- AudioFlinger 回放录制线程
AndioFlinger 作为 Android 的音频系统引擎，重任之一是负责输入输出流设备的管理及音频流数据的处理传输，这是由回放线程（PlaybackThread 及其派生的子类）和录制线程（RecordThread）进行的，我们简单看看回放线程和录制线程类关系：
ThreadBase：PlaybackThread 和 RecordThread 的基类
RecordThread：录制线程类，由 ThreadBase 派生
PlaybackThread：回放线程基类，同由 ThreadBase 派生
MixerThread：混音回放线程类，由 PlaybackThread 派生，负责处理标识为 AUDIO_OUTPUT_FLAG_PRIMARY、AUDIO_OUTPUT_FLAG_FAST、AUDIO_OUTPUT_FLAG_DEEP_BUFFER 的音频流，MixerThread 可以把多个音轨的数据混音后再输出
DirectOutputThread：直输回放线程类，由 PlaybackThread 派生，负责处理标识为 AUDIO_OUTPUT_FLAG_DIRECT 的音频流，这种音频流数据不需要软件混音，直接输出到音频设备即可
DuplicatingThread：复制回放线程类，由 MixerThread 派生，负责复制音频流数据到其他输出设备，使用场景如主声卡设备、蓝牙耳机设备、USB 声卡设备同时输出
OffloadThread：硬解回放线程类，由 DirectOutputThread 派生，负责处理标识为 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD 的音频流，这种音频流未经软件解码的（一般是 MP3、AAC 等格式的数据），需要输出到硬件解码器，由硬件解码器解码成 PCM 数据
PlaybackThread 中有个极为重要的函数 threadLoop()，当 PlaybackThread 被强引用时，threadLoop() 会真正运行起来进入循环主体，处理音频流数据相关事务，threadLoop() 大致流程如下（以 MixerThread 为例）：
bool AudioFlinger::PlaybackThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
mStandbyTimeNs = systemTime();
nsecs_t lastWriteFinished = -1; // time last server write completed
int64_t lastFramesWritten = -1; // track changes in timestamp server frames written
// MIXER
nsecs_t lastWarning = 0;
// DUPLICATING
// FIXME could this be made local to while loop?
writeFrames = 0;
int lastGeneration = 0;
cacheParameters_l();
mSleepTimeUs = mIdleSleepTimeUs;
if (mType == MIXER) {
sleepTimeShift = 0;
}
CpuStats cpuStats;
const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
acquireWakeLock();
#ifdef SRS_PROCESSING
String8 bt_param = String8("bluetooth_enabled=0");
POSTPRO_PATCH_PARAMS_SET(bt_param);
if (mType == MIXER) {
POSTPRO_PATCH_OUTPROC_PLAY_INIT(this, myName);
} else if (mType == OFFLOAD) {
POSTPRO_PATCH_OUTPROC_DIRECT_INIT(this, myName);
POSTPRO_PATCH_OUTPROC_PLAY_ROUTE_BY_VALUE(this, mOutDevice);
} else if (mType == DIRECT) {
POSTPRO_PATCH_OUTPROC_DIRECT_INIT(this, myName);
POSTPRO_PATCH_OUTPROC_PLAY_ROUTE_BY_VALUE(this, mOutDevice);
}
#endif
// mNBLogWriter->log can only be called while thread mutex mLock is held.
// So if you need to log when mutex is unlocked, set logString to a non-NULL string,
// and then that string will be logged at the next convenient opportunity.
const char *logString = NULL;
checkSilentMode_l();
while (!exitPending())
{
cpuStats.sample(myName);
Vector< sp<EffectChain> > effectChains;
{ // scope for mLock
Mutex::Autolock _l(mLock);
processConfigEvents_l();
if (logString != NULL) {
mNBLogWriter->logTimestamp();
mNBLogWriter->log(logString);
logString = NULL;
}
// Gather the framesReleased counters for all active tracks,
// and associate with the sink frames written out. We need
// this to convert the sink timestamp to the track timestamp.
bool kernelLocationUpdate = false;
if (mNormalSink != 0) {
// Note: The DuplicatingThread may not have a mNormalSink.
