Audio笔记之MixerThread

AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
        audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
    :   Thread(false /*canCallJava*/),
        mType(type),
        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
        // mChannelMask
        mChannelCount(0),
        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
        mParamStatus(NO_ERROR),
        mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
        // mName will be set by concrete (non-virtual) subclass
        mDeathRecipient(new PMDeathRecipient(this))
{
}

AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
                                             AudioStreamOut* output,
                                             audio_io_handle_t id,
                                             audio_devices_t device,
                                             type_t type)
    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
        // mStreamTypes[] initialized in constructor body
        mOutput(output),
        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
        mMixerStatus(MIXER_IDLE),
        mMixerStatusIgnoringFastTracks(MIXER_IDLE),
        standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
        mScreenState(AudioFlinger::mScreenState),
        // index 0 is reserved for normal mixer's submix
        // 初始值为1111 1110
        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
{
    snprintf(mName, kNameLength, "AudioOut_%X", id);
    mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);

    // Assumes constructor is called by AudioFlinger with it's mLock held, but
    // it would be safer to explicitly pass initial masterVolume/masterMute as
    // parameter.
    //
    // If the HAL we are using has support for master volume or master mute,
    // then do not attenuate or mute during mixing (just leave the volume at 1.0
    // and the mute set to false).
    mMasterVolume = audioFlinger->masterVolume_l();
    mMasterMute = audioFlinger->masterMute_l();
    if (mOutput && mOutput->audioHwDev) {
        if (mOutput->audioHwDev->canSetMasterVolume()) {
            mMasterVolume = 1.0;
        }

        if (mOutput->audioHwDev->canSetMasterMute()) {
            mMasterMute = false;
        }
    }
    // 负责读取硬件的各类参数
    readOutputParameters();

    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
            stream = (audio_stream_type_t) (stream + 1)) {
        mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
        mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
    }
    // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
    // because mAudioFlinger doesn't have one to copy from
}

//通过输出接口获得Hal层支持的采样率、声道数、采样精度、每帧大小、缓冲区包含的帧数、mix缓冲区大小
void AudioFlinger::PlaybackThread::readOutputParameters()
{
    // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL
    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
    if (!audio_is_output_channel(mChannelMask)) {
        LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
    }
    if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) {
        LOG_FATAL("HAL channel mask %#x not supported for mixed output; "
                "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask);
    }
    mChannelCount = popcount(mChannelMask);
    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
    if (!audio_is_valid_format(mFormat)) {
        LOG_FATAL("HAL format %d not valid for output", mFormat);
    }
    if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
        LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
                mFormat);
    }
    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
    // 缓冲区包含的帧数必须是16的倍数
    if (mFrameCount & 15) {
        ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
                mFrameCount);
    }
    // 如果该输出模块支持非阻塞模式，设置异步回调函数asyncCallback，并创建回调线程
    // AsyncCallbackThread，具体作用还需要分析
    if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) &&
            (mOutput->stream->set_callback != NULL)) {
        if (mOutput->stream->set_callback(mOutput->stream,
                                      AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
            mUseAsyncWrite = true;
            mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
        }
    }

    // Calculate size of normal mix buffer relative to the HAL output buffer size
    // 根据物理缓冲区的大小计算normal mix buffer的大小>= 物理缓冲区
    double multiplier = 1.0;
    if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
            kUseFastMixer == FastMixer_Dynamic)) {
        size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
        size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
        // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
        minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
        maxNormalFrameCount = maxNormalFrameCount & ~15;
        if (maxNormalFrameCount < minNormalFrameCount) {
            maxNormalFrameCount = minNormalFrameCount;
        }
        multiplier = (double) minNormalFrameCount / (double) mFrameCount;
        if (multiplier <= 1.0) {
            multiplier = 1.0;
        } else if (multiplier <= 2.0) {
            if (2 * mFrameCount <= maxNormalFrameCount) {
                multiplier = 2.0;
            } else {
                multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
            }
        } else {
            // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
            // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
            // track, but we sometimes have to do this to satisfy the maximum frame count
            // constraint)
            // FIXME this rounding up should not be done if no HAL SRC
            uint32_t truncMult = (uint32_t) multiplier;
            if ((truncMult & 1)) {
                if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
                    ++truncMult;
                }
            }
            multiplier = (double) truncMult;
        }
    }
    mNormalFrameCount = multiplier * mFrameCount;
    // round up to nearest 16 frames to satisfy AudioMixer
    mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
            mNormalFrameCount);

