AudioTrack音频输出组件
概述
该组件主要实现音频输出对外的接口，它实际会使用到AudioSystem和AudioFlinger两个模块一起完成
音频输出的任务。
AudioTrack工作流程(C++层)
初始化AudioTrack实例
AudioTrack::AudioTrack( int streamType, uint32_t sampleRate, int format,
int channelMask, int frameCount, uint32_t flags, callback_t cbf,
void* user, int notificationFrames, int sessionId)
: mStatus(NO_INIT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0, false, sessionId);
}
设置音频输出参数和创建I/O句柄
status_t AudioTrack::set( int streamType, uint32_t sampleRate, int format,
int channelMask, int frameCount, uint32_t flags, callback_t cbf,
void* user, int notificationFrames, const sp<IMemory>& sharedBuffer,
bool threadCanCallJava, int sessionId)
{
......
// 通过AudioSystem获取音频相关的参数设置
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// 对默认的设置作处理
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (sampleRate == 0) {
sampleRate = afSampleRate;
}
if (format == 0) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
channelMask = AUDIO_CHANNEL_OUT_STEREO;
}
// 验证格式是否正确
if (!audio_is_valid_format(format)) {
return BAD_VALUE;
}
if (!audio_is_linear_pcm(format)) {
flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
}
if (!audio_is_output_channel(channelMask)) {
return BAD_VALUE;
}
uint32_t channelCount = popcount(channelMask);
// 获取一个StreamOutput的对象句柄，之后作为参数传递给createTrack_l方法
audio_io_handle_t output = AudioSystem::getOutput(
(audio_stream_type_t)streamType,
sampleRate,format, channelMask,
(audio_policy_output_flags_t)flags);
if (output == 0) {
return BAD_VALUE;
}
......
// 创建IAudioTrack接口，该接口由TrackHandle实现，
// 参见AudioFlinger一节
status_t status = createTrack_l(streamType, sampleRate,
(uint32_t)format, (uint32_t)channelMask,
frameCount, flags, sharedBuffer,
output, true);
......
// 创建一个播放的工作线程，为音频输出做准备
if (cbf != 0) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
return NO_INIT;
}
}
......
return NO_ERROR;
}
创建IAudioTrack对象
- AudioTrack和AudioFlinger的关系可参考下图
status_t AudioTrack::createTrack_l(
int streamType,
uint32_t sampleRate,
uint32_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount)
{
// 通过AudioSystem获取IAudioFlinger接口，实际的任务由该接口完成
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
return NO_INIT;
}
// 获取一些输出的参数
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
......
// 调用AudioFlinger的createTrack方法
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
format,
channelMask,
frameCount,
((uint16_t)flags) << 16,
sharedBuffer,
output,
&mSessionId,
&status);
......
// 由AudioFlinger创建一块共享内存，再分配到AudioTrack对象中
sp<IMemory> cblk = track->getCblk();
if (cblk == 0) {
return NO_INIT;
}
mAudioTrack.clear();
mAudioTrack = track;
mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
if (sharedBuffer == 0) {
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
} else {
mCblk->buffers = sharedBuffer->pointer();
// Force buffer full condition as data is already present in shared memory
mCblk->stepUser(mCblk->frameCount);
}
mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
mRemainingFrames = mNotificationFramesAct;
mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
return NO_ERROR;
}
创建AudioTrackThread
- 在初始化最后阶段，AudioTrack会创建一个AudioTrackThread对象，该对象是一个线程类。
- 通常将AudioTrack的this指针作为参数传递给AudioTrackThread对象，以便在线程开启时，执行数据处理的方法。
// 构造方法的第一个参数标识了AudioTrack对象
AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
: Thread(bCanCallJava), mReceiver(receiver)
{
}
// 线程的执行方法会调用mReciver的processAudioBuffer方法处理缓冲的数据
bool AudioTrack::AudioTrackThread::threadLoop()
{
return mReceiver.processAudioBuffer(this);
}
开始音频输出处理
-
通常在创建了AudioTrack并设置了参数后，会调用start方法进行音频输出处理，该方法主要有以下作用。
- 开启本地的AudioTrackThread线程，用于pull方式。
-
调用IAudioTrack接口的start方法，用于push方式。
- 最终执行Track类的start方法通知底层输出数据。
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
status_t status = NO_ERROR;
......
