前言：

继上一篇Android AudioFlinger（三）—— AndroidAudio Flinger 之设备管理我们知道PlaybackThread继承自Re’fBase， 在被第一次引用的时候就会调用onFirstRef，实现如下：

void AudioFlinger::PlaybackThread::onFirstRef()
{run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}

很简单就调用了一个run方法去开起了一个ThreadLoop线程：

bool AudioFlinger::PlaybackThread::threadLoop()
{
...
}

接下来我们进一步研究下PlaybackThread的循环主题具体做了什么？

揭开PlaybackThread面纱

当进入到threadloop就说明playbackthread的音频事务正式开启了。代码比较多，但是我们如果仔细看的话会发现关键代码就几处，而且都是threadLoop_前缀的，threadLoop_standby\threadLoop_mix\threadLoop_sleepTime\threadLoop_write等，这也代表这些函数都是threadLoop内部调用的。

bool AudioFlinger::PlaybackThread::threadLoop()
{
...while (!exitPending()){...{ // scope for mLock//这个地方框起来主要就是限制自动锁_l的生命周期，Mutex::Autolock _l(mLock);//处理config事件processConfigEvents_l();            if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||isSuspended()) {if (shouldStandby_l()) {//进入standby状态节省能耗threadLoop_standby();}}//准备音频流mMixerStatus = prepareTracks_l(&tracksToRemove);} // mLock scope ends...if (mBytesRemaining == 0) {mCurrentWriteLength = 0;if (mMixerStatus == MIXER_TRACKS_READY) {//读取所有active设备数据，混音器开始混音threadLoop_mix();} else if ((mMixerStatus != MIXER_DRAIN_TRACK)&& (mMixerStatus != MIXER_DRAIN_ALL)) {//进入休眠threadLoop_sleepTime();}}...if (!waitingAsyncCallback()) {if (mSleepTimeUs == 0) {if (mBytesRemaining) {//把混音器处理好的数据写入到输出流设备ret = threadLoop_write();} else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||(mMixerStatus == MIXER_DRAIN_ALL)) {threadLoop_drain();}...}}
... //移除相关的trackthreadLoop_removeTracks(tracksToRemove);tracksToRemove.clear();clearOutputTracks();effectChains.clear();}threadLoop_exit();if (!mStandby) {threadLoop_standby();mStandby = true;}releaseWakeLock();return false;
}

首先exitPending是threadloop循环的条件，这个函数是Thread的内部函数，它主要就是通过判断mExitPending来决定是否退出线程，这个值默认为false，在收到requestExit或者requestExitAndWait的时候会变为true，然后就会退出循环。
Thread PATH:/system/core/libutils/Threads.cpp

processConfigEvents_l: 处理config时间，当有配置发声变化的时候会调用sendConfigEvent_l来把事件添加到mConfigEvents中，最终processConfigEvents_l检测到就会去处理对应的配置。

threadLoop_standby： 判断当前是否符合standby条件，符合就调用threadLoop_standby，最终的实现其实是hal层实现，会做出关闭音频流等操作。