// We always fetch the timestamp here because often the downstream
// sink will block while writing.
ExtendedTimestamp timestamp; // use private copy to fetch
(void) mNormalSink->getTimestamp(timestamp);
// We keep track of the last valid kernel position in case we are in underrun
// and the normal mixer period is the same as the fast mixer period, or there
// is some error from the HAL.
if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
}
if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
kernelLocationUpdate = true;
} else {
ALOGVV("getTimestamp error - no valid kernel position");
}
// copy over kernel info
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+ mSuspendedFrames; // add frames discarded when suspended
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
}
// mFramesWritten for non-offloaded tracks are contiguous
// even after standby() is called. This is useful for the track frame
// to sink frame mapping.
bool serverLocationUpdate = false;
if (mFramesWritten != lastFramesWritten) {
serverLocationUpdate = true;
lastFramesWritten = mFramesWritten;
}
// Only update timestamps if there is a meaningful change.
// Either the kernel timestamp must be valid or we have written something.
if (kernelLocationUpdate || serverLocationUpdate) {
if (serverLocationUpdate) {
// use the time before we called the HAL write - it is a bit more accurate
// to when the server last read data than the current time here.
//
// If we haven't written anything, mLastWriteTime will be -1
// and we use systemTime().
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1
? systemTime() : mLastWriteTime;
}
const size_t size = mActiveTracks.size();
for (size_t i = 0; i < size; ++i) {
sp<Track> t = mActiveTracks[i].promote();
if (t != 0 && !t->isFastTrack()) {
t->updateTrackFrameInfo(
t->mAudioTrackServerProxy->framesReleased(),
mFramesWritten,
mTimestamp);
}
}
}
saveOutputTracks();
if (mSignalPending) {
// A signal was raised while we were unlocked
mSignalPending = false;
} else if (waitingAsyncCallback_l()) {
if (exitPending()) {
break;
}
bool released = false;
if (!keepWakeLock()) {
releaseWakeLock_l();
released = true;
mWakeLockUids.clear();
mActiveTracksGeneration++;
}
ALOGV("wait async completion");
mWaitWorkCV.wait(mLock);
ALOGV("async completion/wake");
if (released) {
acquireWakeLock_l();
}
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mSleepTimeUs = 0;
continue;
}
if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
isSuspended()) {
// put audio hardware into standby after short delay
if (shouldStandby_l()) {
threadLoop_standby();
mStandby = true;
}
if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
clearOutputTracks();
if (exitPending()) {
break;
}
releaseWakeLock_l();
mWakeLockUids.clear();
mActiveTracksGeneration++;
// wait until we have something to do...
ALOGV("%s going to sleep", myName.string());
mWaitWorkCV.wait(mLock);
ALOGV("%s waking up", myName.string());
acquireWakeLock_l();
mMixerStatus = MIXER_IDLE;
mMixerStatusIgnoringFastTracks = MIXER_IDLE;
mBytesWritten = 0;
mBytesRemaining = 0;
checkSilentMode_l();
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mSleepTimeUs = mIdleSleepTimeUs;
if (mType == MIXER) {
sleepTimeShift = 0;
}
continue;
}
}
// mMixerStatusIgnoringFastTracks is also updated internally
mMixerStatus = prepareTracks_l(&tracksToRemove);
#ifdef DOLBY_ENABLE // DOLBY_DAP_PREGAIN
// If there are no active tracks, then reset volume to zero for this thread.