    delete[] mAllocMixBuffer;
    size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize;
    mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1];
    mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align);
    memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize);

    // force reconfiguration of effect chains and engines to take new buffer size and audio
    // parameters into account
    // Note that mLock is not held when readOutputParameters() is called from the constructor
    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
    // matter.
    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
    Vector< sp<EffectChain> > effectChains = mEffectChains;
    for (size_t i = 0; i < effectChains.size(); i ++) {
        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
    }
}

// ----------------------------------------------------------------------------
//
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
        audio_io_handle_t id, audio_devices_t device, type_t type)
    :   PlaybackThread(audioFlinger, output, id, device, type),
        // mAudioMixer below
        // mFastMixer below
        mFastMixerFutex(0)
        // mOutputSink below
        // mPipeSink below
        // mNormalSink below
{
    ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
            "mFrameCount=%d, mNormalFrameCount=%d",
            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
            mNormalFrameCount);
    //构造AudioMixer对象，负责mix norm tracks
    mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);

    // FIXME - Current mixer implementation only supports stereo output
    if (mChannelCount != FCC_2) {
        ALOGE("Invalid audio hardware channel count %d", mChannelCount);
    }

    // create an NBAIO sink for the HAL output stream, and negotiate
    // 通过输出流构造一个StreamOutSink对象，负责将数据写入到Hal中。
    mOutputSink = new AudioStreamOutSink(output->stream);
    size_t numCounterOffers = 0;
    const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
    ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
    ALOG_ASSERT(index == 0);

    // initialize fast mixer depending on configuration
    bool initFastMixer;
    switch (kUseFastMixer) {
    case FastMixer_Never:
        initFastMixer = false;
        break;
    case FastMixer_Always:
        initFastMixer = true;
        break;
    case FastMixer_Static:
    case FastMixer_Dynamic:
        initFastMixer = mFrameCount < mNormalFrameCount;
        break;
    }
    //如果支持FastMixer模式，需要做如下操作：
    //1、通过获取StreamOutSink的format来构造monoPipe对象，缓冲区大小是AudioMixer的4倍，
    //   负责获得AudioMixer对象的buffer数据,然后通过该monoPipe对象构造SourceAudioBufferProvider，
    //   作为FastTrack[0]的输入,由此我们可以知道这个对象名中有pipe(管道)的原因了，即像管道一样传输数据
    //2、构造FastMixer对象，初始化FastTrack[0]对象，并开启该线程，进行mix FastTrack处理
    //   特别注意，该FastTrack专门关联MixThread的MixBuff，后续作为一路track继续在FastMix中
    //   与其它FastTrack进行AudioMix混音处理
    //3、构造AudioWatchdog对象，负责监控该线程
    if (initFastMixer) {

        // create a MonoPipe to connect our submix to FastMixer
        NBAIO_Format format = mOutputSink->format();
        // This pipe depth compensates for scheduling latency of the normal mixer thread.
        // When it wakes up after a maximum latency, it runs a few cycles quickly before
        // finally blocking.  Note the pipe implementation rounds up the request to a power of 2.
        // 
        MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
        const NBAIO_Format offers[1] = {format};
        size_t numCounterOffers = 0;
        ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
        ALOG_ASSERT(index == 0);
        monoPipe->setAvgFrames((mScreenState & 1) ?
                (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
        mPipeSink = monoPipe;

