好久没写了，今天碰巧有个小伙伴问我关于音频流这一块的，久了还有点记不起来，我就顺便写一下，后面就不用又找一遍代码了，所谓好记性不如烂笔头。
所以，这里是关于如何从AudioTrack 写入数据到audioflinger，以及audioflinger如何写入到hal层的音频流处理流程，主要写一下audioflinger处理流程，和写一些细节。
获取音频流
1、client写入数据：
app client 通过创建AudioTrack后，在播放的时候会不断的调用audiotrack的write方法，不断的向audioflinger写数据。
//frameworks\av\media\libaudioclient\AudioTrack.cpp
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC) {
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)) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", 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;
}
return written;
}
2、service端的写
作为service端，不断的接收并处理client写入的数据。
//frameworks/av/services/audioflinger/Tracks.cpp
bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.raw = data;
//处理客户端write数据，这个就不在这里细说了，这里不打算细讲这个环形缓冲，有兴趣的可以自行阅读代码
}
AudioFlinger音频流处理
主要处理流程：
track预处理
混音处理
向HAL输出数据
bool AudioFlinger::PlaybackThread::threadLoop()
{ //1、track预处理
mMixerStatus = prepareTracks_l(&tracksToRemove);
//2、混音处理
threadLoop_mix();
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*/);
}
//将mMixerBuffer 按hal 层格式 实际最终还是拷贝到 mSinkBuffer 对象上
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
mNormalFrameCount * mChannelCount);
}
//3、向HAL输出数据
ret = threadLoop_write();
}
track预处理
向mAudioMixer更新track对象等操作
//prepareTracks_l
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
Vector< sp<Track> > *tracksToRemove)
{
.....
for (size_t i=0 ; i<count ; i++) {
const sp<Track> t = mActiveTracks[i];
....
// if an active track doesn't exist in the AudioMixer, create it.
if (!mAudioMixer->exists(name)) {
status_t status = mAudioMixer->create(
name,
track->mChannelMask,
track->mFormat,
track->mSessionId);
if (status != OK) {
ALOGW("%s: cannot create track name"
" %d, mask %#x, format %#x, sessionId %d in AudioMixer",
__func__, name, track->mChannelMask, track->mFormat, track->mSessionId);
tracksToRemove->add(track);
track->invalidate(); // consider it dead.
continue;
}
}
....
}
.....
}
混音处理
void AudioFlinger::MixerThread::threadLoop_mix()
{
// mix buffers...
mAudioMixer->process();
mCurrentWriteLength = mSinkBufferSize;
// increase sleep time progressively when application underrun condition clears.
// Only increase sleep time if the mixer is ready for two consecutive times to avoid
// that a steady state of alternating ready/not ready conditions keeps the sleep time
// such that we would underrun the audio HAL.
if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
sleepTimeShift--;
}
mSleepTimeUs = 0;
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
//TODO: delay standby when effects have a tail
}
//frameworks/av/media/libaudioprocessing/include/media/AudioMixerBase.h
void process() {
preProcess();
//这里才是处理数据
(this->*mHook)();
postProcess();
}
数据获取&混音处理 (this->*mHook)();
mHook 以process__genericResampling 为例：
//frameworks/av/media/libaudioprocessing/AudioMixerBase.cpp
// generic code with resampling
void AudioMixerBase::process__genericResampling()
{
ALOGVV("process__genericResampling\n");
int32_t * const outTemp = mOutputTemp.get(); // naked ptr
size_t numFrames = mFrameCount;
for (const auto &pair : mGroups) {
const auto &group = pair.second;
const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
// clear temp buffer
memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
for (const int name : group) {
const std::shared_ptr<TrackBase> &t = mTracks[name];
int32_t *aux = NULL;
if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
aux = t->auxBuffer;
}
// this is a little goofy, on the resampling case we don't
// acquire/release the buffers because it's done by
// the resampler.