if (mMixerStatus == MIXER_IDLE) {
ALOGV("EffectDapController: Reset volumes to zeros for threadType = %d flags = %d", mType, mOutput->flags);
EffectDapController::instance()->updatePregain(mType, mId, mOutput->flags, 0);
}
#endif // DOLBY_END
// compare with previously applied list
if (lastGeneration != mActiveTracksGeneration) {
// update wakelock
updateWakeLockUids_l(mWakeLockUids);
lastGeneration = mActiveTracksGeneration;
}
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
} // mLock scope ends
if (mBytesRemaining == 0) {
mCurrentWriteLength = 0;
if (mMixerStatus == MIXER_TRACKS_READY) {
// threadLoop_mix() sets mCurrentWriteLength
threadLoop_mix();
} else if ((mMixerStatus != MIXER_DRAIN_TRACK)
&& (mMixerStatus != MIXER_DRAIN_ALL)) {
// threadLoop_sleepTime sets mSleepTimeUs to 0 if data
// must be written to HAL
threadLoop_sleepTime();
if (mSleepTimeUs == 0) {
mCurrentWriteLength = mSinkBufferSize;
}
}
// Either threadLoop_mix() or threadLoop_sleepTime() should have set
// mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
// Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
// or mSinkBuffer (if there are no effects).
//
// This is done pre-effects computation; if effects change to
// support higher precision, this needs to move.
//
// mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
// TODO use mSleepTimeUs == 0 as an additional condition.
if (mMixerBufferValid) {
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
// mono blend occurs for mixer threads only (not direct or offloaded)
// and is handled here if we're going directly to the sink.
if (requireMonoBlend() && !mEffectBufferValid) {
mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount,
true /*limit*/);
}
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
mNormalFrameCount * mChannelCount);
}
mBytesRemaining = mCurrentWriteLength;
if (isSuspended()) {
// Simulate write to HAL when suspended (e.g. BT SCO phone call).
mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer.
const size_t framesRemaining = mBytesRemaining / mFrameSize;
mBytesWritten += mBytesRemaining;
mFramesWritten += framesRemaining;
mSuspendedFrames += framesRemaining; // to adjust kernel HAL position
mBytesRemaining = 0;
}
// only process effects if we're going to write
if (mSleepTimeUs == 0 && mType != OFFLOAD &&
!(mType == DIRECT && mIsDirectPcm)) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
}
}
// Process effect chains for offloaded thread even if no audio
// was read from audio track: process only updates effect state
// and thus does have to be synchronized with audio writes but may have
// to be called while waiting for async write callback
if ((mType == OFFLOAD) || (mType == DIRECT && mIsDirectPcm)) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
}
#ifdef SRS_PROCESSING
// Offload thread
if (mType == OFFLOAD) {
char buffer[2];
POSTPRO_PATCH_OUTPROC_DIRECT_SAMPLES(this, AUDIO_FORMAT_PCM_16_BIT, (int16_t *) buffer, 2, 48000, 2);
}
#endif
// Only if the Effects buffer is enabled and there is data in the
// Effects buffer (buffer valid), we need to
// copy into the sink buffer.
// TODO use mSleepTimeUs == 0 as an additional condition.
if (mEffectBufferValid) {
//ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
if (requireMonoBlend()) {
mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount,
true /*limit*/);
}
memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
mNormalFrameCount * mChannelCount);
}
// enable changes in effect chain
unlockEffectChains(effectChains);
if (!waitingAsyncCallback()) {
// mSleepTimeUs == 0 means we must write to audio hardware
if (mSleepTimeUs == 0) {
ssize_t ret = 0;
#ifdef SRS_PROCESSING
if (mType == MIXER && mMixerStatus == MIXER_TRACKS_READY) {
POSTPRO_PATCH_OUTPROC_PLAY_SAMPLES(this, mFormat, mSinkBuffer, mSinkBufferSize, mSampleRate, mChannelCount);
}
#endif
// We save lastWriteFinished here, as previousLastWriteFinished,
// for throttling. On thread start, previousLastWriteFinished will be
// set to -1, which properly results in no throttling after the first write.
