cpp
egacy/alsa_sound/AudioHardwareALSA.h
egacy/alsa_sound/AudioPolicyManagerALSA.cpp
egacy/alsa_sound/AudioPolicyManagerALSA.h
egacy/alsa_sound/AudioStreamInALSA.cpp
egacy/alsa_sound/AudioStreamOutALSA.cpp
egacy/alsa_sound/AudioUsbALSA.cpp
egacy/alsa_sound/AudioUsbALSA.h
egacy/alsa_sound/AudioUtil.cpp
egacy/alsa_sound/AudioUtil.h
egacy/alsa_sound/CleanSpec.mk
egacy/alsa_sound/MODULE_LICENSE_APACHE2
egacy/alsa_sound/NOTICE
egacy/alsa_sound/acoustics_default.cpp
egacy/alsa_sound/alsa_default.cpp
egacy/alsa_sound/audio_hw_hal.cpp
egacy/alsa_sound/audio_policy_hal.cpp
egacy/libalsa-intf/Android.mk
egacy/libalsa-intf/Makefile.am
egacy/libalsa-intf/alsa_audio.h
egacy/libalsa-intf/alsa_mixer.c
egacy/libalsa-intf/alsa_pcm.c
egacy/libalsa-intf/alsa_ucm.c
egacy/libalsa-intf/alsa_ucm.h
egacy/libalsa-intf/alsaucm_test.c
egacy/libalsa-intf/amix.c
egacy/libalsa-intf/aplay.c
egacy/libalsa-intf/arec.c
egacy/libalsa-intf/msm8960_use_cases.h
ibalsa-intf/Android.mk
ibalsa-intf/Makefile.am
ibalsa-intf/alsa_audio.h
ibalsa-intf/alsa_mixer.c
ibalsa-intf/alsa_pcm.c
ibalsa-intf/alsa_ucm.c
ibalsa-intf/alsa_ucm.h
ibalsa-intf/alsaucm_test.c
ibalsa-intf/amix.c
ibalsa-intf/aplay.c
ibalsa-intf/arec.c
ibalsa-intf/msm8960_use_cases.h
|
| 61f449827974e117406693ef2cc6bdfdaf1fc00b |
18-Jan-2013 |
Eric Laurent <elaurent@google.com> |
audio: added support for dual mic capture
Added support for simultaneous capture of front and back mic.
Added device definitions for dual mic config for voice recognition use
case: no pre processing enabled.
stream->channels() reports actual channel mask instead of recontructing
it from channel count.
TODO: check if ACDB settings copied from single mic voice recognition
are correct.
Change-Id: I41282d0af5deb256ef68ec17ee34f5aae7807a6f
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/alsa_default.cpp
ibalsa-intf/msm8960_use_cases.h
|
| c20a61f83701d18da09b17b8eb6d5908c05db7c4 |
11-Jan-2013 |
Devin Kim <dojip.kim@lge.com> |
am b1744367: am e365ed0c: mm-audio: Change delay value when setting device in call mode
* commit 'b17443674f6c5fe12f9658ceb40e357f2b5bbf2d':
mm-audio: Change delay value when setting device in call mode
|
| b17443674f6c5fe12f9658ceb40e357f2b5bbf2d |
09-Jan-2013 |
Devin Kim <dojip.kim@lge.com> |
am e365ed0c: mm-audio: Change delay value when setting device in call mode
* commit 'e365ed0cb095eed5fbc6a86c21642113e2c24f21':
mm-audio: Change delay value when setting device in call mode
|
| e365ed0cb095eed5fbc6a86c21642113e2c24f21 |
07-Jan-2013 |
Devin Kim <dojip.kim@lge.com> |
mm-audio: Change delay value when setting device in call mode
commit 6ae807 (mm-audio: Fix delay at the start of MT voice call) has an issue.
That setting value is too aggressive, RINGTONE buffer is remained and routed
to connected device in call mode.
Add 40ms delay for playing ringtone. This patch will increase about 40ms
for call connection time.
Bug:7946399
Change-Id: I98da5c515b5ba03b413818a8124c213d591c7bc3
lsa_sound/AudioPolicyManagerALSA.cpp
|
| 798b347f4684a1f53cd5dda0a4b9ba864c90d4ae |
20-Dec-2012 |
ty.lee <ty.lee@lge.com> |
am 09f112b2: am 1c9f3b09: audio: separate calibration data for TMUS
* commit '09f112b25fadce8defdc91c8f9a3e6c538034776':
audio: separate calibration data for TMUS
|
| 09f112b25fadce8defdc91c8f9a3e6c538034776 |
20-Dec-2012 |
ty.lee <ty.lee@lge.com> |
am 1c9f3b09: audio: separate calibration data for TMUS
* commit '1c9f3b09a8fcdf65be0bdfa6cb832e6d2ec82fd3':
audio: separate calibration data for TMUS
|
| 1c9f3b09a8fcdf65be0bdfa6cb832e6d2ec82fd3 |
15-Dec-2012 |
ty.lee <ty.lee@lge.com> |
audio: separate calibration data for TMUS
When inserting TMUS SIM, we use different acoustic parameters in call.
Bug: 7716204
Change-Id: Ifce8d6ceb07e4474c28c8c9fe81c4457397e6d0c
lsa_sound/alsa_default.cpp
ibalsa-intf/msm8960_use_cases.h
|
| d30535fefbb23a9892bc29854e9638e0e3bead12 |
11-Dec-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am a82c5b06: am 6ae807c4: mm-audio: Fix delay at the start of MT voice call
* commit 'a82c5b060666b74bdf4c1cee96034ab2fea37b94':
mm-audio: Fix delay at the start of MT voice call
|
| 8969dc0cca45602c2d26166a6a6645c07572d7e7 |
11-Dec-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 7e5b6ac3: am 29555939: mm-audio: Fix delay with first voice call after bootup.
* commit '7e5b6ac34dec11d2e0be2b05272950618a7598cc':
mm-audio: Fix delay with first voice call after bootup.
|
| a82c5b060666b74bdf4c1cee96034ab2fea37b94 |
11-Dec-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 6ae807c4: mm-audio: Fix delay at the start of MT voice call
* commit '6ae807c434c814a2ba6a84198a536d3a4b4153de':
mm-audio: Fix delay at the start of MT voice call
|
| 7e5b6ac34dec11d2e0be2b05272950618a7598cc |
11-Dec-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 29555939: mm-audio: Fix delay with first voice call after bootup.
