android_hardware_qcom_audio, branch stable/cm-12.0-caf-8974-YNG3C

hal: Shush some debug logging

2015-02-25T08:55:50+00:00

Change-Id: I522a0f0379143315f95dbf6a0f40cb6507287c47

audio: Fix for clip skip issue during SSR

2015-02-23T20:30:03+00:00

- Specific AAC clips are skipped to next clip on triggering SSR
- When SSR is triggered while track is waiting for stream end event for
  low bit rate clips, drain is unblocked and playback is switched to
  next track
- Fix is not to post drain ready event during SSR as actual stream end
  is not reached with more compressed data buffered to DSP

  CRs-Fixed: 766541

Change-Id: Id1573160001a2a252dc6613b58f70233e77cc3b4

hal: Enhance compress offload logging

2015-02-23T20:27:05+00:00

Add/Enable msg logging
-Salient funtion calls to tinycompress
-Salient funtion calls from frameworks
-Events sent to frameworks
-miscellaneous important events

Change-Id: I493d98d3f3c1216f7c6030a539e01d1f2dd8c202

hal: align fragment size to 24 instead of 1024 for PCM offload

2015-02-14T08:29:55+00:00

Aligning PCM buffer size to 1024 bytes is not required in case
of PCM offloading. It actually creates problem when playing
6 ch, 96KHz, 16-bit stream PCM in offload mode, due to aligning
PCM data becomes unequal for the six channels.
Align the buffer to LCM of 2, 4, 6, 8 i.e. 24.

Change-Id: I0823ca2ccdd5fceaf23a182b763a1d69f91f602f
CRs-fixed: 780001

hal: Correctly set backend to best available parameters

2015-02-11T23:36:33+00:00

* For the case where CALL/IN_COMMUNICATION, new_bit_width and
  new_sample_rate were unconditionally reset to 0, resulting
  in a sample rate/bit width change always occurring.
  Don't do this for CALL/IN_COMMUNICATION as the original
  logic intended.

Change-Id: Ib647cec084f4af9a044f90935dd57257bba0ec06

hal: Fix use of channel mask APIs

2015-02-09T00:12:14+00:00

Change-Id: I352f05da08dd453174dc7c8d1d0b98f15bf33ed0

hal: Allow building hardware audio encoders

2015-02-05T21:25:50+00:00

 * TARGET_USES_QCOM_MM_AUDIO flag is not used in CM.

Change-Id: I9171b5fe6f1163d407284217cd8f33ffa3291e16

hal: Fix glitch in audio playback during device switch

2015-02-04T19:05:13+00:00

Currently the deep buffer path uses 4 * 20msec buffer.
During device switch the AudioPolicyManager delays routing
command by 2 * output latency (80msec) to ensure that all
the data written to driver/DSP is played out on current
device itself.
The stream side buffering in the DSP in legacy PCM mode is
95msec and hence the depth of pipeline is 80 + 95 = 175msec
which exceeds the device switch delay 160msec (2 * 80). So
the tail (data written to driver before headset plugin) is
heard on headset and perceived as glitch.
Ensure that the buffering in the kernel is greater than or
equal to the buffering in the DSP to fix the issue.

Change-Id: I01a3862d63ce4c258056620693dee08761c7e83f
CRs-Fixed: 771446

msm8974 hal: Remove platform and baseband check for CSD init.

2015-01-23T17:03:22+00:00

  This is already ifdef'd and the apq8084 uses an external
  modem, so no need for the check.
  Also some variants use mdm2 which prohibits csd init.

Change-Id: Ic7fa66b5f82367a08b29c0249eac7c2595f22d1a

Revert "hal: Fix setting of backend samplerate"

2015-01-21T23:37:09+00:00

 * Strange things are afoot at the circle K.
 * This should be fixed at the codec level as the issue it's
   trying to address is with decimation.

This reverts commit 703bd572fd9824d0b9cd9511dc783503a0ed0e04.

Change-Id: I9d9b7bcfa504635325ef7ccc7afe699ba585b553