diff options
| -rw-r--r-- | services/audioflinger/FastCapture.cpp | 4 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 5 |
2 files changed, 7 insertions, 2 deletions
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp index 79ac12b..1bba5f6 100644 --- a/services/audioflinger/FastCapture.cpp +++ b/services/audioflinger/FastCapture.cpp @@ -131,7 +131,9 @@ void FastCapture::onStateChange() // FIXME new may block for unbounded time at internal mutex of the heap // implementation; it would be better to have normal capture thread allocate for // us to avoid blocking here and to prevent possible priority inversion - (void)posix_memalign(&mReadBuffer, 32, frameCount * Format_frameSize(mFormat)); + size_t bufferSize = frameCount * Format_frameSize(mFormat); + (void)posix_memalign(&mReadBuffer, 32, bufferSize); + memset(mReadBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 0a7d4a2..246f6ba 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -6926,6 +6926,7 @@ 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 @@ -6935,7 +6936,9 @@ 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? |