* commit '295559398bf12612fb208ab3dd24ac08b6f04f2a':
mm-audio: Fix delay with first voice call after bootup.
|
| 6ae807c434c814a2ba6a84198a536d3a4b4153de |
09-Dec-2012 |
SathishKumar Mani <smani@codeaurora.org> |
mm-audio: Fix delay at the start of MT voice call
- Reducing the delay for playing the RINGTONE
before voice call is accepted saves,appr 340ms.
- on this platform reducing the delay wont affect
the RINGTONE as the delay in setting up voice
path after accepting the voice call will
compensate RINGTONE buffers with kernel and
firmware played on the device.
Bug-id: 7612431
Change-Id: Iff5b4545ca7e2316178b0db8cb6760b173c189be
lsa_sound/Android.mk
lsa_sound/AudioPolicyManagerALSA.cpp
lsa_sound/AudioPolicyManagerALSA.h
lsa_sound/audio_policy_hal.cpp
|
| 295559398bf12612fb208ab3dd24ac08b6f04f2a |
08-Dec-2012 |
SathishKumar Mani <smani@codeaurora.org> |
mm-audio: Fix delay with first voice call after bootup.
- Add csd init and deinit to HAL constructor and
destructor
- Add conditional check for voice acdb loader
Bug-id: 7612431
Change-Id: I4165e659fa300abb184e2438a5d730bb2158c094
lsa_sound/AudioHardwareALSA.cpp
ibalsa-intf/alsa_ucm.c
ibalsa-intf/msm8960_use_cases.h
|
| b37b12f11e1d6fb1720aab083e7433ed66d9a3e5 |
28-Nov-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am 49de8867: (-s ours) am 365ce897: Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
* commit '49de886719b27d646c50a5171172bf0865387380':
|
| 49de886719b27d646c50a5171172bf0865387380 |
27-Nov-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am 365ce897: Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
* commit '365ce897f96915c2a303dbfe85937ecb395a7a1d':
|
| 365ce897f96915c2a303dbfe85937ecb395a7a1d |
27-Nov-2012 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
Change-Id: Ib9d689c7e8ac4eca802f694d6dbd902653eb84f1
|
| 5e9014424d2567d14a8c8301866dbb7762b4f7ff |
31-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am 9b41ac7b: (-s ours) Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
* commit '9b41ac7ba55c0f6ee044ef1279941ab3fe1e2a06':
Revert "alsa_sound: change voice-call stop sequence"
alsa_sound: change voice-call stop sequence
|
| 9b41ac7ba55c0f6ee044ef1279941ab3fe1e2a06 |
31-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
Change-Id: I145f70e1468eb53bf0ab955ff096ea795797bd3b
|
| 786a53720cff310032e2ddef7756e7d9d08e6a34 |
19-Oct-2012 |
Dave Burke <daveburke@google.com> |
Revert "alsa_sound: change voice-call stop sequence"
This reverts commit 0f5426b31b9a15743e4621a972cccafc4087aa62
Change-Id: I09e3d3cf17c1d775c76e2f8f6276216fbaa9ac92
lsa_sound/alsa_default.cpp
|
| 73df78f5b4a2ec2b15c9d2f794e6d817a0ea9a37 |
15-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: change voice-call stop sequence
When application processor is paired with an external modem, stop
command should be sent to the DSP on the modem before closing the
Slimbus channels on the application processor.
Bug: 7313016
Change-Id: Ibafeaf9a9badbf32cc955c4e8b5c81e5efdcbb0c
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| 014d19fa3d557cfc931b3dd5ecf8cdedc591d6ca |
18-Oct-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 8190752da6481f7c8948962d6fb805a0363ce652 |
17-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am c26e258c: am 9eb1a4fc: audio: add support for HDMI and speaker combo device
* commit 'c26e258cea80ab7197e527d06097cd6999df49a9':
audio: add support for HDMI and speaker combo device
|
| 80293eb3d4663e542507cfad8cef9201131fda10 |
17-Oct-2012 |
samin.ryu <samin.ryu@lge.com> |
am 03ef6dda: am 5062ccee: alsa_sound: change input device for communication
* commit '03ef6dda80829d0cfbd762643bd2a046869cc41b':
alsa_sound: change input device for communication
|
| c26e258cea80ab7197e527d06097cd6999df49a9 |
17-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 9eb1a4fc: audio: add support for HDMI and speaker combo device
* commit '9eb1a4fc659f58d196cc8e990da2a3fbf9a1c630':
audio: add support for HDMI and speaker combo device
|
| 03ef6dda80829d0cfbd762643bd2a046869cc41b |
17-Oct-2012 |
samin.ryu <samin.ryu@lge.com> |
am 5062ccee: alsa_sound: change input device for communication
* commit '5062ccee15e0b9fafb9c1d3edf6ebee3f1b26a96':
alsa_sound: change input device for communication
|
| 9eb1a4fc659f58d196cc8e990da2a3fbf9a1c630 |
04-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: add support for HDMI and speaker combo device
Bug: 7302453
Change-Id: I732656d185435f0f37437aba0b2f2dd49f65b101
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
ibalsa-intf/msm8960_use_cases.h
|
| 5062ccee15e0b9fafb9c1d3edf6ebee3f1b26a96 |
11-Oct-2012 |
samin.ryu <samin.ryu@lge.com> |
alsa_sound: change input device for communication
Match to VOIP speaker device pair with voice call. In voice call using
speakerphone, we want to use BACK_MIC for input.
Bug: 7329372
Change-Id: I9c42719f17e2be6f0b38292dc74ac3ce54767b64
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| ec0b5d3b014b48e944733337cf4ba441783fdd49 |
17-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 9977042d: am 0f5426b3: alsa_sound: change voice-call stop sequence
* commit '9977042dba61b78fd536c87581cd5eea940866e1':
alsa_sound: change voice-call stop sequence
|
| 9977042dba61b78fd536c87581cd5eea940866e1 |
17-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 0f5426b3: alsa_sound: change voice-call stop sequence
* commit '0f5426b31b9a15743e4621a972cccafc4087aa62':
alsa_sound: change voice-call stop sequence
|
| 0f5426b31b9a15743e4621a972cccafc4087aa62 |
15-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: change voice-call stop sequence
When application processor is paired with an external modem, stop
command should be sent to the DSP on the modem before closing the
Slimbus channels on the application processor.