#ifdef TEE_SINK
        if (mTeeSinkOutputEnabled) {
            // create a Pipe to archive a copy of FastMixer's output for dumpsys
            Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format);
            numCounterOffers = 0;
            index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
            ALOG_ASSERT(index == 0);
            mTeeSink = teeSink;
            PipeReader *teeSource = new PipeReader(*teeSink);
            numCounterOffers = 0;
            index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
            ALOG_ASSERT(index == 0);
            mTeeSource = teeSource;
        }
#endif

        // create fast mixer and configure it initially with just one fast track for our submix        
        // 构造FastMixer对象，初始化其中一个FastTrack对象，并开启该线程，进行mix FastTrack处理。
        mFastMixer = new FastMixer();
        FastMixerStateQueue *sq = mFastMixer->sq();
#ifdef STATE_QUEUE_DUMP
        sq->setObserverDump(&mStateQueueObserverDump);
        sq->setMutatorDump(&mStateQueueMutatorDump);
#endif
        FastMixerState *state = sq->begin();
        FastTrack *fastTrack = &state->mFastTracks[0];
        // wrap the source side of the MonoPipe to make it an AudioBufferProvider
        // 通过monoPipe来构造一个MonoPipeReader来读取monoPipe的数据，作为FastTrack[0]的输入端提供数据
        fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
        fastTrack->mVolumeProvider = NULL;
        fastTrack->mGeneration++;
        state->mFastTracksGen++;
        // 设置active track对应的bit位，初始值为0
        state->mTrackMask = 1;
        // fast mixer will use the HAL output sink
        // 使用AudioStreamOutSink来作为输出端，经过fast mixer处理过的数据将会提供给outputSink，最终写入到Hal中
        state->mOutputSink = mOutputSink.get();
        state->mOutputSinkGen++;
        state->mFrameCount = mFrameCount;
        state->mCommand = FastMixerState::COLD_IDLE;
        // already done in constructor initialization list
        //mFastMixerFutex = 0;
        state->mColdFutexAddr = &mFastMixerFutex;
        state->mColdGen++;
        state->mDumpState = &mFastMixerDumpState;
#ifdef TEE_SINK
        state->mTeeSink = mTeeSink.get();
#endif
        mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
        state->mNBLogWriter = mFastMixerNBLogWriter.get();
        sq->end();
        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);

        // start the fast mixer
        mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
        pid_t tid = mFastMixer->getTid();
        int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
        if (err != 0) {
            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                    kPriorityFastMixer, getpid_cached, tid, err);
        }

#ifdef AUDIO_WATCHDOG
        // create and start the watchdog
        mAudioWatchdog = new AudioWatchdog();
        mAudioWatchdog->setDump(&mAudioWatchdogDump);
        mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
        tid = mAudioWatchdog->getTid();
        err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
        if (err != 0) {
            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                    kPriorityFastMixer, getpid_cached, tid, err);
        }
#endif

    } else {
        mFastMixer = NULL;
    }
   
    //根据FastMixer的模式，设置当前的MixerThread的输出端
    //1、never时，始终将mix之后的数据输出到Hal
    //2、dynamic时，将mix之后的数据输出到Hal，但是已经初始化FastMixer对象，后续可以使用
    //2、always时，将mix之后的数据通过pipe输出到FastMixer，作为FastMixer的输入
    //3、static时，判断是Mix buff是否大于Hal buff，如果大于则使用pip将数据输出到FastMixer，
    //   否则将mix后的数据输出到Hal
    switch (kUseFastMixer) {
    case FastMixer_Never:
    case FastMixer_Dynamic:
        mNormalSink = mOutputSink;
        break;
    case FastMixer_Always:
        mNormalSink = mPipeSink;
        break;
    case FastMixer_Static:
        mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
        break;
    }
}