if (t->needs & NEEDS_RESAMPLE) {
(t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
} else {
size_t outFrames = 0;
while (outFrames < numFrames) {
t->buffer.frameCount = numFrames - outFrames;
//获取client写入的数据
t->bufferProvider->getNextBuffer(&t->buffer);
t->mIn = t->buffer.raw;
// t->mIn == nullptr can happen if the track was flushed just after having
// been enabled for mixing.
if (t->mIn == nullptr) break;
(t.get()->*t->hook)(
outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
mResampleTemp.get() /* naked ptr */,
aux != nullptr ? aux + outFrames : nullptr);
outFrames += t->buffer.frameCount;
//通知释放 通知client 有可以写入的buffer了
t->bufferProvider->releaseBuffer(&t->buffer);
}
}
}
/**
这个mainBuffer与 mOutput->write((char *)mSinkBuffer, 0); 中的mSinkBuffer 相关联
//相关代码如下：
//frameworks\av\services\audioflinger\Threads.cpp
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
Vector< sp<Track> > *tracksToRemove)
{
........
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
........
}
*/
//这里是将outTemp 数据按最终输出格式拷贝给到 mainBuffer
convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
}
}
向HAL输出数据
// frameworks\av\services\audioflinger\Threads.cpp
ssize_t AudioFlinger::MixerThread::threadLoop_write()
{
// FIXME we should only do one push per cycle; confirm this is true
// Start the fast mixer if it's not already running
if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (state->mCommand != FastMixerState::MIX_WRITE &&
(kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
if (state->mCommand == FastMixerState::COLD_IDLE) {
// FIXME workaround for first HAL write being CPU bound on some devices
ATRACE_BEGIN("write");
//AudioStreamOut *mOutput;
//致此，audioflinger的写入流程基本结束，剩下的就是写入到audiohal层，
//最后在audiohal处理之后就会通过tinyalsa写入到驱动完成了声音流的输出过程了
mOutput->write((char *)mSinkBuffer, 0);
ATRACE_END();
int32_t old = android_atomic_inc(&mFastMixerFutex);
if (old == -1) {
(void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
}
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
mAudioWatchdog->resume();
}
#endif
}
state->mCommand = FastMixerState::MIX_WRITE;
#ifdef FAST_THREAD_STATISTICS
mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
#endif
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
if (kUseFastMixer == FastMixer_Dynamic) {
mNormalSink = mPipeSink;
}
} else {
sq->end(false /*didModify*/);
}
}
return PlaybackThread::threadLoop_write();
}
补充
每个client 创建audiotrack的时候都是要通过AudioFlinger来创建的。audioflinger交给Threads.cpp这边创建，创建完毕后会保存在Threads 的mTracks 对象里面，当每次loop的时候会先调用prepareTracks_l 来准备数据，此时会及时将mTracks 同步到AudioMixer（不是将对象直接赋值给AudioMixer对象，AudioMixer会同步创建一个AudioMixer::Track来与mTracks对应，这个AudioMixer::Track然后再包含Track，最后实际获取数据的时候是通过这个Track来获取到client数据的）。
sp<IAudioTrack> AudioFlinger::createTrack(const CreateTrackInput& input,
CreateTrackOutput& output,
status_t *status)
{
......
track = thread->createTrack_l(client, streamType, input.attr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, &output.notificationFrameCount,
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
input.clientInfo.clientTid, clientUid, &lStatus, portId);
......
}
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
size_t *pNotificationFrameCount,
uint32_t notificationsPerBuffer,
float speed,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
audio_output_flags_t *flags,
pid_t tid,
uid_t uid,
status_t *status,
audio_port_handle_t portId)
{
......
track = new Track(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, uid, *flags, TrackBase::TYPE_DEFAULT, portId);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
// track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
......
}
最后
sourceinside阅读有些跳转不了，有点恼火。
好了就简单写到这里了，五一就要到了，想下五一去哪玩耍先！