Bug: 7313016
Change-Id: Ibafeaf9a9badbf32cc955c4e8b5c81e5efdcbb0c
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| 9d6166295be5747c9c6d7942ea1b109ee58e8bed |
10-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am af528e0d: (-s ours) Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
* commit 'af528e0d33cb13e868e6f6e951297564432e1c90':
|
| af528e0d33cb13e868e6f6e951297564432e1c90 |
10-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
Change-Id: Iddfb8f231599cb4d38212670febe9048cc97b572
|
| d765cd646e2b22f95c840426926127a714b8dc94 |
10-Oct-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-factory-release history after reset to jb-mr1-dev
|
| 97690417fba0fd40996babab0349c2ea6d002f6e |
10-Oct-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 0ac90af35b6db35770bbb6e5c95d880994f4cf63 |
10-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am f8b71955: am 89c224e2: alsa_sound: prevent lowlatency errors triggering WD reset
* commit 'f8b719552d3cf2cdb56ff8a08833438195835414':
alsa_sound: prevent lowlatency errors triggering WD reset
|
| f8b719552d3cf2cdb56ff8a08833438195835414 |
10-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 89c224e2: alsa_sound: prevent lowlatency errors triggering WD reset
* commit '89c224e298fd280a3f2a69da8cc930ff6d036e18':
alsa_sound: prevent lowlatency errors triggering WD reset
|
| 89c224e298fd280a3f2a69da8cc930ff6d036e18 |
10-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: prevent lowlatency errors triggering WD reset
- HAL tries to recover from pcm write or read failure
by closing and reopening pcm driver.
- There is no check when pcm is reopened. If the
reopen fails it keeps trying multiple times in loop
- Fix the issue by checking error condition and
return on failure.
Bug: 7253359
Change-Id: Ia15153f5b43bb0f255a8b34b70025a6215484cee
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| 99760d222c3db432ce97c1e851a0ef3d94657bdc |
09-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am 694b033a: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '694b033a0da0774a1a38e253cad536e4b1f897c2':
|
| 694b033a0da0774a1a38e253cad536e4b1f897c2 |
09-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr1-release - do not merge
Change-Id: I9083ad000467a93228455a796ffcd353f097a0d8
|
| 9dc7c7f242cbb216f1298a1046e2bd4ca509a27c |
08-Oct-2012 |
samin.ryu <samin.ryu@lge.com> |
am 72f83fe8: am 2c798912: alsa_sound: avoid pcm_open error
* commit '72f83fe83242142ad3f10fb87f19e6d7d33b7c49':
alsa_sound: avoid pcm_open error
|
| 04c661479d9a6c4f16b2d31f52d4288dbd5e661f |
08-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am d0947edb: am bf1c8748: alsa_sound: Add support for multichannel hdmi
* commit 'd0947edb99697b88bc61f574eeef6041bda9022f':
alsa_sound: Add support for multichannel hdmi
|
| 0a3d83394bb35c3b487bdc6d4ff2ab652a1b6278 |
08-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am b9047ffc: am b357a77d: alsa_sound: Add hdmi audio sink capability discovery
* commit 'b9047ffce7e552901e367018d393e415fbc56c43':
alsa_sound: Add hdmi audio sink capability discovery
|
| 1435542ab56fb5778adf771ba7892d3da9ac27f3 |
08-Oct-2012 |
ty.lee <ty.lee@lge.com> |
am a1852d1b: am 87459f08: alsa_sound : audio path change to BUILT_IN_MIC for camcorder Tx
* commit 'a1852d1b2d42d3964400500aca80274d7e326d3b':
alsa_sound : audio path change to BUILT_IN_MIC for camcorder Tx
|
| dda36d3a8c8b38e5d235cd4d9f2858ac033924ce |
08-Oct-2012 |
chahee.kim <chahee.kim@lge.com> |
am 14f773e8: am a8b76531: audio: fix for mismatched UseCase Type
* commit '14f773e88ca13d9279602cdcf81bf8c22c28e985':
audio: fix for mismatched UseCase Type
|
| 72f83fe83242142ad3f10fb87f19e6d7d33b7c49 |
08-Oct-2012 |
samin.ryu <samin.ryu@lge.com> |
am 2c798912: alsa_sound: avoid pcm_open error
* commit '2c79891211108cd8157124c1b6f9d393729602ff':
alsa_sound: avoid pcm_open error
|
| d0947edb99697b88bc61f574eeef6041bda9022f |
08-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am bf1c8748: alsa_sound: Add support for multichannel hdmi
* commit 'bf1c87481d62736cab8832e9085c121fbafdbb6b':
alsa_sound: Add support for multichannel hdmi
|
| b9047ffce7e552901e367018d393e415fbc56c43 |
08-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am b357a77d: alsa_sound: Add hdmi audio sink capability discovery
* commit 'b357a77dc9c72cb6e327e945421f440052233b51':
alsa_sound: Add hdmi audio sink capability discovery
|
| a1852d1b2d42d3964400500aca80274d7e326d3b |
08-Oct-2012 |
ty.lee <ty.lee@lge.com> |
am 87459f08: alsa_sound : audio path change to BUILT_IN_MIC for camcorder Tx
* commit '87459f0857fe7fab51f45337f330bce03a53c3dc':
alsa_sound : audio path change to BUILT_IN_MIC for camcorder Tx
|
| 14f773e88ca13d9279602cdcf81bf8c22c28e985 |
08-Oct-2012 |
chahee.kim <chahee.kim@lge.com> |
am a8b76531: audio: fix for mismatched UseCase Type
* commit 'a8b76531e6eef695eae174fd628396a5b4c95b7f':
audio: fix for mismatched UseCase Type
|
| 4fea7c7372bec27a4489e511dcdd7f24bdd1fc2a |
08-Oct-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 2c79891211108cd8157124c1b6f9d393729602ff |
05-Oct-2012 |
samin.ryu <samin.ryu@lge.com> |
alsa_sound: avoid pcm_open error
sometimes occur pcm_open error caused by mDevices 0.
so, must not change mDevices to 0.
Bug: 7293209
Change-Id: I3ffaca81dea15145bd8f03e3abedb2840d7c9f2c
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| bf1c87481d62736cab8832e9085c121fbafdbb6b |
26-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: Add support for multichannel hdmi
- Multichannel audio support for HDMI for AAC
format with 5.1 channels max.
Bug: 7156174
Change-Id: I42e92fa2b14d35a5882cc6d84c6651a6a4d5092a
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/alsa_default.cpp
|
| b357a77dc9c72cb6e327e945421f440052233b51 |
26-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: Add hdmi audio sink capability discovery
- Add hdmi sink capabilities parsing utility to support
multi-channel output configuration.
- Update getParameters to calculate supported channels by
hdmi sink.
- Update alsa_default to calculate channel count to set control
option.
