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Commit e8426148 authored by Glenn Kasten's avatar Glenn Kasten
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Use larger capture pipe since we support resampling by 6:1 

This avoids overruns on the client AudioRecord side,
without requiring client to use a large buffer.
It should not increase input latency, since a newly started AudioRecord
always joins the stream starting at the latest data.

Change-Id: Ib2b8de75cc40a6a3d493a1f8b46b41220f69264f
parent b4429211

services/audioflinger/Threads.cpp

+3 −3
Original line number Diff line number Diff line
@@ -5555,12 +5555,12 @@ void AudioFlinger::RecordThread::readInputParameters_l() 
    mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);

    mFrameCount = mBufferSize / mFrameSize;

    // This is the formula for calculating the temporary buffer size.

    // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to

    // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to

    // 1 full output buffer, regardless of the alignment of the available input.

    // The "3" is somewhat arbitrary, and could probably be larger.

    // The value is somewhat arbitrary, and could probably be even larger.

    // A larger value should allow more old data to be read after a track calls start(),

    // without increasing latency.

    mRsmpInFrames = mFrameCount * 3;

    mRsmpInFrames = mFrameCount * 7;

    mRsmpInFramesP2 = roundup(mRsmpInFrames);

    delete[] mRsmpInBuffer;

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