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Commit 55d1ea47 authored by Glenn Kasten's avatar Glenn Kasten Committed by Android (Google) Code Review
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Merge "DO NOT MERGE - AudioFlinger: Clear record buffers when starting RecordThread" into lmp-dev

parents e5326d96 8c987fa7

services/audioflinger/FastCapture.cpp

+1 −0
Original line number Original line Diff line number Diff line
@@ -134,6 +134,7 @@ void FastCapture::onStateChange() 
            unsigned channelCount = Format_channelCount(format);

            unsigned channelCount = Format_channelCount(format);

            // FIXME frameSize

            // FIXME frameSize

            readBuffer = new short[frameCount * channelCount];

            readBuffer = new short[frameCount * channelCount];

            memset(readBuffer, 0, frameCount * channelCount * sizeof(readBuffer[0]));

            periodNs = (frameCount * 1000000000LL) / sampleRate;    // 1.00

            periodNs = (frameCount * 1000000000LL) / sampleRate;    // 1.00

            underrunNs = (frameCount * 1750000000LL) / sampleRate;  // 1.75

            underrunNs = (frameCount * 1750000000LL) / sampleRate;  // 1.75

            overrunNs = (frameCount * 500000000LL) / sampleRate;    // 0.50

            overrunNs = (frameCount * 500000000LL) / sampleRate;    // 0.50

services/audioflinger/Threads.cpp

+3 −1
Original line number Original line Diff line number Diff line
@@ -6136,7 +6136,9 @@ void AudioFlinger::RecordThread::readInputParameters_l() 
    // The current value is higher than necessary.  However it should not add to latency.

    // 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

    // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer

    mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];

    size_t bufferSizeInShorts = (mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount;

    mRsmpInBuffer = new int16_t[bufferSizeInShorts];

    memset(mRsmpInBuffer, 0, bufferSizeInShorts * sizeof(mRsmpInBuffer[0]));





    // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.

    // 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?

    // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?