Bug: 7156174
Change-Id: Iabb9844c1e5a8b7aa7f168992f8beef79b7df8d2
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/Android.mk
lsa_sound/AudioUtil.cpp
lsa_sound/AudioUtil.h
lsa_sound/alsa_default.cpp
|
| 87459f0857fe7fab51f45337f330bce03a53c3dc |
29-Sep-2012 |
ty.lee <ty.lee@lge.com> |
alsa_sound : audio path change to BUILT_IN_MIC for camcorder Tx
bug: 7268748
Change-Id: I7fb9ed95f6be42555653e4fc000ce8ca7d19ba59
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| a8b76531e6eef695eae174fd628396a5b4c95b7f |
01-Oct-2012 |
chahee.kim <chahee.kim@lge.com> |
audio: fix for mismatched UseCase Type
- useCase name is compared with static string by length of static string
- But, Some useCase names are equal to each other in front string part
- For example, a "HiFi" and a "HiFi rec" are same in part of "HiFi"
- This patch is matched getUseCaseType() function of alsa_ucm.c file
bug: 7294569
related-to-bug: 7263961
Change-Id: Ia83f705756b750798d408a307f0697f9af9ec5dc
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| b515a451c6baff367c0cfb782cb7f92107ae2432 |
03-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am 5b689625: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit '5b689625cf2aaefbdc4e56651c1f138be5417179':
|
| 5b689625cf2aaefbdc4e56651c1f138be5417179 |
03-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr1-release - do not merge
Change-Id: I029549918f3f7236eb5e058372221167d8618a70
|
| b7c7b51c85554174ee64499399e28db27eb2c6ff |
03-Oct-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| a227332ec5b9a0f0b3aafdb43d73f161e8f2808b |
03-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 546bdab3: am 5f24fd93: alsa_sound: fix watchdog error issue
* commit '546bdab37dd2570cc1f4f5ecc89779ac4f6346d0':
alsa_sound: fix watchdog error issue
|
| 546bdab37dd2570cc1f4f5ecc89779ac4f6346d0 |
03-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
am 5f24fd93: alsa_sound: fix watchdog error issue
* commit '5f24fd93afdcc66bfd9246a0e0000c0fd7283b7f':
alsa_sound: fix watchdog error issue
|
| 5f24fd93afdcc66bfd9246a0e0000c0fd7283b7f |
02-Oct-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix watchdog error issue
- Issue happens when pcm_open fails after routing is done.
- Currently HAL returns zero bytes when pcm_open fails,
Return actual bytes so that audioflinger can drop the buffer.
Bug-id: 7253359
Change-Id: I5d989539b0f9252577dc81a3ba34d467758bf717
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/AudioStreamOutALSA.cpp
|
| d0734d4044e455b0671ade34f13e7ef5121e699d |
02-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
am cf7bd883: (-s ours) Reconcile with jb-mr1-release - do not merge
* commit 'cf7bd88327618664ea31bc9ee2cb76368d0de44b':
alsa_audio: route call voice to phone when on HDMI
alsa_sound: fix for SIP call mute issue.
Revert "alsa_sound: fix for SIP call mute issue."
alsa_sound: fix for SIP call mute issue.
alsa_sound: fix for output device routing during video chat
audio: add the headset tx
hardware/alsa_sound: Change the device disable sequence
|
| cf7bd88327618664ea31bc9ee2cb76368d0de44b |
02-Oct-2012 |
The Android Open Source Project <initial-contribution@android.com> |
Reconcile with jb-mr1-release - do not merge
Change-Id: Ie754a300e8665e6dfd25260714851d7bbe99261f
|
| 9f99e382ef161dda2ff137ea8be80f26f150399e |
29-Sep-2012 |
Sungmin Choi <sungmin.choi@lge.com> |
alsa_audio: route call voice to phone when on HDMI
Currently call voice is routed to HDMI rx/tx when HDMI cable is plugging. So
call voice is muted. MO/MT call voice routes to speaker/back-mic and
communication routes to HDMI rx/back-mic when HDMI cable is plugging.
Bug: 7218296
Change-Id: I53fc4d5b77ad6502a32c4e8d06e17141e2aa4be2
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| 6ebf0aa4df1a2484f52a894384286a084ac0a129 |
29-Sep-2012 |
Sungmin Choi <sungmin.choi@lge.com> |
alsa_audio: route call voice to phone when on HDMI
Currently call voice is routed to HDMI rx/tx when HDMI cable is plugging. So
call voice is muted. MO/MT call voice routes to speaker/back-mic and
communication routes to HDMI rx/back-mic when HDMI cable is plugging.
Bug: 7218296
Change-Id: I53fc4d5b77ad6502a32c4e8d06e17141e2aa4be2
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| 1af98a6183584b3f7254a4ddff6ca629cd77e037 |
27-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix for SIP call mute issue.
-Mute VOIP through audio path is not supported by kernel,
so calling setVoipMicMute() to disable ""Voip Tx Mute"" fails
-Mute VOIP by cleanning buffers read if setMicMute is set
-Remove mVoipMicMute and use mMicMute for voice and VOIP
Bug: 7213748
Change-Id: I62be8456447425cbd1521083782802effcb6d326
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| 6ccde10fdc6591e5cd21ed32c2b5817aaeb1a208 |
27-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix for SIP call mute issue.
-Mute VOIP through audio path is not supported by kernel,
so calling setVoipMicMute() to disable ""Voip Tx Mute"" fails
-Mute VOIP by cleanning buffers read if setMicMute is set
-Remove mVoipMicMute and use mMicMute for voice and VOIP
Bug: 7213748
Change-Id: I62be8456447425cbd1521083782802effcb6d326
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| cda1a832f4aac97b14d788bc742308e9d039e2e1 |
28-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 604ccc983c49e360ab60427a65d507fbc7672300 |
28-Sep-2012 |
Iliyan Malchev <malchev@google.com> |
Revert "alsa_sound: fix for SIP call mute issue."
This reverts commit f859d3e80b790d4b525515918f1c6889383cc495.
bug: 7250052
reopens-bug: 7213748
Change-Id: I33312c0be196b327887edc93c0b766bcf86a94bc
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| 80baa42f653a8ea57847d6c2543df8400349433f |
28-Sep-2012 |
Iliyan Malchev <malchev@google.com> |
Revert "alsa_sound: fix for SIP call mute issue."
This reverts commit f859d3e80b790d4b525515918f1c6889383cc495.
bug: 7250052
reopens-bug: 7213748
Change-Id: I33312c0be196b327887edc93c0b766bcf86a94bc
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| 5481e7629e04e4d50ac5b471d0dafd720c0a5241 |
27-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix for SIP call mute issue.
