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Commit ce8828a0 authored by Glenn Kasten's avatar Glenn Kasten
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Fix underruns when fast track denied due to SRC 

OpenSL ES requests a fast track.  If sample rate conversion is needed,
the request is denied by server, and a larger client buffer is used
to handle the higher latency of a normal track.  However the client
notification period was calculated based on buffer being divided into
2 sub-buffers.  That resulted in the notification period being too long.
The server pulls chunks that are smaller than half the total buffer.
So now the client uses 3 sub-buffers when there is SRC.

Also removed the 'defer wake' optimization because it was incorrect.
This optimization attempted to reduce the number of wakeups of client,
when server releaseBuffer knows that another releaseBuffer will be
following.  But there is no way for the first releaseBuffer to predict
how soon the second releaseBuffer will occur.  In some cases it was
a long time, and the client underran.  So now the client is woken up
immediately if the total number of available frames to client is >=
the minimum number the client wants to see (the notification period).

Also fix bug where minimum frame count was not being used in the
calculation of notification period.

Bug: 10342804
Change-Id: I3c246f4e7bc3684a344f2cf08268dc082e338e2a
parent d176ec1e

include/private/media/AudioTrackShared.h

+0 −2
Original line number Diff line number Diff line
@@ -383,8 +383,6 @@ public: 
protected:

    size_t      mAvailToClient; // estimated frames available to client prior to releaseBuffer()

    int32_t     mFlush;         // our copy of cblk->u.mStreaming.mFlush, for streaming output only

private:

    bool        mDeferWake;     // whether another releaseBuffer() is expected soon

};



// Proxy used by AudioFlinger for servicing AudioTrack

media/libmedia/AudioTrack.cpp

+35 −32
Original line number Diff line number Diff line
@@ -815,12 +815,29 @@ status_t AudioTrack::createTrack_l( 
        return NO_INIT;

    }



    // Not all of these values are needed under all conditions, but it is easier to get them all



    uint32_t afLatency;

    if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) {

    status = AudioSystem::getLatency(output, streamType, &afLatency);

    if (status != NO_ERROR) {

        ALOGE("getLatency(%d) failed status %d", output, status);

        return NO_INIT;

    }



    size_t afFrameCount;

    status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);

    if (status != NO_ERROR) {

        ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);

        return NO_INIT;

    }



    uint32_t afSampleRate;

    status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);

    if (status != NO_ERROR) {

        ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);

        return NO_INIT;

    }



    // Client decides whether the track is TIMED (see below), but can only express a preference

    // for FAST.  Server will perform additional tests.

    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
@@ -836,6 +853,14 @@ status_t AudioTrack::createTrack_l( 
    }

    ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);



    // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where

    //  n = 1   fast track; nBuffering is ignored

    //  n = 2   normal track, no sample rate conversion

    //  n = 3   normal track, with sample rate conversion

    //          (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)

    //  n > 3   very high latency or very small notification interval; nBuffering is ignored

    const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;



    mNotificationFramesAct = mNotificationFramesReq;



    if (!audio_is_linear_pcm(format)) {
@@ -844,13 +869,6 @@ status_t AudioTrack::createTrack_l( 
            // Same comment as below about ignoring frameCount parameter for set()

            frameCount = sharedBuffer->size();

        } else if (frameCount == 0) {

            size_t afFrameCount;

            status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);

            if (status != NO_ERROR) {

                ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType,

                        status);

                return NO_INIT;

            }

            frameCount = afFrameCount;

        }

        if (mNotificationFramesAct != frameCount) {
@@ -880,26 +898,13 @@ status_t AudioTrack::createTrack_l( 
    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {



        // FIXME move these calculations and associated checks to server

        uint32_t afSampleRate;

        status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);

        if (status != NO_ERROR) {

            ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType,

                    status);

            return NO_INIT;

        }

        size_t afFrameCount;

        status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);

        if (status != NO_ERROR) {

            ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);

            return NO_INIT;

        }



        // Ensure that buffer depth covers at least audio hardware latency

        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);

        ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",

                afFrameCount, minBufCount, afSampleRate, afLatency);

        if (minBufCount <= 2) {

            minBufCount = sampleRate == afSampleRate ? 2 : 3;

        if (minBufCount <= nBuffering) {

            minBufCount = nBuffering;

        }



        size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
@@ -909,18 +914,16 @@ status_t AudioTrack::createTrack_l( 


        if (frameCount == 0) {

            frameCount = minFrameCount;

        }

        // Make sure that application is notified with sufficient margin

        // before underrun

        if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {

            mNotificationFramesAct = frameCount/2;

        }

        if (frameCount < minFrameCount) {

        } else if (frameCount < minFrameCount) {

            // not ALOGW because it happens all the time when playing key clicks over A2DP

            ALOGV("Minimum buffer size corrected from %d to %d",

                     frameCount, minFrameCount);

            frameCount = minFrameCount;

        }

        // Make sure that application is notified with sufficient margin before underrun

        if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {

            mNotificationFramesAct = frameCount/nBuffering;

        }



    } else {

        // For fast tracks, the frame count calculations and checks are done by server
@@ -1001,8 +1004,8 @@ status_t AudioTrack::createTrack_l( 
            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);

            mFlags = flags;

            if (sharedBuffer == 0) {

                if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {

                    mNotificationFramesAct = frameCount/2;

                if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {

                    mNotificationFramesAct = frameCount/nBuffering;

                }

            }

        }

media/libmedia/AudioTrackShared.cpp

+2 −5
Original line number Diff line number Diff line
@@ -481,7 +481,7 @@ size_t StaticAudioTrackClientProxy::getBufferPosition() 
ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,

        size_t frameSize, bool isOut, bool clientInServer)

    : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer),

      mAvailToClient(0), mFlush(0), mDeferWake(false)

      mAvailToClient(0), mFlush(0)

{

}
@@ -559,9 +559,6 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer) 
            &((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL;

    buffer->mNonContig = availToServer - part1;

    mUnreleased = part1;

    // optimization to avoid waking up the client too early

    // FIXME need to test for recording

    mDeferWake = part1 < ask && availToServer >= ask;

    return part1 > 0 ? NO_ERROR : WOULD_BLOCK;

    }

no_init:
@@ -607,7 +604,7 @@ void ServerProxy::releaseBuffer(Buffer* buffer) 
        minimum = half;

    }

    // FIXME AudioRecord wakeup needs to be optimized; it currently wakes up client every time

    if (!mIsOut || (!mDeferWake && mAvailToClient + stepCount >= minimum)) {

    if (!mIsOut || (mAvailToClient + stepCount >= minimum)) {

        ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum);

        int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);

        if (!(old & CBLK_FUTEX_WAKE)) {