nsecs_t previousLastWriteFinished = lastWriteFinished;
nsecs_t delta = 0;
if (mBytesRemaining) {
// FIXME rewrite to reduce number of system calls
mLastWriteTime = systemTime(); // also used for dumpsys
ret = threadLoop_write();
lastWriteFinished = systemTime();
delta = lastWriteFinished - mLastWriteTime;
if (ret < 0) {
mBytesRemaining = 0;
} else {
mBytesWritten += ret;
mBytesRemaining -= ret;
mFramesWritten += ret / mFrameSize;
}
} else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
(mMixerStatus == MIXER_DRAIN_ALL)) {
threadLoop_drain();
}
if (mType == MIXER && !mStandby) {
// write blocked detection
if (delta > maxPeriod) {
mNumDelayedWrites++;
if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) {
ATRACE_NAME("underrun");
ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
(unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
lastWarning = lastWriteFinished;
}
}
if (mThreadThrottle
&& mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks)
&& ret > 0) { // we wrote something
// Limit MixerThread data processing to no more than twice the
// expected processing rate.
//
// This helps prevent underruns with NuPlayer and other applications
// which may set up buffers that are close to the minimum size, or use
// deep buffers, and rely on a double-buffering sleep strategy to fill.
//
// The throttle smooths out sudden large data drains from the device,
// e.g. when it comes out of standby, which often causes problems with
// (1) mixer threads without a fast mixer (which has its own warm-up)
// (2) minimum buffer sized tracks (even if the track is full,
// the app won't fill fast enough to handle the sudden draw).
//
// Total time spent in last processing cycle equals time spent in
// 1. threadLoop_write, as well as time spent in
// 2. threadLoop_mix (significant for heavy mixing, especially
// on low tier processors)
// it's OK if deltaMs is an overestimate.
const int32_t deltaMs =
(lastWriteFinished - previousLastWriteFinished) / 1000000;
const int32_t throttleMs = mHalfBufferMs - deltaMs;
if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) {
usleep(throttleMs * 1000);
// notify of throttle start on verbose log
ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs,
"mixer(%p) throttle begin:"
" ret(%zd) deltaMs(%d) requires sleep %d ms",
this, ret, deltaMs, throttleMs);
mThreadThrottleTimeMs += throttleMs;
// Throttle must be attributed to the previous mixer loop's write time
// to allow back-to-back throttling.
lastWriteFinished += throttleMs * 1000000;
} else {
uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs;
if (diff > 0) {
// notify of throttle end on debug log
// but prevent spamming for bluetooth
ALOGD_IF(!audio_is_a2dp_out_device(outDevice()),
"mixer(%p) throttle end: throttle time(%u)", this, diff);
mThreadThrottleEndMs = mThreadThrottleTimeMs;
}
}
}
}
} else {
ATRACE_BEGIN("sleep");
Mutex::Autolock _l(mLock);
if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) {
mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs));
}
ATRACE_END();
}
}
// 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();
// FIXME Note that the above .clear() is no longer necessary since effectChains
// is now local to this block, but will keep it for now (at least until merge done).