-Mute VOIP through audio path is not supported by kernel,
so calling setVoipMicMute() to disable ""Voip Tx Mute"" fails
-Mute VOIP by cleanning buffers read if setMicMute is set
-Remove mVoipMicMute and use mMicMute for voice and VOIP
Bug: 7213748
Change-Id: Ie2a200470c16da2e4ae991c0814ec8dfb4666833
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| 466e8a8fa8d4e6186b72c25cd0289356828177b3 |
27-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audio: reset output device when disconnecting HDMI
When HDMI is disconnected, the audio HAL should stop writing to
the PCM driver to avoid a write error and a long timeout.
THe problem is that if no audio track is active, the policy manager
will send a 0 device upon HDMI disconnection which is normally ignored.
The fix consists in forcing the device to speaker when transitioning from
HDMI to 0 device.
Bug 7141149.
Change-Id: I3e5878a1d6ec0446f7044eff95e6641332c718bf
lsa_sound/ALSAStreamOps.cpp
|
| f859d3e80b790d4b525515918f1c6889383cc495 |
27-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix for SIP call mute issue.
-Mute VOIP through audio path is not supported by kernel,
so calling setVoipMicMute() to disable ""Voip Tx Mute"" fails
-Mute VOIP by cleanning buffers read if setMicMute is set
-Remove mVoipMicMute and use mMicMute for voice and VOIP
Bug: 7213748
Change-Id: Ie2a200470c16da2e4ae991c0814ec8dfb4666833
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
|
| 341c0602f27fe7e3cc629a0b6ac1c2b5df256309 |
27-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 77780382b45794eb5bc0e8589d9b7c96bb406772 |
22-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: Add support for flexible buffer size for recording
- In the current implementation, all the read calls to the driver
are limited to 320 bytes only. This results performance overhead
for recording at higher sampling rates.
- Added support for flexible buffer size to allow upto 4096 bytes.
Bug: 7223456
Change-Id: Ic0522d92de905b04481a0d8daa103c77552257e8
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/alsa_default.cpp
|
| 453d78c3bd4bd706bf381a82788e0ed88d2f96d0 |
22-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix for output device routing during video chat
Bug: 7163358
- Output device for MODE_IN_COMMUNICATION could be earpiece
or speaker, do not update output device in switchDevice()
if input device is BUILTIN_MIC
Change-Id: I2a97a63f4cb57ef114695d3fd917282612b7a346
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| c669c27ef9c5663692c3dd8818bb6e7fcdc39b7a |
22-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: fix for output device routing during video chat
Bug: 7163358
- Output device for MODE_IN_COMMUNICATION could be earpiece
or speaker, do not update output device in switchDevice()
if input device is BUILTIN_MIC
Change-Id: I2a97a63f4cb57ef114695d3fd917282612b7a346
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
|
| 85f3fc5fd4b2b6ae79502c172148b9a27fda6fa5 |
25-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 925596f513542570d660da7902e9d2219c9cfc69 |
12-Sep-2012 |
ty.lee <ty.lee@lge.com> |
audio: separate device for voice call
1. avoid to change gain as below scenarioes
- after call end, immediately music start with headset
- when device is changed from headset to speaker in voice call
with touch effect sound, temporarily voice call gain is increased
until changed to speaker
2. this commit set different configuration of voice and audio.
Change-Id: I89f1c1736d1ba56cb90a5e94f673b3f0439ab089
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/alsa_default.cpp
ibalsa-intf/msm8960_use_cases.h
|
| 9858010096e3469c3b1ccb5372ebba9cfbdeb384 |
20-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: ucm: Route voice call only after enabling device
- Voice call is sometimes enabled without enabling both Rx and
Tx device. This is resulting in improper configuration of
voice call
- Fix this problem by checking if both Rx and Tx devices are
enabled before routing voice call
Change-Id: Ie0850eb7d0beafe7db150138b3aff1b2696db591
ibalsa-intf/alsa_ucm.c
ibalsa-intf/msm8960_use_cases.h
|
| ed3c0fdecb72890240d52a8ba3b5ecd05331736a |
21-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: Correct the input buffer size computation
- getInputBufferSize() expects the input channelCount to
compute the buffer size but not the channelMask.
- Removed the condition check to allow opening multple
input streams.
Change-Id: I927f2ed0423353a164d907035adfd883c5dae556
Bug-id: 7184317
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/audio_hw_hal.cpp
|
| 9a0ca062f87d50bef085d112292e11b8d790ffbb |
21-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 639ce754c700a46d191ef114afd1fdfc44fb80a5 |
20-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: Remove acdb, csd-client dependency
- Remove acdb and csd-client modules header files
dependency from HAL and UCM
- cleanup makefiles to not include acdb and csd
modules headers
Change-Id: I18712fcd50803f2d819ccfbaa0324d22d21b6293
Bug-id: 6815609
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/alsa_default.cpp
ibalsa-intf/Android.mk
ibalsa-intf/alsa_ucm.c
|
| 98913ad4e326628372f2c2ca3ba6b774c336ee44 |
19-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 9efed76df6b7c75d170e8f900f875f4329587719 |
19-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: enable necessary audio logs for debugging purpose
- Remove LOCAL_LOGD flag and enable audio necessary logs
Change-Id: Ia767ff8deaae4aa27252450eb94c64506cb8a495
lsa_sound/ALSAStreamOps.cpp
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/alsa_default.cpp
ibalsa-intf/alsa_ucm.c
|
| 8b9ec98dfd49dad9c11ca4a208e8aca0572639cd |
18-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 2aa250712d5be43075f85fc5b663616915c775a7 |
18-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: remove redundant configuration files from audio HAL
- Remove UCM files from audio HAL
- Related files are stored in OEM folder under device
Change-Id: I0f86e57e33a16fc543b2cc81145426b6724be6f2
Signed-off-by: Iliyan Malchev <malchev@google.com>
ibalsa-intf/Android.mk
ibalsa-intf/snd_soc_msm/snd_soc_msm
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x_Fusion3
ibalsa-intf/snd_soc_msm/snd_soc_msm_Sitar
|
| e9c7854606c7b731eae7748c15db180ff7e4a368 |
18-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: Decrease the lowlatency playback buffersize to 1024 bytes
- Decrease the lowlatency playback buffersize to 1024 bytes
and latency to 11ms.
Change-Id: I39e0580846fcf53cb4d2a708b53c39abc6e62469
lsa_sound/AudioHardwareALSA.h
|
| 304e61b784982c00938f519ae7762247e3c9a94a |
15-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Remove audio_policy.conf here
Use device/... version instead
Bug: 7172210
Change-Id: I8835e6b924758565c139f58e84a6585247c15730
lsa_sound/Android.mk
lsa_sound/CleanSpec.mk
lsa_sound/audio_policy.conf
|
| f9e4091e036cc653c0d1af97b7b5c7b44e00f6d6 |
16-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 5ff7a02ca497656b711919b81a783e6a10cdcc76 |
14-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: Remove HAL, UCM dependency on acdb and csd client modules
b/6815609
- acdb and csd client modules are linked into HAL, UCM
at compile time.
