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Commit ab5cdbaf authored by Eric Laurent's avatar Eric Laurent
Browse files

AudioTrack: add support for compressed audio 

Add support for compressed audio playback by use
of an AudioTrack attached to a direct output thread.

Bug: 9428304.

Change-Id: I4a61be9cf0e31003ca85935d6e0ee38ca8192e03
parent 29a8b2fe

include/media/AudioTrack.h

+9 −0
Original line number Diff line number Diff line
@@ -632,6 +632,8 @@ protected: 
            nsecs_t processAudioBuffer();



            bool     isOffloaded() const;

            bool     isDirect() const;

            bool     isOffloadedOrDirect() const;



            // caller must hold lock on mLock for all _l methods
@@ -648,6 +650,13 @@ protected: 
            bool     isOffloaded_l() const

                { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }



            bool     isOffloadedOrDirect_l() const

                { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|

                                                AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }



            bool     isDirect_l() const

                { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }



    // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0

    sp<IAudioTrack>         mAudioTrack;

    sp<IMemory>             mCblkMemory;

include/media/IAudioFlinger.h

+1 −0
Original line number Diff line number Diff line
@@ -50,6 +50,7 @@ public: 
        TRACK_TIMED   = 1,  // client requests a TimedAudioTrack

        TRACK_FAST    = 2,  // client requests a fast AudioTrack or AudioRecord

        TRACK_OFFLOAD = 4,  // client requests offload to hw codec

        TRACK_DIRECT = 8,   // client requests a direct output

    };

    typedef uint32_t track_flags_t;

media/libmedia/AudioTrack.cpp

+52 −15
Original line number Diff line number Diff line
@@ -635,7 +635,7 @@ void AudioTrack::getAuxEffectSendLevel(float* level) const 


status_t AudioTrack::setSampleRate(uint32_t rate)

{

    if (mIsTimed || isOffloaded()) {

    if (mIsTimed || isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }
@@ -666,7 +666,7 @@ uint32_t AudioTrack::getSampleRate() const 
    // sample rate can be updated during playback by the offloaded decoder so we need to

    // query the HAL and update if needed.

// FIXME use Proxy return channel to update the rate from server and avoid polling here

    if (isOffloaded_l()) {

    if (isOffloadedOrDirect_l()) {

        if (mOutput != AUDIO_IO_HANDLE_NONE) {

            uint32_t sampleRate = 0;

            status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
@@ -680,7 +680,7 @@ uint32_t AudioTrack::getSampleRate() const 


status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)

{

    if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {

    if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }
@@ -714,7 +714,7 @@ void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 
status_t AudioTrack::setMarkerPosition(uint32_t marker)

{

    // The only purpose of setting marker position is to get a callback

    if (mCbf == NULL || isOffloaded()) {

    if (mCbf == NULL || isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }
@@ -727,7 +727,7 @@ status_t AudioTrack::setMarkerPosition(uint32_t marker) 


status_t AudioTrack::getMarkerPosition(uint32_t *marker) const

{

    if (isOffloaded()) {

    if (isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }

    if (marker == NULL) {
@@ -743,7 +743,7 @@ status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)

{

    // The only purpose of setting position update period is to get a callback

    if (mCbf == NULL || isOffloaded()) {

    if (mCbf == NULL || isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }
@@ -756,7 +756,7 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 


status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const

{

    if (isOffloaded()) {

    if (isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }

    if (updatePeriod == NULL) {
@@ -771,7 +771,7 @@ status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 


status_t AudioTrack::setPosition(uint32_t position)

{

    if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {

    if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }

    if (position > mFrameCount) {
@@ -804,10 +804,10 @@ status_t AudioTrack::getPosition(uint32_t *position) const 
    }



    AutoMutex lock(mLock);

    if (isOffloaded_l()) {

    if (isOffloadedOrDirect_l()) {

        uint32_t dspFrames = 0;



        if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) {

        if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {

            ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);

            *position = mPausedPosition;

            return NO_ERROR;
@@ -842,7 +842,7 @@ status_t AudioTrack::getBufferPosition(uint32_t *position) 


status_t AudioTrack::reload()

{

    if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {

    if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {

        return INVALID_OPERATION;

    }
@@ -1038,6 +1038,10 @@ status_t AudioTrack::createTrack_l(size_t epoch) 
        trackFlags |= IAudioFlinger::TRACK_OFFLOAD;

    }



    if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {

        trackFlags |= IAudioFlinger::TRACK_DIRECT;

    }



    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,

                                // but we will still need the original value also

    sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
@@ -1129,6 +1133,16 @@ status_t AudioTrack::createTrack_l(size_t epoch) 
            //return NO_INIT;

        }

    }

    if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {

        if (trackFlags & IAudioFlinger::TRACK_DIRECT) {

            ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful");

        } else {

            ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server");

            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT);

            // FIXME This is a warning, not an error, so don't return error status

            //return NO_INIT;

        }

    }



    // We retain a copy of the I/O handle, but don't own the reference

    mOutput = output;
@@ -1324,6 +1338,16 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 
        return INVALID_OPERATION;

    }



    if (isDirect()) {

        AutoMutex lock(mLock);

        int32_t flags = android_atomic_and(

                            ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),

                            &mCblk->mFlags);

        if (flags & CBLK_INVALID) {

            return DEAD_OBJECT;

        }

    }



    if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {

        // Sanity-check: user is most-likely passing an error code, and it would

        // make the return value ambiguous (actualSize vs error).
@@ -1472,7 +1496,7 @@ nsecs_t AudioTrack::processAudioBuffer() 
        // for offloaded tracks restoreTrack_l() will just update the sequence and clear

