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| author | Steve Kondik <steve@cyngn.com> | 2015-12-07 18:45:35 -0800 |
|---|---|---|
| committer | Steve Kondik <steve@cyngn.com> | 2015-12-07 18:45:35 -0800 |
| commit | 697a7ab8c538ca84a92b605e50824da1cbf9b466 (patch) | |
| tree | fe32b5c0b8e0fa115cd6358c5428b8a48e285dec /services/audioflinger/Threads.cpp | |
| parent | 7f8512331f105565054202486fa0a6094ad693c1 (diff) | |
| parent | 5dc9ffe50ef517591b8ffad66c7e4d6ec82b8b4b (diff) | |
| download | frameworks_av-697a7ab8c538ca84a92b605e50824da1cbf9b466.zip frameworks_av-697a7ab8c538ca84a92b605e50824da1cbf9b466.tar.gz frameworks_av-697a7ab8c538ca84a92b605e50824da1cbf9b466.tar.bz2 | |
Merge tag 'android-6.0.1_r3' of https://android.googlesource.com/platform/frameworks/av into cm-13.0
Android 6.0.1 release 3
Change-Id: I2f2a1fe1b58c828e8341556996211562d6e195ab
Diffstat (limited to 'services/audioflinger/Threads.cpp')
| -rw-r--r-- | services/audioflinger/Threads.cpp | 5 |
1 files changed, 4 insertions, 1 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 680ff13..e80221e 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -6962,6 +6962,7 @@ void AudioFlinger::RecordThread::readInputParameters_l() mRsmpInFrames = mFrameCount * 7; mRsmpInFramesP2 = roundup(mRsmpInFrames); free(mRsmpInBuffer); + mRsmpInBuffer = NULL; // TODO optimize audio capture buffer sizes ... // Here we calculate the size of the sliding buffer used as a source @@ -6971,7 +6972,9 @@ void AudioFlinger::RecordThread::readInputParameters_l() // The current value is higher than necessary. However it should not add to latency. // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer - (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); + size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; + (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); + memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? |