- Remove the compile time dependency by loading the modules at
runtime.
Change-Id: Iab9684a4564e5dccd8eb6e07017959886d2f57b1
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/alsa_default.cpp
ibalsa-intf/Android.mk
ibalsa-intf/alsa_ucm.c
ibalsa-intf/msm8960_use_cases.h
|
| 9ea52b2d34e3ce5d52073be595945e6212c444b7 |
14-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| d4f2d069126d564b2ef504e0cb59369d8e1e9f32 |
03-Sep-2012 |
ty.lee <ty.lee@lge.com> |
audio: fix problem of routing device 0 after A2DP play
This patch avoids abnormal operation in AudioStreamOutALSA::write().
When BT device is turned On/Off, if routing device is 0, then pcm open
will be failed.
- A variable "mDevices" is set to "0" when device is switched to BT A2DP
from speaker during playing music.
- If variable "mDevices" is "0", when PCM data is writing to buffer
in A2DP stream, then a routing device is set "no device".
- And when device is switched to ALSA stream, and PCM open will be failed.
This causes media crash as killing ALSA.
Change-Id: Id416bed3f59bfbe857b2a9fd504f3f18746fb4cd
lsa_sound/AudioStreamOutALSA.cpp
|
| e42406e02e32dbb398a2fa0c452e52526b763c8d |
30-Aug-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: Remove pcm stream close for flushing buffers
- Currently pcm stream is closed to flush buffers while switching
between speaker+headset combo device and headset device.
- Fix this problem by removing unnecessary pcm stream close.
Bug-id: 7051374
Change-Id: Id2faaeadd2a8e009f67e85beee42bbfe47b959f4
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/alsa_default.cpp
|
| 308dc9a1a8969a10f5600f697870cfb87c22146d |
13-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 018d1c5b5746dc189b2d7b86a87f1d2fd219994a |
11-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: increase deep-buffer playback buffer size and latency
b/7129131
- Increase buffersize to 4k and latency to 170ms when HAL is
configured for deep buffer output
Change-Id: Ib4d83eda81714f3aff169f84f29e582aec50e9c7
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
|
| 8861338c2a520669a870ee5f3e0c740e438c99bc |
14-Aug-2012 |
SathishKumar Mani <smani@codeaurora.org> |
alsa_sound: add support for low latency playback and recording
b/6865729
- Add deep buffer output in config file
- Configure deep buffer output if flag
AUDIO_OUTPUT_FLAG_DEEP_BUFFER is set,
otherwise configure low latency output.
- Add support for low latency recording
- Enable low latency recording path with
system property
- For 2 buffers and 2048 bytes, reduce
PLAYBACK_LOW_LATENCY to 21.5 ms
Change-Id: I3c0d54fa473fe89df5a3924de483f16975f4000e
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/alsa_default.cpp
lsa_sound/audio_hw_hal.cpp
lsa_sound/audio_policy.conf
|
| 92ce81d8aeb4a3af6af27e3db020fb90c8eb9b85 |
12-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 94de74c9450130f9dd01972414b18cb358062876 |
03-Sep-2012 |
ty.lee <ty.lee@lge.com> |
audio: avoid latent media server crash during device is booting
Sometimes media server is killed by alsa_ucm during device is booting.
A alsa_ucm gets a index of "use case device" from get_use_case_index()
function.
But, get_use_case_index() function can be returned a wrong value
(dev_index = -22 used index of dev_list[] array) at booting time.
This patch avoids latent media server crash causing abnormal value
as using index of dev_list[] array.
Change-Id: I961465f99b994d3ab1b26ee74d6d0978159960f4
ibalsa-intf/alsa_ucm.c
|
| 0a019914794cc27f5dacddf4e5dbc019dbe21788 |
11-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audio: fix for log spamming from ALSA
b/6984795
- Use ALOGD instead of ALOGE for debugging logs
- Use ALOGE only for error messages
Change-Id: Ifad36acbe93d48f2bdce3e29c2b06654daf08821
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/alsa_default.cpp
ibalsa-intf/alsa_pcm.c
ibalsa-intf/alsa_ucm.c
|
| 8a9785b5e6a199a6c64ac671afc2e8036c7ec13f |
11-Sep-2012 |
Ajay Dudani <adudani@codeaurora.org> |
Reduce debug logs for some normal usecases
b/6984795
Change-Id: Ic5c53a458fc1f72cbe7269e58374412200fb6ff8
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/alsa_default.cpp
|
| f742751fd3fe3fe2ed0dfbed6278351b3c50d53a |
09-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 7801df9b35653562139ddaaa5c29c61f0eddc92f |
28-Aug-2012 |
Eric Laurent <elaurent@google.com> |
audio: changes for audio device API 2.0.
Removed implementations of obsolete function get_supported_devices()
Audio HAL wrapper provides conversion between new and old device enums.
It exposes a rev 2.0 audio device API to the audio framework and allow
legacy implementation to use old device enums.
TODO: make sure that Qualcomm proprietary device IDs are defined properly.
Change-Id: I6779d6c9bccb531d70d84136cdc8d56208b5c934
lsa_sound/audio_hw_hal.cpp
|
| 39a97909720475e814f8e3aa280c1468f09a021d |
05-Sep-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| a4e236767e1253c0393b15a82c3fe1cd1304c2ff |
28-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "audio: fix harmoic caused by resampler in high frequency"
This reverts commit 2e7101d04a211e1b32a5cb7842e7dfa53935de8d.
