Android Audio 的播放

Android Audio 的播放

Android OMX

Android Audio 的播放

AwesomePlayer 中 媒体流buffer 中的传递

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前面已经分析了Video 的播放，现在看下Audio 的播放过程。Audio decode 后的数据是送给AudioFlinger处理， 由AudioFlinger 和硬件打交道。

一 prepare 创建AudiO OMXCodec

1. status_t AwesomePlayer:: prepare() 的prepare 函数会调用到prepareAsync_l， prepareAsync_l 向TimedEventQueue post 了AwesomePlayer:: onPrepareAsyncEvent 事件
2. 在AwesomePlayer:: onPrepareAsyncEvent 事件中调用 initAudioDecoder 完成了Audio 的初始化。
3. initAudioDecoder 中创建OMXCodec mAudioSource

status_t AwesomePlayer::prepareAsync_l() {

    mAsyncPrepareEvent = new AwesomeEvent(
            this, &AwesomePlayer::onPrepareAsyncEvent);

    mQueue.postEvent(mAsyncPrepareEvent);

    return OK;
}

void AwesomePlayer::onPrepareAsyncEvent() {
    Mutex::Autolock autoLock(mLock);

    if (mAudioTrack != NULL && mAudioSource == NULL) {
        status_t err = initAudioDecoder();

        if (err != OK) {
            abortPrepare(err);
            return;
        }
    }
}

status_t AwesomePlayer::initAudioDecoder() {

    if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW)) {
        mAudioSource = mAudioTrack;
    } else {
        mAudioSource = OMXCodec::Create(
                mClient.interface(), mAudioTrack->getFormat(),
                false, // createEncoder
                mAudioTrack);
    }

    if (mAudioSource != NULL) {
        status_t err = mAudioSource->start();

        if (err != OK) {
            mAudioSource.clear();
            return err;
        }
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_QCELP)) {
        // For legacy reasons we're simply going to ignore the absence
        // of an audio decoder for QCELP instead of aborting playback
        // altogether.
        return OK;
    }

    return mAudioSource != NULL ? OK : UNKNOWN_ERROR;
}

二 播放

AwesomePlayer::play() 函数调用play_l。在play_l 中 new 了一个AudioPlayer。这个好像和video的流程不同。 这里还有一个mAudioSink变量。

status_t AwesomePlayer::play_l() {

    if (mAudioSource != NULL) {
        if (mAudioPlayer == NULL) {
            if (mAudioSink != NULL) {
                mAudioPlayer = new AudioPlayer(mAudioSink, allowDeepBuffering, this);
                mAudioPlayer->setSource(mAudioSource);
            }
        }

        if (mVideoSource == NULL) {
            status_t err = startAudioPlayer_l(false /* sendErrorNotification */);
        }
    }
    return OK;
}

AudioSink

AudioSink 是什么， 来自哪里。首先看AudioSink 来自哪里，在MediaPlayerService 创建完player后，根据hardwareOutput的结果设置 setAudioSink。这里hardwareOutput 为虚函数，默认结果为false。new了一个AudioOutput 。

sp<MediaPlayerBase> MediaPlayerService::Client::setDataSource_pre(
        player_type playerType)
{
    sp<MediaPlayerBase> p = createPlayer(playerType);
    if (p == NULL) {
        return p;
    }

    if (!p->hardwareOutput()) {
        mAudioOutput = new AudioOutput(mAudioSessionId);
        static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
    }

    return p;
}

AudioOutput 定义在MediaPlayerService.h 中，继承了AudioSink。AudioSink为一个纯虚函数，具体实现在AudioOutput中。AudioOutput的代码比较简单大部分是调用mTrack的实现功能。mTrack为AudioTrack 类。AudioOutput 其实就是对AudioTrack的一个简单封装。

class MediaPlayerService : public BnMediaPlayerService
{
    class AudioOutput : public MediaPlayerBase::AudioSink
    {
        class CallbackData {
        public:
            CallbackData(AudioOutput *cookie) {
                mData = cookie;
                mSwitching = false;
            }
        }
    }; // AudioOutput
}

ssize_t MediaPlayerService::AudioOutput::frameCount() const
{
    if (mTrack == 0) return NO_INIT;
    return mTrack->frameCount();
}

ssize_t MediaPlayerService::AudioOutput::channelCount() const
{
    if (mTrack == 0) return NO_INIT;
    return mTrack->channelCount();
}

AudioPlayer

AudioPlayer 中的成员变量中主要有AudioTrack sp mSource 和 MediaBuffer *mInputBuffer。
是对音频播放的一个封装。

mAudioSource = OMXCodec::Create(
        mClient.interface(), mAudioTrack->getFormat(),
        false, // createEncoder
        mAudioTrack);

startAudioPlayer_l

在startAudioPlayer_l 中调用AudioPlayer 的start函数。这里走mAudioSink->open 。在open 中new了 AudioTrack出来。
需要注意的是在open 中传递的 AudioPlayer::AudioSinkCallback 回调函数。然后调用start 开始播放。