AutoMutex lock(mLock);
// 获取IAudioTrack接口
sp <IAudioTrack> audioTrack = mAudioTrack;
sp <IMemory> iMem = mCblkMemory;
// 获取缓冲区对象
audio_track_cblk_t* cblk = mCblk;
if (mActive == 0) {
......
// 执行AudioTrackThread线程
if (t != 0) {
t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
}
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
// 调用IAudioTrack::start方法
status = mAudioTrack->start();
cblk->lock.lock();
......
}
......
}
......
}
status_t AudioFlinger::PlaybackThread::Track::start()
{
status_t status = NO_ERROR;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
int state = mState;
if (mState == PAUSED) {
mState = TrackBase::RESUMING;
} else {
mState = TrackBase::ACTIVE;
}
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
// 通知HAL层开始音频输出
status = AudioSystem::startOutput(thread->id(),
(audio_stream_type_t)mStreamType,
mSessionId);
thread->mLock.lock();
......
}
if (status == NO_ERROR) {
// 将当前的Track加入活动列表
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->addTrack_l(this);
} else {
mState = state;
}
} else {
status = BAD_VALUE;
}
return status;
}
停止音频输出处理
-
播放结束后，还需要调用stop方法，该方法会被析构析函数析调用。
- 最终也会调用Track的stop方法停止输出数据。
void AudioTrack::stop()
{
sp<AudioTrackThread> t = mAudioTrackThread;
......
AutoMutex lock(mLock);
if (mActive == 1) {
mActive = 0;
mCblk->cv.signal();
// 调用IAudioTrack::stop方法
mAudioTrack->stop();
// 清空缓冲播放设置
setLoop_l(0, 0, 0);
mMarkerReached = false;
// 刷新位置以便AudioFlinger停止播放
if (mSharedBuffer != 0) {
flush_l();
}
......
}
......
}
void AudioFlinger::PlaybackThread::Track::stop()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
int state = mState;
if (mState > STOPPED) {
mState = STOPPED;
// 查找活动的Track
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
// 通知HAL层停止音频输出
AudioSystem::stopOutput(thread->id(),
(audio_stream_type_t)mStreamType,
mSessionId);
thread->mLock.lock();
......
}
}
}
音频数据处理(pull方式)
-
当使用pull方式时，AudioTrackThread才真正起作用，
mCbf是一个callback_t类型的回调，EVENT_MORE_DATA标识表示需要从用户那里获取数据。
// pull方式的回调
typedef void (*callback_t)(int event, void* user, void *info);
// 事件类型
enum event_type {
EVENT_MORE_DATA = 0, // 需要更多的数据
EVENT_UNDERRUN = 1, // 表示Audio硬件处于低负荷状态
EVENT_LOOP_END = 2, // 表示已经到达播放终点
EVENT_MARKER = 3, // 数据使用的警戒通知，且只会通知一次
EVENT_NEW_POS = 4, // 数据使用进度通知，以帧为单位
EVENT_BUFFER_END = 5 // 数据全部被消耗
};
- 音频数据的处理通过共享内存来实现，而这个共享对象由AudioFlinger端创建，这个共享对象被抽象为audio_track_cblk_t。
-
音频的数据缓冲区通过
obtainBuffer方法获取，再通过releaseBuffer方法释放。 -
在处理音频数据时需要考虑
underrun状态，该状态是指生产者提供数据的速度跟不上消费者使用数据的速度。一般遇到这种情况的处理方法是暂停输出，等数据准备好后再恢复输出。
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
uint32_t frames;
size_t writtenSize;
......