prepareTracks_l： 这个函数非常复杂，我们简单概括下，挑几个重点谈一谈

// prepareTracks_l() must be called with ThreadBase::mLock held
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(Vector< sp<Track> > *tracksToRemove)
{//获取当前活跃的track数量size_t count = mActiveTracks.size();for (size_t i=0 ; i<count ; i++) {//循环每个活跃的trackconst sp<Track> t = mActiveTracks[i];// this const just means the local variable doesn't changeTrack* const track = t.get();// process fast tracksif (track->isFastTrack()) {//如果是fasttrack改如何处理}
...{   // local variable scope to avoid goto warning//数据块准备操作audio_track_cblk_t* cblk = track->cblk();//获取track的音频信息const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();desiredFrames = sourceFramesNeededWithTimestretch(sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());uint32_t minFrames = 1;if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {//至少需要准备的音频帧数minFrames = desiredFrames;}size_t framesReady = track->framesReady();if ((framesReady >= minFrames) && track->isReady() &&!track->isPaused() && !track->isTerminated()){mixedTracks++;// compute volume for this trackuint32_t vl, vr;       // in U8.24 integer formatfloat vlf, vrf, vaf;   // in [0.0, 1.0] float format//左声道，右声道，aux level音量// read original volumes with volume controlfloat typeVolume = mStreamTypes[track->streamType()].volume;//获取每个stream类型的音频音量float v = masterVolume * typeVolume;//主音量和类型音量相乘if (track->isPausing() || mStreamTypes[track->streamType()].mute) {vl = vr = 0;vlf = vrf = vaf = 0.;//设置0，代表静音操作if (track->isPausing()) {track->setPaused();//track设置暂停}} else {sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;gain_minifloat_packed_t vlr = proxy->getVolumeLR();//得到音量的增益值vlf = float_from_gain(gain_minifloat_unpack_left(vlr));vrf = float_from_gain(gain_minifloat_unpack_right(vlr));//转换为浮点值// track volumes come from shared memory, so can't be trusted and must be clamped//判断是否在合理范围内if (vlf > GAIN_FLOAT_UNITY) {ALOGV("Track left volume out of range: %.3g", vlf);vlf = GAIN_FLOAT_UNITY;}if (vrf > GAIN_FLOAT_UNITY) {ALOGV("Track right volume out of range: %.3g", vrf);vrf = GAIN_FLOAT_UNITY;}const float vh = track->getVolumeHandler()->getVolume(track->mAudioTrackServerProxy->framesReleased()).first;// now apply the master volume and stream type volume and shaper volumevlf *= v * vh;vrf *= v * vh;// assuming master volume and stream type volume each go up to 1.0,// then derive vl and vr as U8.24 versions for the effect chainconst float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;vl = (uint32_t) (scaleto8_24 * vlf);vr = (uint32_t) (scaleto8_24 * vrf);// vl and vr are now in U8.24 formatuint16_t sendLevel = proxy->getSendLevel_U4_12();// send level comes from shared memory and so may be corruptif (sendLevel > MAX_GAIN_INT) {ALOGV("Track send level out of range: %04X", sendLevel);sendLevel = MAX_GAIN_INT;}// vaf is represented as [0.0, 1.0] float by rescaling sendLevelvaf = v * sendLevel * (1. / MAX_GAIN_INT);}track->setFinalVolume((vrf + vlf) / 2.f);// XXX: these things DON'T need to be done each timemAudioMixer->setBufferProvider(name, track);mAudioMixer->enable(name);mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);...} else {...}}   // local variable scope to avoid goto warning}return mixerStatus;
}

mActiveTracks记录了当前处于活跃状态的track，接着就是循环遍历每一个track进行处理，获取对应的音频参数。

audio_track_cblk_t是音频数据块，后面我们会扩展讲解。

在之后minFrames代表了此次音频播放所需要的最小帧数，他的初始值为1。当track->sharedBuffer() == 0的时候，说明这个AudioTrack不是STATIC模式（数据不是一次性传送完成的）。

getUnreleasedFrames用来获取音频缓冲区中尚未被音频硬件处理的帧数。

当我们计算出minFrames之后，就开始判断当前音频的各种指标是否符合标准。

vlf, vrf, vaf分别表示，左声道音量，右声道音量，AUX level音量，浮点数表示。

根据streamType获取对应stream类型音频的音量，然后进行判断是否在合理范围内，最终经过计算设置到AudioMixer对象中。当准备工作完成后，就进入到了真正的混音操作中了。

threadloop_mix:主要就是调用AudioMixer的process函数进行处理，这样就进入了audiomixer。

void AudioFlinger::MixerThread::threadLoop_mix()
{// 启动混音mAudioMixer->process();mCurrentWriteLength = mSinkBufferSize;//当应用程序欠载情况清除时，逐步增加睡眠时间。//仅当混频器连续两次准备就绪时才增加睡眠时间，//以避免交替的就绪/未就绪条件的稳定状态保持睡眠时间，从而导致音频 HAL 欠载。if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {sleepTimeShift--;}mSleepTimeUs = 0;mStandbyTimeNs = systemTime() + mStandbyDelayNs;//TODO: delay standby when effects have a tail}

最后就是将数据写入HAL层了，threadloop_write。
当mNormalSink存在的时候调用他的write函数写入，不存在就调用mOutput的write函数，mOutput就是 AudioStreamOut。

ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{ssize_t bytesWritten;// If an NBAIO sink is present, use it to write the normal mixer's submixif (mNormalSink != 0) {ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);ATRACE_END();if (framesWritten > 0) {bytesWritten = framesWritten * mFrameSize;} else {bytesWritten = framesWritten;}// otherwise use the HAL / AudioStreamOut directly} else {bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);}return bytesWritten;
}

写入完成后调用各种清理的函数，remove，clear等。

// Finally let go of removed track(s), without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.  This will also mutate and push a new fast mixer state.
threadLoop_removeTracks(tracksToRemove);
tracksToRemove.clear();// FIXME I don't understand the need for this here;
//       it was in the original code but maybe the
//       assignment in saveOutputTracks() makes this unnecessary?
clearOutputTracks();// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();

void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
{size_t count = tracksToRemove.size();if (count > 0) {for (size_t i = 0 ; i < count ; i++) {const sp<Track>& track = tracksToRemove.itemAt(i);if (track->isExternalTrack()) {AudioSystem::stopOutput(mId, track->streamType(),track->sessionId());if (track->isTerminated()) {AudioSystem::releaseOutput(mId, track->streamType(),track->sessionId());}}}}
}