lsa_sound/AudioHardwareALSA.h
|
| 13e874988aaca5bae568b9b14e6f62de136e3b9e |
29-Aug-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| 15defbaaf797973200e6156a0d00982aa039da4e |
13-Aug-2012 |
ehgrace.kim <ehgrace.kim@lge.com> |
audio: add the headset tx
Change-Id: If15ef8362dd2800c41fb945b96488844db29e5db
lsa_sound/alsa_default.cpp
|
| 30f8dc20ad0e7ec0844450df61e5421f52b1205f |
29-Aug-2012 |
Iliyan Malchev <malchev@google.com> |
Merge "audio: add the headset tx" into jb-mr1-dev
|
| 249f87d48a57a50480e41b397cb4ca31687abb34 |
13-Aug-2012 |
ehgrace.kim <ehgrace.kim@lge.com> |
audio: add the headset tx
Change-Id: If15ef8362dd2800c41fb945b96488844db29e5db
lsa_sound/alsa_default.cpp
|
| 50b9a1eb110a0f6b95472b340d65bb5eb3589a64 |
28-Aug-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge "Update (disabled) audio policy implementation to new interface" into jb-mr1-dev
|
| a6ec2c4e605e0c50342aa543337853ebbc5f38f3 |
28-Aug-2012 |
The Android Automerger <android-build@android.com> |
merge in jb-mr1-release history after reset to jb-mr1-dev
|
| a97e6f657522513449514a677f19a0ff28df0059 |
15-Aug-2012 |
ty.lee <ty.lee@lge.com> |
audio: fix parameter for tty_mode
Change-Id: Ic386415a1a62e133bbd5e9726240afd7e4dc301f
lsa_sound/AudioHardwareALSA.cpp
|
| e6be9bd48a3abc4809dd1eba9bbcbdb226d3d9c9 |
14-Aug-2012 |
ty.lee <ty.lee@lge.com> |
audio: fix fluence acdb id
Change-Id: I322bd9f04badc9c33de618b58d47894f39abdce1
ibalsa-intf/msm8960_use_cases.h
|
| 2e7101d04a211e1b32a5cb7842e7dfa53935de8d |
14-Aug-2012 |
ty.lee <ty.lee@lge.com> |
audio: fix harmoic caused by resampler in high frequency
Change-Id: I437577e9b56c4559d843fe2b97e60d4a9b520b25
lsa_sound/AudioHardwareALSA.h
|
| e195436fce6ef106a2156c7eefd97a3ea21e4d18 |
27-Aug-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Update (disabled) audio policy implementation to new interface
Update implementation of AudioPolicyInterface to use the
audio_devices_t type of system/audio.h instead of the
device enum in AudioSystem.
Note that the affected file is currently disabled in the makefile.
Change-Id: I54f07aa3c803643ea530de0e4d27aee73a9fc1fc
lsa_sound/audio_policy_hal.cpp
|
| fb65fc8a29c7eb9e3ede46168df60c55943821b5 |
21-Aug-2012 |
Ajay Dudani <adudani@codeaurora.org> |
hardware/alsa_sound: Change the device disable sequence
Before tearing down codec, disable the MDM device first so that
slimbus channel is disconnected from MDM to avoid overflow on
codec slimbus interface
Change-Id: I20ef6fe97b3e72b6fe17243a55e4e20ed30df93c
lsa_sound/alsa_default.cpp
|
| 91bf891a50aef64d975870c18658a8be3261d3d6 |
21-Aug-2012 |
Ajay Dudani <adudani@codeaurora.org> |
hardware/alsa_sound: Change the device disable sequence
Before tearing down codec, disable the MDM device first so that
slimbus channel is disconnected from MDM to avoid overflow on
codec slimbus interface
Change-Id: I20ef6fe97b3e72b6fe17243a55e4e20ed30df93c
lsa_sound/alsa_default.cpp
|
| d1c97008da7b8081eb39bfa8fddb10196b2945f8 |
14-Aug-2012 |
SathishKumar Mani <smani@codeaurora.org> |
libalsa-intf: Add support for low-latency playback/recording
- Add mixer controls to support low latency playback
and recording
- Add usecase verb, modifier for low latency Rx and Tx
- Add low latency Rx and Tx capabilties in UCM
Change-Id: Icb26577f7a03886e9bdef06d9d742efced05bfee
Signed-off-by: Iliyan Malchev <malchev@google.com>
ibalsa-intf/alsa_ucm.c
ibalsa-intf/msm8960_use_cases.h
ibalsa-intf/snd_soc_msm/snd_soc_msm
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x_Fusion3
ibalsa-intf/snd_soc_msm/snd_soc_msm_Sitar
|
| 4bb2a8f5b006a3016077a4675fbf176924ed6b15 |
18-Aug-2012 |
ehgrace.kim <ehgrace.kim@lge.com> |
audio: enable the audio calibration for APQ side
For audio calibration of Multimedia, add the libacdbloader
Change-Id: If822f486f10aeed33a537710ff4d5075225cba8a
ibalsa-intf/Android.mk
|
| db7a5cc416289d3f3a24622aeb816752a0e6b82b |
07-Aug-2012 |
ehgrace.kim <ehgrace.kim@lge.com> |
audio: enable the QACT connection
Change-Id: Icb98d638f30f21be834d8e65883420425b31537a
lsa_sound/Android.mk
|
| 74060dee8c9a414ad0345176e2422de18a4ff183 |
01-Aug-2012 |
ty.lee <ty.lee@lge.com> |
audio: add separate audio Input path feature
- Separate audio input path can be set using input source of audio parameter
1. Voice Recognition
2. Camcording
3. etc.
Change-Id: I7ab3b529a8d39af412d10d2d7ab4ce111db967bb
lsa_sound/ALSAStreamOps.cpp
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.h
lsa_sound/alsa_default.cpp
ibalsa-intf/msm8960_use_cases.h
|
| 924f79851aaa4eebe593a456203d1c39d3c53d73 |
01-Aug-2012 |
ty.lee <ty.lee@lge.com> |
audio: add Dual MIC scenario in call feature
- Dual MIC solution(Fluence) feature in Built-in MIC used scenarioes.
1. Handset
2. 3-Pole Headphones
Change-Id: I5cb7d909785ac583ced01276b987c4ba811d8404
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/alsa_default.cpp
|
| 10dfa859150b9ff603e56459041c3314536b7a0d |
01-Aug-2012 |
ty.lee <ty.lee@lge.com> |
audio: change output device in speaker phone call
Change-Id: Ie2c6e1f1e483fdb6caaeccf09d85dd3413945799
lsa_sound/alsa_default.cpp
|
| 86c852bd4ef7ab3bce78f27e102ca7c00d457966 |
20-Jul-2012 |
Ajay Dudani <adudani@codeaurora.org> |
alsa: Wrap verbose logs around LOCAL_LOGD and keep them disabled
Change-Id: I52f432ff0d9a0af7a050eac83131556afeab0af9
lsa_sound/ALSAStreamOps.cpp
lsa_sound/alsa_default.cpp
ibalsa-intf/alsa_ucm.c
|
| 92919435cbe39335d24a7c0fa6a0adfd4d47d974 |
28-Jun-2012 |
Ajay Dudani <adudani@codeaurora.org> |
alsa_sound: enable csd-client for fusion3 voice call
Change-Id: Ibefe43d9dc669af74c88b8f4054af7ca5ce2caac
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioStreamInALSA.cpp
|
| 498654c8b2d7d3abe03f564f4583a1735e89c471 |
05-Jul-2012 |
Ajay Dudani <adudani@codeaurora.org> |
Add supported devices for primary input and output
Change-Id: I225bbee57b05ee94e8fdaa00fe08afc61fd42fca
lsa_sound/audio_policy.conf
|
| 91e9fad9dff8746a5cd9d44ee4f1313d3ad7cfdc |
03-Jul-2012 |
ehgrace.kim <ehgrace.kim@lge.com> |
audio: modify the SPEAKER_RX_ACDB to support both MONO and STEREO speaker
Change-Id: I2e66f430b18ecfa3bfcdbb7c9f6d50354e2d0709
lsa_sound/alsa_default.cpp
ibalsa-intf/alsa_ucm.c
ibalsa-intf/msm8960_use_cases.h
|
| c1748db63cc975291af971c85c9f902aa7a73f30 |
02-Jul-2012 |
Sungmin Choi <sungmin.choi@lge.com> |
Revert "audio: fix the audio ucm file"
This reverts commit 31e69ccf001c02e2a9e820e6589e5c9cf9e074cf.