}
threadLoop_exit();
if (!mStandby) {
threadLoop_standby();
mStandby = true;
}
#ifdef SRS_PROCESSING
if (mType == MIXER) {
POSTPRO_PATCH_OUTPROC_PLAY_EXIT(this, myName);
} else if (mType == OFFLOAD) {
POSTPRO_PATCH_OUTPROC_DIRECT_EXIT(this, myName);
} else if (mType == DIRECT) {
POSTPRO_PATCH_OUTPROC_DIRECT_EXIT(this, myName);
}
#endif
releaseWakeLock();
mWakeLockUids.clear();
mActiveTracksGeneration++;
ALOGV("Thread %p type %d exiting", this, mType);
return false;
}threadLoop() 循环的条件是 exitPending() 返回 false，如果想要 PlaybackThread 结束循环，则可以调用 requestExit() 来请求退出；
processConfigEvents_l() ：处理配置事件；当有配置改变的事件发生时，需要调用 sendConfigEvent_l() 来通知 PlaybackThread，这样 PlaybackThread 才能及时处理配置事件；常见的配置事件是切换音频通路；
检查此时此刻是否符合 standby 条件，比如当前并没有 ACTIVE 状态的 Track（mActiveTracks.size() = 0），那么调用 threadLoop_standby() 关闭音频硬件设备以节省能耗；
prepareTracks_l()： 准备音频流和混音器，该函数非常复杂，这里不详细分析了，仅列一下流程要点：
遍历 mActiveTracks，逐个处理 mActiveTracks 上的 Track，检查该 Track 是否为 ACTIVE 状态；
如果 Track 设置是 ACTIVE 状态，则再检查该 Track 的数据是否准备就绪了；
根据音频流的音量值、格式、声道数、音轨的采样率、硬件设备的采样率，配置好混音器参数；
如果 Track 的状态是 PAUSED 或 STOPPED，则把该 Track 添加到 tracksToRemove 向量中；
threadLoop_mix()：读取所有置了 ACTIVE 状态的音频流数据，混音器开始处理这些数据；
threadLoop_write()： 把混音器处理后的数据写到输出流设备；
threadLoop_removeTracks()： 把 tracksToRemove 上的所有 Track 从 mActiveTracks 中移除出来；这样下一次循环时就不会处理这些 Track 了。
这里说说 PlaybackThread 与输出流设备的关系：PlaybackThread 实例与输出流设备是一一对应的，比方说 OffloadThread 只会将音频数据输出到 compress_offload 设备中，MixerThread(with FastMixer) 只会将音频数据输出到 low_latency 设备中。
从 Audio HAL 中，我们通常看到如下 4 种输出流设备，分别对应着不同的播放场景：
primary_out：主输出流设备，用于铃声类声音输出，对应着标识为 AUDIO_OUTPUT_FLAG_PRIMARY 的音频流和一个 MixerThread 回放线程实例
low_latency：低延迟输出流设备，用于按键音、游戏背景音等对时延要求高的声音输出，对应着标识为 AUDIO_OUTPUT_FLAG_FAST 的音频流和一个 MixerThread 回放线程实例
deep_buffer：音乐音轨输出流设备，用于音乐等对时延要求不高的声音输出，对应着标识为 AUDIO_OUTPUT_FLAG_DEEP_BUFFER 的音频流和一个 MixerThread 回放线程实例
compress_offload：硬解输出流设备，用于需要硬件解码的数据输出，对应着标识为 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD 的音频流和一个 OffloadThread 回放线程实例
其中 primary_out 设备是必须声明支持的，而且系统启动时就已经打开 primary_out 设备并创建好对应的 MixerThread 实例。其他类型的输出流设备并非必须声明支持的，主要是看硬件上有无这个能力。
可能有人产生这样的疑问：既然 primary_out 设备一直保持打开，那么能耗岂不是很大？这里阐释一个概念：输出流设备属于逻辑设备，并不是硬件设备。所以即使输出流设备一直保持打开，只要硬件设备不工作，那么就不会影响能耗。那么硬件设备什么时候才会打开呢？答案是 PlaybackThread 将音频数据写入到输出流设备时。
下图简单描述 AudioTrack、PlaybackThread、输出流设备三者的对应关系：
我们可以这么说：输出流设备决定了它对应的 PlaybackThread 是什么类型。怎么理解呢？意思是说：只有支持了该类型的输出流设备，那么该类型的 PlaybackThread 才有可能被创建。举个例子：只有硬件上具备硬件解码器，系统才建立 compress_offload 设备，然后播放 mp3 格式的音乐文件时，才会创建 OffloadThread 把数据输出到 compress_offload 设备上；反之，如果硬件上并不具备硬件解码器，系统则不应该建立 compress_offload 设备，那么播放 mp3 格式的音乐文件时，通过 MixerThread 把数据输出到其他输出流设备上。
那么有无可能出现这种情况：底层并不支持 compress_offload 设备，但偏偏有个标识为 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD 的音频流送到 AudioFlinger 了呢？这是不可能的。系统启动时，会检查并保存输入输出流设备的支持信息；播放器在播放 mp3 文件时，首先看 compress_offload 设备是否支持了，如果支持，那么不进行软件解码，直接把数据标识为 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD；如果不支持，那么先进行软件解码，然后把解码好的数据标识为 AUDIO_OUTPUT_FLAG_DEEP_BUFFER，前提是 deep_buffer 设备是支持了的；如果 deep_buffer 设备也不支持，那么把数据标识为 AUDIO_OUTPUT_FLAG_PRIMARY。
系统启动时，就已经打开 primary_out、low_latency、deep_buffer 这三种输出流设备，并创建对应的 MixerThread 了；而此时 DirectOutputThread 与 OffloadThread 不会被创建，直到标识为 AUDIO_OUTPUT_FLAG_DIRECT/AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD 的音频流需要输出时，才开始创建 DirectOutputThread/OffloadThread 和打开 direct_out/compress_offload 设备。这一点请参考如下代码，注释非常清晰：
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
Thread(false),
#endif //AUDIO_POLICY_TEST
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mA2dpSuspended(false),
mAudioPortGeneration(1),
mBeaconMuteRefCount(0),
mBeaconPlayingRefCount(0),
mBeaconMuted(false),
mTtsOutputAvailable(false),
mMasterMono(false)
{
mUidCached = getuid();
mpClientInterface = clientInterface;
// TODO: remove when legacy conf file is removed. true on devices that use DRC on the
// DEVICE_CATEGORY_SPEAKER path to boost soft sounds, used to adjust volume curves accordingly.
// Note: remove also speaker_drc_enabled from global configuration of XML config file.
bool speakerDrcEnabled = false;
#ifdef USE_XML_AUDIO_POLICY_CONF
mVolumeCurves = new VolumeCurvesCollection();
AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices,
mDefaultOutputDevice, speakerDrcEnabled,
static_cast<VolumeCurvesCollection *>(mVolumeCurves));
PolicySerializer serializer;
if (serializer.deserialize(AUDIO_POLICY_XML_CONFIG_FILE, config) != NO_ERROR) {
#else
mVolumeCurves = new StreamDescriptorCollection();
AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices,
mDefaultOutputDevice, speakerDrcEnabled);
if ((ConfigParsingUtils::loadConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, config) != NO_ERROR) &&
(ConfigParsingUtils::loadConfig(AUDIO_POLICY_CONFIG_FILE, config) != NO_ERROR)) {
#endif
ALOGE("could not load audio policy configuration file, setting defaults");
config.setDefault();
}
// must be done after reading the policy (since conditionned by Speaker Drc Enabling)
mVolumeCurves->initializeVolumeCurves(speakerDrcEnabled);
// Once policy config has been parsed, retrieve an instance of the engine and initialize it.
audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance();
if (!engineInstance) {
ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__);
return;
}
// Retrieve the Policy Manager Interface
mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
if (mEngine == NULL) {
ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__);
return;
}
mEngine->setObserver(this);
status_t status = mEngine->initCheck();
(void) status;
ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status);
// mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
// open all output streams needed to access attached devices
audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
for (size_t i = 0; i < mHwModules.size(); i++) {
mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->getName());
if (mHwModules[i]->mHandle == 0) {
ALOGW("could not open HW module %s", mHwModules[i]->getName());
continue;
}
// open all output streams needed to access attached devices
// except for direct output streams that are only opened when they are actually
// required by an app.
// This also validates mAvailableOutputDevices list
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
if (!outProfile->hasSupportedDevices()) {
ALOGW("Output profile contains no device on module %s", mHwModules[i]->getName());
continue;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) {
mTtsOutputAvailable = true;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
audio_devices_t profileType = outProfile->getSupportedDevicesType();
if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) {
profileType = mDefaultOutputDevice->type();
} else {
// chose first device present in profile's SupportedDevices also part of
// outputDeviceTypes
profileType = outProfile->getSupportedDeviceForType(outputDeviceTypes);
}
if ((profileType & outputDeviceTypes) == 0) {
continue;
}
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
const DeviceVector &devicesForType = supportedDevices.getDevicesFromType(profileType);
String8 address = devicesForType.size() > 0 ? devicesForType.itemAt(0)->mAddress
: String8("");
outputDesc->mDevice = profileType;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = outputDesc->mSamplingRate;
config.channel_mask = outputDesc->mChannelMask;
config.format = outputDesc->mFormat;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openOutput(outProfile->getModuleHandle(),
&output,
&config,
&outputDesc->mDevice,
address,
&outputDesc->mLatency,
outputDesc->mFlags);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
mHwModules[i]->getName());
} else {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
for (size_t k = 0; k < supportedDevices.size(); k++) {
ssize_t index = mAvailableOutputDevices.indexOf(supportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
mAvailableOutputDevices[index]->attach(mHwModules[i]);
}
}
if (mPrimaryOutput == 0 &&
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
addOutput(output, outputDesc);
setOutputDevice(outputDesc,
outputDesc->mDevice,
true,
0,
NULL,
address.string());
}
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
if (!inProfile->hasSupportedDevices()) {
ALOGW("Input profile contains no device on module %s", mHwModules[i]->getName());
continue;
}
// chose first device present in profile's SupportedDevices also part of
// inputDeviceTypes
audio_devices_t profileType = inProfile->getSupportedDeviceForType(inputDeviceTypes);
if ((profileType & inputDeviceTypes) == 0) {
continue;
}
sp<AudioInputDescriptor> inputDesc =
new AudioInputDescriptor(inProfile);
inputDesc->mDevice = profileType;
// find the address
DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
// the inputs vector must be of size 1, but we don't want to crash here
String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
: String8("");
ALOGV(" for input device 0x%x using address %s", profileType, address.string());
ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = inputDesc->mSamplingRate;
config.channel_mask = inputDesc->mChannelMask;
config.format = inputDesc->mFormat;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openInput(inProfile->getModuleHandle(),
&input,
&config,
&inputDesc->mDevice,
address,
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE);
if (status == NO_ERROR) {
const DeviceVector &supportedDevices = inProfile->getSupportedDevices();
for (size_t k = 0; k < supportedDevices.size(); k++) {
ssize_t index = mAvailableInputDevices.indexOf(supportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if (index >= 0) {
sp<DeviceDescriptor> devDesc = mAvailableInputDevices[index];
if (!devDesc->isAttached()) {
devDesc->attach(mHwModules[i]);
devDesc->importAudioPort(inProfile);
}
}
}
mpClientInterface->closeInput(input);
} else {
ALOGW("Cannot open input stream for device %08x on hw module %s",
inputDesc->mDevice,
mHwModules[i]->getName());
}
}
}
// make sure all attached devices have been allocated a unique ID
for (size_t i = 0; i < mAvailableOutputDevices.size();) {
if (!mAvailableOutputDevices[i]->isAttached()) {
ALOGW("Output device %08x unreachable", mAvailableOutputDevices[i]->type());
mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
continue;
}
// The device is now validated and can be appended to the available devices of the engine
mEngine->setDeviceConnectionState(mAvailableOutputDevices[i],
AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
for (size_t i = 0; i < mAvailableInputDevices.size();) {
if (!mAvailableInputDevices[i]->isAttached()) {
ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type());
mAvailableInputDevices.remove(mAvailableInputDevices[i]);
continue;
}
// The device is now validated and can be appended to the available devices of the engine
mEngine->setDeviceConnectionState(mAvailableInputDevices[i],
AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
// make sure default device is reachable
if (mDefaultOutputDevice == 0 || mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type());
}
ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
updateDevicesAndOutputs();
#ifdef AUDIO_POLICY_TEST
if (mPrimaryOutput != 0) {
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"), 0);
mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, outputCmd.toString());
mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
mTestSamplingRate = 44100;
mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
mTestLatencyMs = 0;
mCurOutput = 0;
mDirectOutput = false;
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
mTestOutputs[i] = 0;
}
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "AudioPolicyManagerTest");
run(buffer, ANDROID_PRIORITY_AUDIO);
}
#endif //AUDIO_POLICY_TEST
}其中 mpClientInterface->openOutput() 最终会调用到 AudioFlinger::openOutput()：打开输出流设备，并创建 PlaybackThread 对象：
AudioFlinger OffloadThread
系统启动时，就已经打开 primary_out、low_latency、deep_buffer 这三种输出流设备，并创建对应的 MixerThread 了；而此时 DirectOutputThread 与 OffloadThread 不会被创建，直到标识为 AUDIO_OUTPUT_FLAG_DIRECT/AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD 的音频流需要输出时，才开始创建 DirectOutputThread/OffloadThread 和打开 direct_out/compress_offload 设备。
这里不知大家有无疑问：为什么 DirectOutputThread 与 OffloadThread 会被单独对待？DirectOuputThread 使用率较低，尚可以理解，但 OffloadThread 的使用率还是很高的，为什么不让 OffloadThread/compress_offload 设备也进入待命状态呢？
要回答这个问题：我们首先得明白 compress_offload 设备是什么东东，与其他输出流设备有什么不同。先看个图：
红色的是 Offload 音频流，它与其他音频流有什么本质的不同？Offload 音频流是未经 NuPlayerDecoder 进行解码的（NuPlayerDecoder 设置了 Passthrough 模式），所以必须把这些音频流数据送到 DSP，让 DSP 对其解码，解码后的 PCM 数据再送到 Codec 输出。
compress_offload 设备，说白了，就是驱动 DSP 解码数据、Codec 输出声音。 而 DSP 要解码数据，首先得知道数据的编码信息，如编码器 codec_id、采样率 sample_rate、声道数 channel_mask、比特率 bit_rate 等信息，由于 DSP 并没有实现 DataSource/Parser 部件，不能自己解析数据的编码信息，所以得有“人”告诉它，这个“人”无疑是 compress_offload 设备。
AudioTrack 构造函数有个 offloadInfo 的参数，参数原型定义如下：
NuPlayer DataSource/Parser 解析 mp3、flac 等文件得到数据编码信息，并在构造 AudioTrack 实例时作为参数传入，AudioFlinger 将基于这些编码信息打开 compress_offload 设备。
到这里，大家明白了吗？每个 mp3/flac 文件的编码信息可能是不一样的，比如 a.mp3 文件的编码信息是 mp3&44.1KHZ&16bit… ，而 b.flac 文件的编码信息是 flac&48KHz&24bit…； 播放 a.mp3 时，AudioFlinger 打开一个配置为 mp3&44.1KHz&16bit… 的 compress_offload 设备，接着播放 b.flac，就需要关闭之前的 compress_offload 设备，重新打开一个配置为 flac&48KHz&24bit… 的 compress_offload 设备。所以系统不会提前打开 compress_offload 设备，只有等到播放 mp3、flac 时取到明确的数据编码信息，才基于这些编码信息打开 compress_offload 设备。
编码信息包含很多条目，切换音源时，是否编码信息有一点点不一样，都需要重新打开 compress_offload 设备呢？不能运行时更新信息到 DSP 吗？其实 stagefright 和 compress_offload 是支持运行期更新某些信息的，也就是无缝切换，至于是哪些信息，依赖于 DSP 算法实现；有兴趣深入的可以参考 sendMetaDataToHal() 和 compress_set_gapless_metadata()。
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