status_t AudioPlayer::start(bool sourceAlreadyStarted) {
    mFirstBufferResult = mSource->read(&mFirstBuffer, &options);
    if (mFirstBufferResult == INFO_FORMAT_CHANGED) {
        mFirstBufferResult = OK;
        mIsFirstBuffer = false;
    } else {
        mIsFirstBuffer = true;
    }
  
    if (mAudioSink.get() != NULL) {
        status_t err = mAudioSink->open(
                mSampleRate, numChannels, channelMask, AUDIO_FORMAT_PCM_16_BIT,
                DEFAULT_AUDIOSINK_BUFFERCOUNT,
                &AudioPlayer::AudioSinkCallback,
                this,
                (mAllowDeepBuffering ?
                            AUDIO_OUTPUT_FLAG_DEEP_BUFFER :
                            AUDIO_OUTPUT_FLAG_NONE));
            return err;
        }

        mAudioSink->start();
    } else {
        mAudioTrack = new AudioTrack(
                AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
                0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);

        mAudioTrack->start();
    }

    return OK;
}

status_t MediaPlayerService::AudioOutput::open(
        uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
        audio_format_t format, int bufferCount,
        AudioCallback cb, void *cookie,
        audio_output_flags_t flags)
{
    mCallback = cb;

    if (mCallback != NULL) {
        newcbd = new CallbackData(this);
        t = new AudioTrack(
                mStreamType,
                sampleRate,
                format,
                channelMask,
                frameCount,
                flags,
                CallbackWrapper,
                newcbd,
                0,  // notification frames
                mSessionId);
    } else {
        t = new AudioTrack(
                mStreamType,
                sampleRate,
                format,
                channelMask,
                frameCount,
                flags,
                NULL,
                NULL,
                0,
                mSessionId);
    }
}


在Video 的播放中，是启动了一个线程不停的循环读取数据，可是在AudioTrack 的start 中没有发现这样的逻辑。 

回过头来看下AudioTrack的构造函数，在构造函数中调用了set ，set 中赋值回调函数，我们在上边提到的，然后new了一个AudioTrackThread。原来线程在这里。在AudioTrackThread 的Loop 中主要做了三件事：

1. obtainBuffer 获取空闲的buffer。
2. mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer)； 通知AudioPlayer 填充buffer
3. releaseBuffer(&audioBuffer); 释放已经填充数据的buffer 。AudioFlinger 可以读这部分数据了。

status_t AudioTrack::set(
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        int channelMask,
        int frameCount,
        audio_output_flags_t flags,
        callback_t cbf,
        void* user,
        int notificationFrames,
        const sp<IMemory>& sharedBuffer,
        bool threadCanCallJava,
        int sessionId)
{
    mCbf = cbf;
    
    if (cbf != NULL) {
        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
    }

    // create the IAudioTrack
    status_t status = createTrack_l(streamType,
                                  sampleRate,
                                  format,
                                  (uint32_t)channelMask,
                                  frameCount,
                                  flags,
                                  sharedBuffer,
                                  output);


    return NO_ERROR;
}

bool AudioTrack::AudioTrackThread::threadLoop()
{
    {
        AutoMutex _l(mMyLock);
        if (mPaused) {
            mMyCond.wait(mMyLock);
            // caller will check for exitPending()
            return true;
        }
    }
    if (!mReceiver.processAudioBuffer(this)) {
        pause();
    }
    return true;
}

bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
    Buffer audioBuffer;

    do {

        status_t err = obtainBuffer(&audioBuffer, waitCount);
    
        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer)；

        releaseBuffer(&audioBuffer);
    }
    while (frames);
    
    return true;
}

callback

mCbf 根据传递的参数类型为CallbackWrapper， 为静态函数。调用了mCallback。在上面AudiOOUtput open的时候提到过这个回调。类型为：AudioPlayer::AudioSinkCallback。在AudioSinkCallback中调用OMXcode的read函数读出解码后的数据，然后copy 到callback 传递的buffer中，完成数据的copy。

static void MediaPlayerService::AudioOutput::CallbackWrapper(
        int event, void *cookie, void *info) {
    //ALOGV("callbackwrapper");
    if (event != AudioTrack::EVENT_MORE_DATA) {
        return;
    }

    CallbackData *data = (CallbackData*)cookie;
    data->lock();
    AudioOutput *me = data->getOutput();
    AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;

    size_t actualSize = (*me->mCallback)(
            me, buffer->raw, buffer->size, me->mCallbackCookie);
}

// static
size_t AudioPlayer::AudioSinkCallback(
        MediaPlayerBase::AudioSink *audioSink,
        void *buffer, size_t size, void *cookie) {
    AudioPlayer *me = (AudioPlayer *)cookie;

    return me->fillBuffer(buffer, size);
}

size_t AudioPlayer::fillBuffer(void *data, size_t size) {
    
    while (size_remaining > 0) {
        if (mInputBuffer == NULL) {
            if (mIsFirstBuffer) {
                mInputBuffer = mFirstBuffer;
                mFirstBuffer = NULL;
                err = mFirstBufferResult;

                mIsFirstBuffer = false;
            } else {
                err = mSource->read(&mInputBuffer, &options);
            }
        }  

        memcpy((char *)data + size_done,
               (const char *)mInputBuffer->data() + mInputBuffer->range_offset(),
               copy);
    }

    return size_done;
}