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Commit 4fa0e6cc authored by Ram Mohan's avatar Ram Mohan Committed by Wonsik Kim
Browse files

codec2: Fix issues in Gapless playback C2SoftMP3 

While the decoder delay samples are removed from output, adjust timestamp
and buffer size informations accordingly

Bug: 74770949
Test: setprop debug.stagefright.ccodec true
Test: stagefright -ao -N c2.google.mp3.decoder /sdcard/clips/bbb.mp3

Change-Id: Iac2a9eb119762f9ef45a53c1ff53ddd46adcff89
parent 0a458539

media/libstagefright/codecs/mp3dec/C2SoftMP3.cpp

+23 −17
Original line number Diff line number Diff line
@@ -32,7 +32,7 @@ 


namespace android {



constexpr char kComponentName[] = "c2.google.aac.encoder";

constexpr char kComponentName[] = "c2.google.mp3.decoder";



static std::shared_ptr<C2ComponentInterface> BuildIntf(

        const char *name, c2_node_id_t id,
@@ -68,6 +68,8 @@ c2_status_t C2SoftMP3::onStop() { 
    mSignalledError = false;

    mIsFirst = true;

    mSignalledOutputEos = false;

    mAnchorTimeStamp = 0;

    mProcessedSamples = 0;



    return C2_OK;

}
@@ -105,6 +107,8 @@ status_t C2SoftMP3::initDecoder() { 
    mIsFirst = true;

    mSignalledError = false;

    mSignalledOutputEos = false;

    mAnchorTimeStamp = 0;

    mProcessedSamples = 0;



    return OK;

}
@@ -269,11 +273,6 @@ c2_status_t C2SoftMP3::drain( 


// TODO: Can overall error checking be improved? As in the check for validity of

//       work, pool ptr, work->input.buffers.size() == 1, ...

// TODO: Gapless playback: decoder has a delay of 529 samples. For the first

//       frame we intend to remove 529 samples worth of data. When this is

//       done it is going to effect the timestamps of future frames. This

//       timestamp correction is handled by the client or plugin? Soft omx mp3

//       plugin also has this problem

// TODO: Blind removal of 529 samples from the output may not work. Because

//       mpeg layer 1 frame size is 384 samples per frame. This should introduce

//       negative values and can cause SEG faults. Soft omx mp3 plugin can have
@@ -292,10 +291,10 @@ void C2SoftMP3::process( 
    bool eos = ((work->input.flags & C2FrameData::FLAG_END_OF_STREAM) != 0);

    size_t inOffset = inBuffer.offset();

    size_t inSize = inBuffer.size();

    C2ReadView rView = inBuffer.map().get();

    C2ReadView rView = work->input.buffers[0]->data().linearBlocks().front().map().get();

    if (inSize && rView.error()) {

        ALOGE("read view map failed %d", rView.error());

        work->result = C2_CORRUPTED;

        work->result = rView.error();

        return;

    }
@@ -315,8 +314,8 @@ void C2SoftMP3::process( 


    size_t calOutSize;

    std::vector<size_t> decodedSizes;

    if (OK != calculateOutSize(const_cast<uint8 *>(rView.data() + inOffset),

                               inSize, &decodedSizes)) {

    const uint8_t *inPtr = rView.data() + inOffset;

    if (OK != calculateOutSize(const_cast<uint8 *>(inPtr), inSize, &decodedSizes)) {

        work->result = C2_CORRUPTED;

        return;

    }
@@ -332,21 +331,22 @@ void C2SoftMP3::process( 
    C2WriteView wView = block->map().get();

    if (wView.error()) {

        ALOGE("write view map failed %d", wView.error());

        work->result = C2_CORRUPTED;

        work->result = wView.error();

        return;

    }



    int outSize = 0;

    int outOffset = 0;

    auto it = decodedSizes.begin();

    while (inOffset < inSize) {

    size_t inPos = 0;

    while (inPos < inSize) {

        if (it == decodedSizes.end()) {

            ALOGE("unexpected trailing bytes, ignoring them");

            break;

        }



        mConfig->pInputBuffer = const_cast<uint8 *>(rView.data() + inOffset);

        mConfig->inputBufferCurrentLength = (inSize - inOffset);

        mConfig->pInputBuffer = const_cast<uint8 *>(inPtr + inPos);

        mConfig->inputBufferCurrentLength = (inSize - inPos);

        mConfig->inputBufferMaxLength = 0;

        mConfig->inputBufferUsedLength = 0;

        mConfig->outputFrameSize = (calOutSize - outSize);
@@ -382,7 +382,7 @@ void C2SoftMP3::process( 
            return;

        }

        outSize += mConfig->outputFrameSize * sizeof(int16_t);

        inOffset += mConfig->inputBufferUsedLength;

        inPos += mConfig->inputBufferUsedLength;

        it++;

    }

    if (mIsFirst) {
@@ -391,13 +391,19 @@ void C2SoftMP3::process( 
        // the start of the first output buffer. This essentially makes this

        // decoder have zero delay, which the rest of the pipeline assumes.

        outOffset = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);

        mAnchorTimeStamp = work->input.ordinal.timestamp.peekull();

    }

    ALOGV("out buffer attr. offset %d size %d", outOffset, outSize);

    uint64_t outTimeStamp = mProcessedSamples * 1000000ll / mSamplingRate;

    mProcessedSamples += ((outSize - outOffset) / (mNumChannels * sizeof(int16_t)));

    ALOGV("out buffer attr. offset %d size %d timestamp %u", outOffset, outSize - outOffset,

          (uint32_t)(mAnchorTimeStamp + outTimeStamp));

    decodedSizes.clear();

    work->worklets.front()->output.flags = work->input.flags;

    work->worklets.front()->output.buffers.clear();

    work->worklets.front()->output.buffers.push_back(createLinearBuffer(block, outOffset, outSize));

    work->worklets.front()->output.buffers.push_back(

            createLinearBuffer(block, outOffset, outSize - outOffset));

    work->worklets.front()->output.ordinal = work->input.ordinal;

    work->worklets.front()->output.ordinal.timestamp = mAnchorTimeStamp + outTimeStamp;

    work->workletsProcessed = 1u;

    if (eos) {

        mSignalledOutputEos = true;

media/libstagefright/codecs/mp3dec/C2SoftMP3.h

+2 −0
Original line number Diff line number Diff line
@@ -61,6 +61,8 @@ private: 
    bool mIsFirst;

    bool mSignalledError;

    bool mSignalledOutputEos;

    uint64_t mAnchorTimeStamp;

    uint64_t mProcessedSamples;



    status_t initDecoder();