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Commit 745fcc05 authored by James Dong's avatar James Dong
Browse files

Limit the amount of audio record data in each buffer 

o The size of each input buffer should be less than or equal to kMaxBufferSize
o related-to-bug: 5977032

Change-Id: I04343169aac3df56694aad4ba7967ec45337ad7e
parent a9daa164

include/media/stagefright/AudioSource.h

+1 −0
Original line number Diff line number Diff line
@@ -95,6 +95,7 @@ private: 
        int32_t startFrame, int32_t rampDurationFrames,

        uint8_t *data,   size_t bytes);



    void queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs);

    void releaseQueuedFrames_l();

    void waitOutstandingEncodingFrames_l();

    status_t reset();

media/libstagefright/AudioSource.cpp

+38 −19
Original line number Diff line number Diff line
@@ -282,8 +282,6 @@ status_t AudioSource::dataCallbackTimestamp( 
        mPrevSampleTimeUs = mStartTimeUs;

    }



    int64_t timestampUs = mPrevSampleTimeUs;



    size_t numLostBytes = 0;

    if (mNumFramesReceived > 0) {  // Ignore earlier frame lost

        // getInputFramesLost() returns the number of lost frames.
@@ -293,37 +291,58 @@ status_t AudioSource::dataCallbackTimestamp( 


    CHECK_EQ(numLostBytes & 1, 0u);

    CHECK_EQ(audioBuffer.size & 1, 0u);

    size_t bufferSize = numLostBytes + audioBuffer.size;

    MediaBuffer *buffer = new MediaBuffer(bufferSize);

    if (numLostBytes > 0) {

        memset(buffer->data(), 0, numLostBytes);

        memcpy((uint8_t *) buffer->data() + numLostBytes,

                    audioBuffer.i16, audioBuffer.size);

        // Loss of audio frames should happen rarely; thus the LOGW should

        // not cause a logging spam

        ALOGW("Lost audio record data: %d bytes", numLostBytes);

    }



    while (numLostBytes > 0) {

        size_t bufferSize = numLostBytes;

        if (numLostBytes > kMaxBufferSize) {

            numLostBytes -= kMaxBufferSize;

            bufferSize = kMaxBufferSize;

        } else {

            numLostBytes = 0;

        }

        MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);

        memset(lostAudioBuffer->data(), 0, bufferSize);

        lostAudioBuffer->set_range(0, bufferSize);

        queueInputBuffer_l(lostAudioBuffer, timeUs);

    }



    if (audioBuffer.size == 0) {

        ALOGW("Nothing is available from AudioRecord callback buffer");

            buffer->release();

        return OK;

    }



    const size_t bufferSize = audioBuffer.size;

    MediaBuffer *buffer = new MediaBuffer(bufferSize);

    memcpy((uint8_t *) buffer->data(),

            audioBuffer.i16, audioBuffer.size);

    buffer->set_range(0, bufferSize);

    queueInputBuffer_l(buffer, timeUs);

    return OK;

}



    buffer->set_range(0, bufferSize);

    timestampUs += ((1000000LL * (bufferSize >> 1)) +

void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {

    const size_t bufferSize = buffer->range_length();

    const size_t frameSize = mRecord->frameSize();

    const int64_t timestampUs =

                mPrevSampleTimeUs +

                    ((1000000LL * (bufferSize / frameSize)) +

                        (mSampleRate >> 1)) / mSampleRate;



    if (mNumFramesReceived == 0) {

        buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);

    }



    buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);

    buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);

    mPrevSampleTimeUs = timestampUs;

    mNumFramesReceived += buffer->range_length() / sizeof(int16_t);

    mNumFramesReceived += bufferSize / frameSize;

    mBuffersReceived.push_back(buffer);

    mFrameAvailableCondition.signal();



    return OK;

}



void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {