AudioTrack源码解读(2)
本篇介绍
本篇介绍下AudioTrack的操作，比如Playback的线程运行，播放，write，暂停等流程。
源码介绍
播放线程运行
首先从播放线程的拉起开始，这儿以PlaybackThread为例， 由于PlaybackThread是以智能指针形式存在的，因此在创建后，会调用onFirstRef，因此这整个流程从这个函数开始：
void AudioFlinger::PlaybackThread::onFirstRef()
{
if (mOutput == nullptr || mOutput->stream == nullptr) {
ALOGE("The stream is not open yet"); // This should not happen.
} else {
// setEventCallback will need a strong pointer as a parameter. Calling it
// here instead of constructor of PlaybackThread so that the onFirstRef
// callback would not be made on an incompletely constructed object.
if (mOutput->stream->setEventCallback(this) != OK) {
ALOGE("Failed to add event callback");
}
}
run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}最后会执行run，这个方法会进行调用threadLoop，这个方法比较长，我们分段看下重点部分：
AudioFlinger::PlaybackThread::threadLoop() {
...
// 处理配置命令
processConfigEvents_l();
// 删除该PalybackThread管理的Track中不需要播放的部分，比如播放结束的，暂停的等
mMixerStatus = prepareTracks_l(&tracksToRemove);
// 拷贝一份需要播放的Track
// Acquire a local copy of active tracks with lock (release w/o lock).
//
// Control methods on the track acquire the ThreadBase lock (e.g. start()
// stop(), pause(), etc.), but the threadLoop is entitled to call audio
// data / buffer methods on tracks from activeTracks without the ThreadBase lock.
activeTracks.insert(activeTracks.end(), mActiveTracks.begin(), mActiveTracks.end());
// mix
if (mMixerStatus == MIXER_TRACKS_READY) {
// threadLoop_mix() sets mCurrentWriteLength
threadLoop_mix();
}
// 音效处理
if (mSleepTimeUs == 0 && mType != OFFLOAD) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
// TODO: Write haptic data directly to sink buffer when mixing.
if (activeHapticSessionId != AUDIO_SESSION_NONE
&& activeHapticSessionId == effectChains[i]->sessionId()) {
// Haptic data is active in this case, copy it directly from
// in buffer to out buffer.
const size_t audioBufferSize = mNormalFrameCount
* audio_bytes_per_frame(mChannelCount, EFFECT_BUFFER_FORMAT);
memcpy_by_audio_format(
(uint8_t*)effectChains[i]->outBuffer() + audioBufferSize,
EFFECT_BUFFER_FORMAT,
(const uint8_t*)effectChains[i]->inBuffer() + audioBufferSize,
EFFECT_BUFFER_FORMAT, mNormalFrameCount * mHapticChannelCount);
}
}
}
// 写音频数据
ret = threadLoop_write();
// 结束处理
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();
...
}从这儿可以看出以下几点：
- PlaybackThread会从所管理的Track中拿出需要播放的音频，然后进行混流，这样的话，可以猜想到，如果有些音频暂停，结束之类的，只需要不让对应的Track参与混流就可以了；
- 每个PlaybackThread都对应一个AudioStreamOut，用来将音频数据写到hal，然后由hal写给驱动。
这儿再看下音频数据如何写入Hal，具体实现是threadLoop_write：
// shared by MIXER and DIRECT, overridden by DUPLICATING
ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
if (mNormalSink != 0) {
const size_t count = mBytesRemaining / mFrameSize;
ATRACE_BEGIN("write");
// update the setpoint when AudioFlinger::mScreenState changes
uint32_t screenState = AudioFlinger::mScreenState;
if (screenState != mScreenState) {
mScreenState = screenState;
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
if (pipe != NULL) {
pipe->setAvgFrames((mScreenState & 1) ?
(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
}
}
ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); // 写入normal sink
ATRACE_END();
if (framesWritten > 0) {
bytesWritten = framesWritten * mFrameSize;
#ifdef TEE_SINK
mTee.write((char *)mSinkBuffer + offset, framesWritten); // 写入tee sink
#endif
} else {
bytesWritten = framesWritten;
}
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
mWriteAckSequence |= 1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
ATRACE_BEGIN("write");
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 写入hal
ATRACE_END();
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
mWriteAckSequence &= ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
}
mNumWrites++;
mInWrite = false;
if (mStandby) {
mThreadMetrics.logBeginInterval();
mStandby = false;
}
return bytesWritten;
}这儿是调用了AudioStreamOut的write接口，继续看下内部实现：
ssize_t AudioStreamOut::write(const void *buffer, size_t numBytes)
{
size_t bytesWritten;
status_t result = stream->write(buffer, numBytes, &bytesWritten);
if (result == OK && bytesWritten > 0 && mHalFrameSize > 0) {
mFramesWritten += bytesWritten / mHalFrameSize;
}
return result == OK ? bytesWritten : result;
}可以想到，这儿的stream 就是连接hal的，具体可以看下构造的地方：
status_t AudioStreamOut::open(
audio_io_handle_t handle,
audio_devices_t deviceType,
struct audio_config *config,
const char *address)
{
sp<StreamOutHalInterface> outStream;
audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
? (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
: flags;
int status = hwDev()->openOutputStream(
handle,
deviceType,
customFlags,
config,
address,
&outStream);
// Some HALs may not recognize AUDIO_FORMAT_IEC61937. But if we declare
// it as PCM then it will probably work.
if (status != NO_ERROR && config->format == AUDIO_FORMAT_IEC61937) {
struct audio_config customConfig = *config;
customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
status = hwDev()->openOutputStream(
handle,
deviceType,
customFlags,
&customConfig,
address,
&outStream);
ALOGV("AudioStreamOut::open(), treat IEC61937 as PCM, status = %d", status);
}
if (status == NO_ERROR) {
stream = outStream;
mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
status = stream->getFrameSize(&mHalFrameSize);
LOG_ALWAYS_FATAL_IF(status != OK, "Error retrieving frame size from HAL: %d", status);
LOG_ALWAYS_FATAL_IF(mHalFrameSize <= 0, "Error frame size was %zu but must be greater than"
" zero", mHalFrameSize);
}
return status;
}这儿就拿到了hal的stream 接口。
Play接口实现
public void play()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("play() called on uninitialized AudioTrack.");
}
//FIXME use lambda to pass startImpl to superclass
final int delay = getStartDelayMs();
if (delay == 0) {
startImpl();
} else {
new Thread() {
public void run() {
try {
Thread.sleep(delay);
} catch (InterruptedException e) {
e.printStackTrace();
}
baseSetStartDelayMs(0);
try {
startImpl();
} catch (IllegalStateException e) {
// fail silently for a state exception when it is happening after
// a delayed start, as the player state could have changed between the
// call to start() and the execution of startImpl()
}
}
}.start();
}
}本质上就是调用startImpl：
private void startImpl() {
synchronized(mPlayStateLock) {
baseStart();
native_start();
if (mPlayState == PLAYSTATE_PAUSED_STOPPING) {
mPlayState = PLAYSTATE_STOPPING;
} else {
mPlayState = PLAYSTATE_PLAYING;
mOffloadEosPending = false;
}
}
}这样就一个native调用就下去了，可以想象到Native的实现一定会走到AudioFlinger的播放线程的Track里面。可以沿着这个思路看下代码： 先到JNI
static void
android_media_AudioTrack_start(JNIEnv *env, jobject thiz)
{
sp<AudioTrack> lpTrack = getAudioTrack(env, thiz);
if (lpTrack == NULL) {
jniThrowException(env, "java/lang/IllegalStateException",
"Unable to retrieve AudioTrack pointer for start()");
return;
}
lpTrack->start();
}再到Native的AudioTrack：
status_t AudioTrack::start()
{
...
if (!(flags & CBLK_INVALID)) {
status = mAudioTrack->start(); // 调用Track的开始
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
}
}
if (flags & CBLK_INVALID) {
status = restoreTrack_l("start");
}
// resume or pause the callback thread as needed.
sp<AudioTrackThread> t = mAudioTrackThread;
if (status == NO_ERROR) {
if (t != 0) {
if (previousState == STATE_STOPPING) {
mProxy->interrupt();
} else {
t->resume(); // 启动回调线程
}
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
// Start our local VolumeHandler for restoration purposes.
mVolumeHandler->setStarted();
} else {
ALOGE("%s(%d): status %d", __func__, mPortId, status);
mState = previousState;
if (t != 0) {
if (previousState != STATE_STOPPING) {
t->pause();
}
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
return status;
}
}先看下Track的开始：
status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
audio_session_t triggerSession __unused)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
if (isOffloaded()) {
Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
Mutex::Autolock _lth(thread->mLock);
sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
(ec != 0 && ec->isNonOffloadableEnabled())) {
invalidate();
return PERMISSION_DENIED;
}
}
Mutex::Autolock _lth(thread->mLock);
track_state state = mState;
// clear mPauseHwPending because of pause (and possibly flush) during underrun.
mPauseHwPending = false;
if (state == PAUSED || state == PAUSING) {
if (mResumeToStopping) {
// happened we need to resume to STOPPING_1
mState = TrackBase::STOPPING_1;
ALOGV("%s(%d): PAUSED => STOPPING_1 on thread %d",
__func__, mId, (int)mThreadIoHandle);
} else {
mState = TrackBase::RESUMING;
ALOGV("%s(%d): PAUSED => RESUMING on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
} else {
mState = TrackBase::ACTIVE; //修改了标记
ALOGV("%s(%d): ? => ACTIVE on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
// states to reset position info for non-offloaded/direct tracks
if (!isOffloaded() && !isDirect()
&& (state == IDLE || state == STOPPED || state == FLUSHED)) {
mFrameMap.reset();
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (isFastTrack()) {
// refresh fast track underruns on start because that field is never cleared
// by the fast mixer; furthermore, the same track can be recycled, i.e. start
// after stop.
mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex);
}
status = playbackThread->addTrack_l(this); // 添加Track，这样ThreadLoop就可以遍历到这个Track了
if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
// restore previous state if start was rejected by policy manager
if (status == PERMISSION_DENIED) {
mState = state;
}
}
// Audio timing metrics are computed a few mix cycles after starting.
{
mLogStartCountdown = LOG_START_COUNTDOWN;
mLogStartTimeNs = systemTime();
mLogStartFrames = mAudioTrackServerProxy->getTimestamp()
.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
mLogLatencyMs = 0.;
}
if (status == NO_ERROR || status == ALREADY_EXISTS) {
// for streaming tracks, remove the buffer read stop limit.
mAudioTrackServerProxy->start(); // 重设标记内存读取位置
}
// track was already in the active list, not a problem
if (status == ALREADY_EXISTS) {
status = NO_ERROR;
} else {
// Acknowledge any pending flush(), so that subsequent new data isn't discarded.
// It is usually unsafe to access the server proxy from a binder thread.
// But in this case we know the mixer thread (whether normal mixer or fast mixer)
// isn't looking at this track yet: we still hold the normal mixer thread lock,
// and for fast tracks the track is not yet in the fast mixer thread's active set.
// For static tracks, this is used to acknowledge change in position or loop.
ServerProxy::Buffer buffer;
buffer.mFrameCount = 1;
(void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
}
} else {
status = BAD_VALUE;
}
if (status == NO_ERROR) {
forEachTeePatchTrack([](auto patchTrack) { patchTrack->start(); });
}
return status;
}接下来看下开启回调, 开启回调后，在配置了用回调获取数据的场景，就会从Native向Java要数据，并写入共享内存：
bool AudioTrack::AudioTrackThread::threadLoop()
{
{
AutoMutex _l(mMyLock);
if (mPaused) {
// TODO check return value and handle or log
mMyCond.wait(mMyLock);
// caller will check for exitPending()
return true;
}
if (mIgnoreNextPausedInt) {
mIgnoreNextPausedInt = false;
mPausedInt = false;
}
if (mPausedInt) {
// TODO use futex instead of condition, for event flag "or"
if (mPausedNs > 0) {
// TODO check return value and handle or log
(void) mMyCond.waitRelative(mMyLock, mPausedNs);
} else {
// TODO check return value and handle or log
mMyCond.wait(mMyLock);
}
mPausedInt = false;
return true;
}
}
if (exitPending()) {
return false;
}
nsecs_t ns = mReceiver.processAudioBuffer(); // 驱动回调
switch (ns) {
case 0:
return true;
case NS_INACTIVE:
pauseInternal();
return true;
case NS_NEVER:
return false;
case NS_WHENEVER:
// Event driven: call wake() when callback notifications conditions change.
ns = INT64_MAX;
FALLTHROUGH_INTENDED;
default:
LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
__func__, mReceiver.mPortId, (long long)ns);
pauseInternal(ns);
return true;
}
}可以猜想到，回调接下来就会通过jni向java要数据：
...
nsecs_t AudioTrack::processAudioBuffer()
{
// perform callbacks while unlocked
if (newUnderrun) {
mCbf(EVENT_UNDERRUN, mUserData, NULL);
}
while (loopCountNotifications > 0) {
mCbf(EVENT_LOOP_END, mUserData, NULL);
--loopCountNotifications;
}
if (flags & CBLK_BUFFER_END) {
mCbf(EVENT_BUFFER_END, mUserData, NULL);
}
if (markerReached) {
mCbf(EVENT_MARKER, mUserData, &markerPosition);
}
while (newPosCount > 0) {
size_t temp = newPosition.value(); // FIXME size_t != uint32_t
mCbf(EVENT_NEW_POS, mUserData, &temp);
newPosition += updatePeriod;
newPosCount--;
}
if (mObservedSequence != sequence) {
mObservedSequence = sequence;
mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
// for offloaded tracks, just wait for the upper layers to recreate the track
if (isOffloadedOrDirect()) {
return NS_INACTIVE;
}
}
// if inactive, then don't run me again until re-started
if (!active) {
return NS_INACTIVE;
}
// ...
}这儿的mCbl就是jni的audioCallback
static void audioCallback(int event, void* user, void *info) {
...
if (postEvent) {
JNIEnv *env = AndroidRuntime::getJNIEnv();
if (env != NULL) {
env->CallStaticVoidMethod(
callbackInfo->audioTrack_class,
javaAudioTrackFields.postNativeEventInJava,
callbackInfo->audioTrack_ref, event, arg, 0, NULL);
if (env->ExceptionCheck()) {
env->ExceptionDescribe();
env->ExceptionClear();
}
}
}
{
Mutex::Autolock l(sLock);
callbackInfo->busy = false;
callbackInfo->cond.broadcast();
}
}这儿向Java侧抛一个事件，然后Java侧就会调用索要数据接口：
private static void postEventFromNative(Object audiotrack_ref,
int what, int arg1, int arg2, Object obj) {
//logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2);
final AudioTrack track = (AudioTrack) ((WeakReference) audiotrack_ref).get();
if (track == null) {
return;
}
if (what == AudioSystem.NATIVE_EVENT_ROUTING_CHANGE) {
track.broadcastRoutingChange();
return;
}
if (what == NATIVE_EVENT_CODEC_FORMAT_CHANGE) {
ByteBuffer buffer = (ByteBuffer) obj;
buffer.order(ByteOrder.nativeOrder());
buffer.rewind();
AudioMetadataReadMap audioMetaData = AudioMetadata.fromByteBuffer(buffer);
if (audioMetaData == null) {
Log.e(TAG, "Unable to get audio metadata from byte buffer");
return;
}
track.mCodecFormatChangedListeners.notify(0 /* eventCode, unused */, audioMetaData);
return;
}
if (what == NATIVE_EVENT_CAN_WRITE_MORE_DATA
|| what == NATIVE_EVENT_NEW_IAUDIOTRACK
|| what == NATIVE_EVENT_STREAM_END) {
track.handleStreamEventFromNative(what, arg1);
return;
}
NativePositionEventHandlerDelegate delegate = track.mEventHandlerDelegate;
if (delegate != null) {
Handler handler = delegate.getHandler();
if (handler != null) {
Message m = handler.obtainMessage(what, arg1, arg2, obj);
handler.sendMessage(m);
}
}
}利用looper抛一个消息，接下来处理消息：
public void handleMessage(Message msg) {
final LinkedList<StreamEventCbInfo> cbInfoList;
synchronized (mStreamEventCbLock) {
if (msg.what == NATIVE_EVENT_STREAM_END) {
synchronized (mPlayStateLock) {
if (mPlayState == PLAYSTATE_STOPPING) {
if (mOffloadEosPending) {
native_start();
mPlayState = PLAYSTATE_PLAYING;
} else {
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
mPlayState = PLAYSTATE_STOPPED;
}
mOffloadEosPending = false;
mPlayStateLock.notify();
}
}
}
if (mStreamEventCbInfoList.size() == 0) {
return;
}
cbInfoList = new LinkedList<StreamEventCbInfo>(mStreamEventCbInfoList);
}
final long identity = Binder.clearCallingIdentity();
try {
for (StreamEventCbInfo cbi : cbInfoList) {
switch (msg.what) {
case NATIVE_EVENT_CAN_WRITE_MORE_DATA:
cbi.mStreamEventExec.execute(() ->
cbi.mStreamEventCb.onDataRequest(AudioTrack.this, msg.arg1));
break;
case NATIVE_EVENT_NEW_IAUDIOTRACK:
// TODO also release track as it's not longer usable
cbi.mStreamEventExec.execute(() ->
cbi.mStreamEventCb.onTearDown(AudioTrack.this));
break;
case NATIVE_EVENT_STREAM_END:
cbi.mStreamEventExec.execute(() ->
cbi.mStreamEventCb.onPresentationEnded(AudioTrack.this));
break;
}
}
} finally {
Binder.restoreCallingIdentity(identity);
}
}这样就完成了数据索要流程，也就完成了Start流程。
Pause介绍
这两个流程是类似的，也是从AudioTrack到AudioFlinger的Track，调用过去后设置一个标记，这样ThreadLoop的时候就可以针对Pause进行处理了，这儿就不看Java侧的实现了，流程就是调用native的接口，没做其他主要事情，直接从Native开始看：
android_media_AudioTrack_pause(JNIEnv *env, jobject thiz)
{
sp<AudioTrack> lpTrack = getAudioTrack(env, thiz);
if (lpTrack == NULL) {
jniThrowException(env, "java/lang/IllegalStateException",
"Unable to retrieve AudioTrack pointer for pause()");
return;
}
lpTrack->pause();
}看下pause的实现：
void AudioTrack::pause()
{
const int64_t beginNs = systemTime();
AutoMutex lock(mLock);
mediametrics::Defer defer([&]() {
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
.record(); });
ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
if (mState == STATE_ACTIVE) {
mState = STATE_PAUSED;
} else if (mState == STATE_STOPPING) {
mState = STATE_PAUSED_STOPPING;
} else {
return;
}
mProxy->interrupt();
mAudioTrack->pause();
}看下AudioFlinger中的pause实现：
void AudioFlinger::PlaybackThread::Track::pause()
{
ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
switch (mState) {
case STOPPING_1:
case STOPPING_2:
if (!isOffloaded()) {
/* nothing to do if track is not offloaded */
break;
}
// Offloaded track was draining, we need to carry on draining when resumed
mResumeToStopping = true;
FALLTHROUGH_INTENDED;
case ACTIVE:
case RESUMING:
mState = PAUSING;
ALOGV("%s(%d): ACTIVE/RESUMING => PAUSING on thread %d",
__func__, mId, (int)mThreadIoHandle);
if (isOffloadedOrDirect()) {
mPauseHwPending = true;
}
playbackThread->broadcast_l();
break;
default:
break;
}
}
// Pausing the TeePatch to avoid a glitch on underrun, at the cost of buffered audio loss.
forEachTeePatchTrack([](auto patchTrack) { patchTrack->pause(); });
}果然没做其他事情，就是设置了下标记。
Stop实现
这儿直接从AudioTrack native的实现看，可以猜想到这儿需要完成的事情主要有调用Track的stop，并停止回调线程
void AudioTrack::stop()
{
if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
return;
}
if (isOffloaded_l()) {
mState = STATE_STOPPING;
} else {
mState = STATE_STOPPED;
ALOGD_IF(mSharedBuffer == nullptr,
"%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
mReleased = 0;
}
mProxy->stop(); // notify server not to read beyond current client position until start().
mProxy->interrupt();
mAudioTrack->stop(); // 停止Track
// Note: legacy handling - stop does not clear playback marker
// and periodic update counter, but flush does for streaming tracks.
if (mSharedBuffer != 0) {
// clear buffer position and loop count.
mStaticProxy->setBufferPositionAndLoop(0 /* position */,
0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
}
sp<AudioTrackThread> t = mAudioTrackThread; // 停止回调索要数据
if (t != 0) {
if (!isOffloaded_l()) {
t->pause();
} else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
// causes wake up of the playback thread, that will callback the client for
// EVENT_STREAM_END in processAudioBuffer()
t->wake();
}
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}这儿主要看下Track中的stop:
void AudioFlinger::PlaybackThread::Track::stop()
{
ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
track_state state = mState;
if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
// If the track is not active (PAUSED and buffers full), flush buffers
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
mState = STOPPED;
} else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
mState = STOPPED;
} else {
// For fast tracks prepareTracks_l() will set state to STOPPING_2
// presentation is complete
// For an offloaded track this starts a drain and state will
// move to STOPPING_2 when drain completes and then STOPPED
mState = STOPPING_1;
if (isOffloaded()) {
mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload;
}
}
playbackThread->broadcast_l();
ALOGV("%s(%d): not stopping/stopped => stopping/stopped on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
}
forEachTeePatchTrack([](auto patchTrack) { patchTrack->stop(); });
}这儿也是设置下标记。看起来stop和pause没啥区别，实际上还是有区别的，对于stop的Track是可以被释放的，而pause的却不会。
Write的实现
这儿以native_write_byte为例：
static jint writeToTrack(const sp<AudioTrack>& track, jint audioFormat, const T *data,
jint offsetInSamples, jint sizeInSamples, bool blocking) {
// give the data to the native AudioTrack object (the data starts at the offset)
ssize_t written = 0;
// regular write() or copy the data to the AudioTrack's shared memory?
size_t sizeInBytes = sizeInSamples * sizeof(T);
if (track->sharedBuffer() == 0) {
written = track->write(data + offsetInSamples, sizeInBytes, blocking);
// for compatibility with earlier behavior of write(), return 0 in this case
if (written == (ssize_t) WOULD_BLOCK) {
written = 0;
}
} else {
// writing to shared memory, check for capacity
if ((size_t)sizeInBytes > track->sharedBuffer()->size()) {
sizeInBytes = track->sharedBuffer()->size();
}
memcpy(track->sharedBuffer()->unsecurePointer(), data + offsetInSamples, sizeInBytes);
written = sizeInBytes;
}
if (written >= 0) {
return written / sizeof(T);
}
return interpretWriteSizeError(written);
}如果有共享buffer就直接copy，如果没有，那么就需要调用AudioTrack的write：
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
return INVALID_OPERATION;
}
if (isDirect()) {
AutoMutex lock(mLock);
int32_t flags = android_atomic_and(
~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
&mCblk->mFlags);
if (flags & CBLK_INVALID) {
return DEAD_OBJECT;
}
}
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Validation: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
__func__, mPortId, buffer, userSize, userSize);
return BAD_VALUE;
}
size_t written = 0;
Buffer audioBuffer;
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
status_t err = obtainBuffer(&audioBuffer,
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
if (err < 0) {
if (written > 0) {
break;
}
if (err == TIMED_OUT || err == -EINTR) {
err = WOULD_BLOCK;
}
return ssize_t(err);
}
size_t toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
}
if (written > 0) {
mFramesWritten += written / mFrameSize;
if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
const sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
// causes wake up of the playback thread, that will callback the client for
// more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
t->wake();
}
}
}
return written;
}获取一个buffer，然后再写入； 看下obtainBuffer：
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed, size_t *nonContig)
{
// previous and new IAudioTrack sequence numbers are used to detect track re-creation
uint32_t oldSequence = 0;
Proxy::Buffer buffer;
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
do {
// obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
// keep them from going away if another thread re-creates the track during obtainBuffer()
sp<AudioTrackClientProxy> proxy;
sp<IMemory> iMem;
{ // start of lock scope
AutoMutex lock(mLock);
uint32_t newSequence = mSequence;
// did previous obtainBuffer() fail due to media server death or voluntary invalidation?
if (status == DEAD_OBJECT) {
// re-create track, unless someone else has already done so
if (newSequence == oldSequence) {
status = restoreTrack_l("obtainBuffer");
if (status != NO_ERROR) {
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
}
}
oldSequence = newSequence;
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
}
// Keep the extra references
proxy = mProxy;
iMem = mCblkMemory;
if (mState == STATE_STOPPING) {
status = -EINTR;
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
// Non-blocking if track is stopped or paused
if (mState != STATE_ACTIVE) {
requested = &ClientProxy::kNonBlocking;
}
} // end of lock scope
buffer.mFrameCount = audioBuffer->frameCount;
// FIXME starts the requested timeout and elapsed over from scratch
status = proxy->obtainBuffer(&buffer, requested, elapsed); // 从共享buffer中申请内存
} while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
audioBuffer->size = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
audioBuffer->sequence = oldSequence;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
}
return status;
}为什么这儿的proxy就可以获取共享内存？ 因为在AudioFlinger中创建Track的时候就分配了一块共享内存，并把内存fd 通过binder共享给了调用方，这样调用方就可以直接在这块内存上获取可用内存了，具体实现如下：
status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested,
struct timespec *elapsed)
{
for (;;) {
int32_t flags = android_atomic_and(~CBLK_INTERRUPT, &cblk->mFlags);
// check for track invalidation by server, or server death detection
if (flags & CBLK_INVALID) {
ALOGV("Track invalidated");
status = DEAD_OBJECT;
goto end;
}
if (flags & CBLK_DISABLED) {
ALOGV("Track disabled");
status = NOT_ENOUGH_DATA;
goto end;
}
// check for obtainBuffer interrupted by client
if (!ignoreInitialPendingInterrupt && (flags & CBLK_INTERRUPT)) {
ALOGV("obtainBuffer() interrupted by client");
status = -EINTR;
goto end;
}
ignoreInitialPendingInterrupt = false;
// compute number of frames available to write (AudioTrack) or read (AudioRecord)
int32_t front;
int32_t rear;
if (mIsOut) {
// The barrier following the read of mFront is probably redundant.
// We're about to perform a conditional branch based on 'filled',
// which will force the processor to observe the read of mFront
// prior to allowing data writes starting at mRaw.
// However, the processor may support speculative execution,
// and be unable to undo speculative writes into shared memory.
// The barrier will prevent such speculative execution.
front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
rear = cblk->u.mStreaming.mRear;
} else {
// On the other hand, this barrier is required.
rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
front = cblk->u.mStreaming.mFront;
}
// write to rear, read from front
ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
if (mIsOut) {
ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); "
"shutting down", filled, mFrameCount);
mIsShutdown = true;
status = NO_INIT;
goto end;
}
// for input, sync up on overrun
filled = 0;
cblk->u.mStreaming.mFront = rear;
(void) android_atomic_or(CBLK_OVERRUN, &cblk->mFlags);
}
// Don't allow filling pipe beyond the user settable size.
// The calculation for avail can go negative if the buffer size
// is suddenly dropped below the amount already in the buffer.
// So use a signed calculation to prevent a numeric overflow abort.
ssize_t adjustableSize = (ssize_t) getBufferSizeInFrames();
ssize_t avail = (mIsOut) ? adjustableSize - filled : filled;
if (avail < 0) {
avail = 0;
} else if (avail > 0) {
// 'avail' may be non-contiguous, so return only the first contiguous chunk
size_t part1;
if (mIsOut) {
rear &= mFrameCountP2 - 1;
part1 = mFrameCountP2 - rear;
} else {
front &= mFrameCountP2 - 1;
part1 = mFrameCountP2 - front;
}
if (part1 > (size_t)avail) {
part1 = avail;
}
if (part1 > buffer->mFrameCount) {
part1 = buffer->mFrameCount;
}
buffer->mFrameCount = part1;
buffer->mRaw = part1 > 0 ?
&((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL;
buffer->mNonContig = avail - part1;
mUnreleased = part1;
status = NO_ERROR;
break;
}
struct timespec remaining;
const struct timespec *ts;
switch (timeout) {
case TIMEOUT_ZERO:
status = WOULD_BLOCK;
goto end;
case TIMEOUT_INFINITE:
ts = NULL;
break;
case TIMEOUT_FINITE:
timeout = TIMEOUT_CONTINUE;
if (MAX_SEC == 0) {
ts = requested;
break;
}
FALLTHROUGH_INTENDED;
case TIMEOUT_CONTINUE:
// FIXME we do not retry if requested < 10ms? needs documentation on this state machine
if (!measure || requested->tv_sec < total.tv_sec ||
(requested->tv_sec == total.tv_sec && requested->tv_nsec <= total.tv_nsec)) {
status = TIMED_OUT;
goto end;
}
remaining.tv_sec = requested->tv_sec - total.tv_sec;
if ((remaining.tv_nsec = requested->tv_nsec - total.tv_nsec) < 0) {
remaining.tv_nsec += 1000000000;
remaining.tv_sec++;
}
if (0 < MAX_SEC && MAX_SEC < remaining.tv_sec) {
remaining.tv_sec = MAX_SEC;
remaining.tv_nsec = 0;
}
ts = &remaining;
break;
default:
LOG_ALWAYS_FATAL("obtainBuffer() timeout=%d", timeout);
ts = NULL;
break;
}
int32_t old = android_atomic_and(~CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
if (measure && !beforeIsValid) {
clock_gettime(CLOCK_MONOTONIC, &before);
beforeIsValid = true;
}
errno = 0;
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old & ~CBLK_FUTEX_WAKE, ts);
status_t error = errno; // clock_gettime can affect errno
// update total elapsed time spent waiting
if (measure) {
struct timespec after;
clock_gettime(CLOCK_MONOTONIC, &after);
total.tv_sec += after.tv_sec - before.tv_sec;
// Use auto instead of long to avoid the google-runtime-int warning.
auto deltaNs = after.tv_nsec - before.tv_nsec;
if (deltaNs < 0) {
deltaNs += 1000000000;
total.tv_sec--;
}
if ((total.tv_nsec += deltaNs) >= 1000000000) {
total.tv_nsec -= 1000000000;
total.tv_sec++;
}
before = after;
beforeIsValid = true;
}
switch (error) {
case 0: // normal wakeup by server, or by binderDied()
case EWOULDBLOCK: // benign race condition with server
case EINTR: // wait was interrupted by signal or other spurious wakeup
case ETIMEDOUT: // time-out expired
// FIXME these error/non-0 status are being dropped
break;
default:
status = error;
ALOGE("%s unexpected error %s", __func__, strerror(status));
goto end;
}
}
}
}这样就完成了数据的write。
flush实现介绍
Flush用来清空buffer，这样调用方后续就可以使用更多的内存了。 直接从Native看：
void AudioTrack::flush()
{
const int64_t beginNs = systemTime();
AutoMutex lock(mLock);
mediametrics::Defer defer([&]() {
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
.record(); });
ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
if (mSharedBuffer != 0) {
return;
}
if (mState == STATE_ACTIVE) {
return;
}
flush_l();
}
void AudioTrack::flush_l()
{
ALOG_ASSERT(mState != STATE_ACTIVE);
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
mRefreshRemaining = true;
mState = STATE_FLUSHED;
mReleased = 0;
if (isOffloaded_l()) {
mProxy->interrupt();
}
mProxy->flush();
mAudioTrack->flush();
}看下Track中的flush：
void AudioFlinger::PlaybackThread::Track::flush()
{
ALOGV("%s(%d)", __func__, mId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
// Flush the ring buffer now if the track is not active in the PlaybackThread.
// Otherwise the flush would not be done until the track is resumed.
// Requires FastTrack removal be BLOCK_UNTIL_ACKED
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
(void)mServerProxy->flushBufferIfNeeded();
}
if (isOffloaded()) {
// If offloaded we allow flush during any state except terminated
// and keep the track active to avoid problems if user is seeking
// rapidly and underlying hardware has a significant delay handling
// a pause
if (isTerminated()) {
return;
}
ALOGV("%s(%d): offload flush", __func__, mId);
reset();
if (mState == STOPPING_1 || mState == STOPPING_2) {
ALOGV("%s(%d): flushed in STOPPING_1 or 2 state, change state to ACTIVE",
__func__, mId);
mState = ACTIVE;
}
mFlushHwPending = true;
mResumeToStopping = false;
} else {
if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// FLUSHED state
mState = FLUSHED;
// do not reset the track if it is still in the process of being stopped or paused.
// this will be done by prepareTracks_l() when the track is stopped.
// prepareTracks_l() will see mState == FLUSHED, then
// remove from active track list, reset(), and trigger presentation complete
if (isDirect()) {
mFlushHwPending = true;
}
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
}
// Prevent flush being lost if the track is flushed and then resumed
// before mixer thread can run. This is important when offloading
// because the hardware buffer could hold a large amount of audio
playbackThread->broadcast_l();
}
// Flush the Tee to avoid on resume playing old data and glitching on the transition to new data
forEachTeePatchTrack([](auto patchTrack) { patchTrack->flush(); });
}这儿没做其他的，就是设置了一个标记mFlushHwPending。在threadLoop里面会调用flushAck，而该方法的实现就是处理flush：
// must be called with thread lock held
void AudioFlinger::PlaybackThread::Track::flushAck()
{
if (!isOffloaded() && !isDirect())
return;
// Clear the client ring buffer so that the app can prime the buffer while paused.
// Otherwise it might not get cleared until playback is resumed and obtainBuffer() is called.
mServerProxy->flushBufferIfNeeded();
mFlushHwPending = false;
}看下flushBufferIfNeeded的实现：
void ServerProxy::flushBufferIfNeeded()
{
audio_track_cblk_t* cblk = mCblk;
// The acquire_load is not really required. But since the write is a release_store in the
// client, using acquire_load here makes it easier for people to maintain the code,
// and the logic for communicating ipc variables seems somewhat standard,
// and there really isn't much penalty for 4 or 8 byte atomics.
int32_t flush = android_atomic_acquire_load(&cblk->u.mStreaming.mFlush);
if (flush != mFlush) {
ALOGV("ServerProxy::flushBufferIfNeeded() mStreaming.mFlush = 0x%x, mFlush = 0x%0x",
flush, mFlush);
// shouldn't matter, but for range safety use mRear instead of getRear().
int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
int32_t front = cblk->u.mStreaming.mFront;
// effectively obtain then release whatever is in the buffer
const size_t overflowBit = mFrameCountP2 << 1;
const size_t mask = overflowBit - 1;
int32_t newFront = (front & ~mask) | (flush & mask);
ssize_t filled = audio_utils::safe_sub_overflow(rear, newFront);
if (filled >= (ssize_t)overflowBit) {
// front and rear offsets span the overflow bit of the p2 mask
// so rebasing newFront on the front offset is off by the overflow bit.
// adjust newFront to match rear offset.
ALOGV("flush wrap: filled %zx >= overflowBit %zx", filled, overflowBit);
newFront += overflowBit;
filled -= overflowBit;
}
// Rather than shutting down on a corrupt flush, just treat it as a full flush
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, "
"filled %zd=%#x",
mFlush, flush, front, rear,
(unsigned)mask, newFront, filled, (unsigned)filled);
newFront = rear;
}
mFlush = flush;
android_atomic_release_store(newFront, &cblk->u.mStreaming.mFront);
// There is no danger from a false positive, so err on the side of caution
if (true /*front != newFront*/) {
int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
}
}
mFlushed += (newFront - front) & mask;
}
}这时候就完成了AudioTrack主要流程的实现解读。
本文参与 腾讯云自媒体同步曝光计划，分享自作者个人站点/博客。
原始发表：2021-10-04，如有侵权请联系 cloudcommunity@tencent.com 删除
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