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Commit 403bff76 authored by Andy Hung's avatar Andy Hung Committed by Android Git Automerger
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am 4c6e77ff: AudioFlinger: Clear record buffers when starting RecordThread 

* commit '4c6e77ff':
  AudioFlinger: Clear record buffers when starting RecordThread
parents 036b6593 4c6e77ff

services/audioflinger/FastCapture.cpp

+3 −1
Original line number Diff line number Diff line
@@ -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

services/audioflinger/Threads.cpp

+4 −1
Original line number Diff line number Diff line
@@ -6932,6 +6932,7 @@ 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
@@ -6941,7 +6942,9 @@ 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?