ibalsa-intf/msm8960_use_cases.h
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x_Fusion3
|
| 3bf1700cac9597e9be0866bfa9e54c689d63b3e1 |
30-Jun-2012 |
Iliyan Malchev <malchev@google.com> |
hardware/qcom/audio: fix build for other targets
Change-Id: I4f5c999f6cc215ded54eb6f4b9ea54666c51b7a0
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
|
| 31e69ccf001c02e2a9e820e6589e5c9cf9e074cf |
23-Jun-2012 |
ehgrace.kim <ehgrace.kim@lge.com> |
audio: fix the audio ucm file
Change-Id: I2435ac3af619610b3b18b34f68339058c10fb4a4
ibalsa-intf/msm8960_use_cases.h
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x_Fusion3
|
| 9746c4758b161e26eec92b1ef1ff1bf0ba0bd268 |
19-Jun-2012 |
Ajay Dudani <adudani@codeaurora.org> |
audio: Updates to comply with standard libhardware_legacy audio hal
Revert back to original libhardware_legacy audio hal headers from AOSP.
This makes corresponding code changes to disable features to match
audio hal implementation with standard libhardware_legacy audio hal.
Change-Id: Ibf1e50d3fffc8280ba417a26172c0f04206474e3
ndroid.mk
lsa_sound/ALSAControl.cpp
lsa_sound/ALSAMixer.cpp
lsa_sound/ALSAStreamOps.cpp
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioPolicyManagerALSA.cpp
lsa_sound/AudioPolicyManagerALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/AudioUsbALSA.cpp
lsa_sound/AudioUsbALSA.h
lsa_sound/acoustics_default.cpp
lsa_sound/alsa_default.cpp
lsa_sound/audio_hw_hal.cpp
lsa_sound/audio_policy.conf
lsa_sound/audio_policy_hal.cpp
ibalsa-intf/Android.mk
ibalsa-intf/alsa_audio.h
ibalsa-intf/alsa_mixer.c
ibalsa-intf/alsa_pcm.c
ibalsa-intf/alsa_ucm.c
ibalsa-intf/aplay.c
ibalsa-intf/arec.c
ibalsa-intf/msm8960_use_cases.h
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x_Fusion3
ibalsa-intf/snd_soc_msm/snd_soc_msm_Sitar
|
| 4113f34dfbaa8d82a5e1ef0265e916317161984d |
11-Jun-2012 |
Iliyan Malchev <malchev@google.com> |
working msm8960 audio.primary and audio_policy HALs
These HALs build on JB and work on mako
Change-Id: I89bff4f1269d47a33d8e2a53a0b65d69aaf53240
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAControl.cpp
lsa_sound/ALSAMixer.cpp
lsa_sound/ALSAStreamOps.cpp
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioPolicyManagerALSA.cpp
lsa_sound/AudioPolicyManagerALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/AudioUsbALSA.cpp
lsa_sound/acoustics_default.cpp
lsa_sound/alsa_default.cpp
lsa_sound/audio_hw_hal.cpp
lsa_sound/audio_policy_hal.cpp
|
| 4765c439491ddda3de658e62cc4a64d10e726b34 |
11-Jun-2012 |
Iliyan Malchev <malchev@google.com> |
initial audio HAL implementation for mako
alsa_sound is imported from codeaurora at:
c1217338f349fe746e0933fcf9b1b288b532808d
[remote "quic"]
url = git://git-android.quicinc.com/platform/hardware/alsa_sound.git
review = review-android.quicinc.com
projectname = platform/hardware/alsa_sound
fetch = +refs/heads/*:refs/remotes/quic/*
Change-Id: Ic985cc3a1088c3957b6e2ac5537e2c36caaf7212
Signed-off-by: Iliyan Malchev <malchev@google.com>
lsa_sound/ALSAControl.cpp
lsa_sound/ALSAMixer.cpp
lsa_sound/ALSAStreamOps.cpp
lsa_sound/Android.mk
lsa_sound/AudioHardwareALSA.cpp
lsa_sound/AudioHardwareALSA.h
lsa_sound/AudioPolicyManagerALSA.cpp
lsa_sound/AudioPolicyManagerALSA.h
lsa_sound/AudioStreamInALSA.cpp
lsa_sound/AudioStreamOutALSA.cpp
lsa_sound/AudioUsbALSA.cpp
lsa_sound/AudioUsbALSA.h
lsa_sound/MODULE_LICENSE_APACHE2
lsa_sound/NOTICE
lsa_sound/acoustics_default.cpp
lsa_sound/alsa_default.cpp
lsa_sound/audio_hw_hal.cpp
lsa_sound/audio_policy_hal.cpp
ibalsa-intf/Android.mk
ibalsa-intf/Makefile.am
ibalsa-intf/alsa_audio.h
ibalsa-intf/alsa_mixer.c
ibalsa-intf/alsa_pcm.c
ibalsa-intf/alsa_ucm.c
ibalsa-intf/alsa_ucm.h
ibalsa-intf/alsaucm_test.c
ibalsa-intf/amix.c
ibalsa-intf/aplay.c
ibalsa-intf/arec.c
ibalsa-intf/msm8960_use_cases.h
ibalsa-intf/snd_soc_msm/snd_soc_msm
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x
ibalsa-intf/snd_soc_msm/snd_soc_msm_2x_Fusion3
ibalsa-intf/snd_soc_msm/snd_soc_msm_Sitar
|
| 1e47753cbf3fa2a5178464ba806fa40273338f54 |
11-Jun-2012 |
Chad Jones <chadj@google.com> |
Initial empty repository
|