Android AudioFlinger（四）—— 揭开PlaybackThread面纱
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就分析playbackthread的音频变乱正式开启了。代码比力多，但是我们如果细致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track
threadLoop_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大概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，当有设置发声变化的时候会调用sendConfigEvent_l来把变乱添加到mConfigEvents中，最终processConfigEvents_l检测到就会去处理对应的设置。
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track
const sp<Track> t = mActiveTracks;
// this const just means the local variable doesn't change
Track* const track = t.get();
// process fast tracks
if (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 track
uint32_t vl, vr; // in U8.24 integer format
float vlf, vrf, vaf; // in float format//左声道，右声道，aux level音量
// read original volumes with volume control
float typeVolume = mStreamTypes.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.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 volume
vlf *= 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 chain
const 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 format
uint16_t sendLevel = proxy->getSendLevel_U4_12();
// send level comes from shared memory and so may be corrupt
if (sendLevel > MAX_GAIN_INT) {
ALOGV("Track send level out of range: %04X", sendLevel);
sendLevel = MAX_GAIN_INT;
}
// vaf is represented as float by rescaling sendLevel
vaf = v * sendLevel * (1. / MAX_GAIN_INT);
}
track->setFinalVolume((vrf + vlf) / 2.f);
// XXX: these things DON'T need to be done each time
mAudioMixer->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举行处理，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必要的最小帧数，他的初始值为1。当track->sharedBuffer() == 0的时候，分析这个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。
当我们盘算出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范例音频的音量，然后举行判断是否在公道范围内，最终经过盘算设置到AudioMixer对象中。当预备工作完成后，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函数举行处理，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 submix
if (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整理的函数，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());
}
}
}
}
}
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