        // AudioSystem cache. We should not exit here but after calling the callback so

        // that the upper layers can recreate the track

        if (!isOffloaded_l() || (mSequence == mObservedSequence)) {

        if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {

            status_t status = restoreTrack_l("processAudioBuffer");

            mLock.unlock();

            // Run again immediately, but with a new IAudioTrack
@@ -1598,7 +1622,7 @@ nsecs_t AudioTrack::processAudioBuffer() 
        mObservedSequence = sequence;

        mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);

        // for offloaded tracks, just wait for the upper layers to recreate the track

        if (isOffloaded()) {

        if (isOffloadedOrDirect()) {

            return NS_INACTIVE;

        }

    }
@@ -1756,7 +1780,7 @@ nsecs_t AudioTrack::processAudioBuffer() 
status_t AudioTrack::restoreTrack_l(const char *from)

{

    ALOGW("dead IAudioTrack, %s, creating a new one from %s()",

          isOffloaded_l() ? "Offloaded" : "PCM", from);

          isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);

    ++mSequence;

    status_t result;
@@ -1764,7 +1788,7 @@ status_t AudioTrack::restoreTrack_l(const char *from) 
    // output parameters in createTrack_l()

    AudioSystem::clearAudioConfigCache();



    if (isOffloaded_l()) {

    if (isOffloadedOrDirect_l()) {

        // FIXME re-creation of offloaded tracks is not yet implemented

        return DEAD_OBJECT;

    }
@@ -1850,6 +1874,19 @@ bool AudioTrack::isOffloaded() const 
    return isOffloaded_l();

}



bool AudioTrack::isDirect() const

{

    AutoMutex lock(mLock);

    return isDirect_l();

}



bool AudioTrack::isOffloadedOrDirect() const

{

    AutoMutex lock(mLock);

    return isOffloadedOrDirect_l();

}





status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const

{

services/audioflinger/PlaybackTracks.h

+1 −0
Original line number Diff line number Diff line
@@ -54,6 +54,7 @@ public: 
                return mStreamType;

            }

            bool        isOffloaded() const { return (mFlags & IAudioFlinger::TRACK_OFFLOAD) != 0; }

            bool        isDirect() const { return (mFlags & IAudioFlinger::TRACK_DIRECT) != 0; }

            status_t    setParameters(const String8& keyValuePairs);

            status_t    attachAuxEffect(int EffectId);

            void        setAuxBuffer(int EffectId, int32_t *buffer);

services/audioflinger/Threads.cpp

+25 −12
Original line number Diff line number Diff line
@@ -1660,7 +1660,7 @@ bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 
    track->mState = TrackBase::STOPPED;

    if (!trackActive) {

        removeTrack_l(track);

    } else if (track->isFastTrack() || track->isOffloaded()) {

    } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {

        track->mState = TrackBase::STOPPING_1;

    }
@@ -1868,7 +1868,9 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l() 
    }

    mNormalFrameCount = multiplier * mFrameCount;

    // round up to nearest 16 frames to satisfy AudioMixer

    if (mType == MIXER || mType == DUPLICATING) {

        mNormalFrameCount = (mNormalFrameCount + 15) & ~15;

    }

    ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,

            mNormalFrameCount);
@@ -2656,7 +2658,7 @@ status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 
    if (mNormalSink != 0) {

        return mNormalSink->getTimestamp(timestamp);

    }

    if (mType == OFFLOAD && mOutput->stream->get_presentation_position) {

    if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) {

        uint64_t position64;

        int ret = mOutput->stream->get_presentation_position(

                                                mOutput->stream, &position64, &timestamp.mTime);
@@ -3947,14 +3949,16 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep 
        // The first time a track is added we wait

        // for all its buffers to be filled before processing it

        uint32_t minFrames;

        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {

        if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) {

            minFrames = mNormalFrameCount;

        } else {

            minFrames = 1;

        }



        if ((track->framesReady() >= minFrames) && track->isReady() &&

                !track->isPaused() && !track->isTerminated())

        ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ",

              minFrames, track->mState, track->framesReady());

        if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&

                !track->isStopping_2() && !track->isStopped())

        {

            ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -3981,17 +3985,26 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep 
            if (!mEffectChains.isEmpty() && last) {

                mEffectChains[0]->clearInputBuffer();

            }



            ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer);

            if ((track->sharedBuffer() != 0) || track->isTerminated() ||

                    track->isStopped() || track->isPaused()) {

            if (track->isStopping_1()) {

                track->mState = TrackBase::STOPPING_2;

            }

            if ((track->sharedBuffer() != 0) || track->isStopped() ||

                    track->isStopping_2() || track->isPaused()) {

                // We have consumed all the buffers of this track.

                // Remove it from the list of active tracks.

                // TODO: implement behavior for compressed audio

                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;

                size_t audioHALFrames;

                if (audio_is_linear_pcm(mFormat)) {

                    audioHALFrames = (latency_l() * mSampleRate) / 1000;

                } else {

                    audioHALFrames = 0;

                }



                size_t framesWritten = mBytesWritten / mFrameSize;

                if (mStandby || !last ||

                        track->presentationComplete(framesWritten, audioHALFrames)) {

                    if (track->isStopping_2()) {

                        track->mState = TrackBase::STOPPED;

                    }

                    if (track->isStopped()) {

                        track->reset();

                    }