// 处理underrun的情况
if (mActive && (cblk->framesAvailable() == cblk->frameCount)) {
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
// server是读位置，frameCount是buffer中的数据总和。
// 当读位置等于数据总和时，表示数据已经用完了。
if (cblk->server == cblk->frameCount) {
mCbf(EVENT_BUFFER_END, mUserData, 0);
}
if (mSharedBuffer != 0) return false;
}
}
// 处理循环结束的情况
while (mLoopCount > cblk->loopCount) {
int loopCount = -1;
mLoopCount--;
if (mLoopCount >= 0) loopCount = mLoopCount;
// 一次循环播放完毕，loopCount表示还剩多少次
mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
}
// 处理阀值阀情况
if (!mMarkerReached && (mMarkerPosition > 0)) {
if (cblk->server >= mMarkerPosition) {
// 如果数据的使用超过警戒值，则通知用户
mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
// 只通知一次，因为该值被设置为true
mMarkerReached = true;
}
}
// 处理位置更新
if (mUpdatePeriod > 0) {
while (cblk->server >= mNewPosition) {
// 以帧数为单位，进行通知
mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
mNewPosition += mUpdatePeriod;
}
}
......
do {
audioBuffer.frameCount = frames;
// 获取一块可写的缓冲区
status_t err = obtainBuffer(&audioBuffer, waitCount);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
return false;
}
break;
}
if (err == status_t(STOPPED)) return false;
// 将缓冲区大小除2，以便处理PCM8格式
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
audioBuffer.size >>= 1;
}
// 从用户那里pull数据
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
writtenSize = audioBuffer.size;
// Sanity check on returned size
if (ssize_t(writtenSize) <= 0) {
// The callback is done filling buffers
// Keep this thread going to handle timed events and
// still try to get more data in intervals of WAIT_PERIOD_MS
// but don't just loop and block the CPU, so wait
usleep(WAIT_PERIOD_MS*1000);
break;
}
if (writtenSize > reqSize) writtenSize = reqSize;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// 8 to 16 bit conversion
const int8_t *src = audioBuffer.i8 + writtenSize-1;
int count = writtenSize;
int16_t *dst = audioBuffer.i16 + writtenSize-1;
while(count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// PCM8数据转PCM16
audioBuffer.frameCount = writtenSize/mCblk->frameSize;
frames -= audioBuffer.frameCount;
// 写完后释放缓冲区
releaseBuffer(&audioBuffer);
}
while (frames);
if (frames == 0) {
mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
return true;
}
音频数据处理(push方式)
- push方式是由用户主动调用write写数据，这相当于数据被push到AudioTrack。
-
在音频数据的处理pull方式中，我们看见了在处理线程中会一直
obtainBuffer，然后再releaseBuffer。这些数据是由回调函数产生。那么在push方式中，就需要向外提供一个write方法，用来获取用户的音频数据。
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
return BAD_VALUE;
}
......
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
size_t frameSz = (size_t)frameSize();
do {
// 以帧为单位处理数据
audioBuffer.frameCount = userSize/frameSz;
// 获取空闲的共享内存数据
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
if (err == status_t(NO_MORE_BUFFERS))
break;
return ssize_t(err);
}
size_t toWrite;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
// PCM8转换PCM16
int count = toWrite;
int16_t *dst = (int16_t *)(audioBuffer.i8);
while(count--) {
*dst++ = (int16_t)(*src++^0x80) << 8;
}
} else {
// 通过memcpy拷贝数据
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
src += toWrite;
}
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
} while (userSize >= frameSz);
return written;
}
音频缓冲区对象
-
首先为了跨进程处理，这个缓冲区需要是一块共享内存，它由audio_track_cblk_t类实现。回到前面“创建IAudioTrack对象”一节，这个缓冲区对象是通过
track->getCblk()获得。 - AudioTrack端称为缓冲区数据的生产者，负责向缓冲区中写数据。AudioFlinger端则是缓冲区数据的消费者，负责读数据。
- 在跨进程的访问中，缓冲区对象同时也考虑了同步问题。
struct audio_track_cblk_t
struct audio_track_cblk_t
{
Mutex lock;
Condition cv; // 两个用于同步的变量
volatile uint32_t user; // 当前写位置(生产者写到什么位置)
volatile uint32_t server; // 当前读位置
uint32_t userBase;
uint32_t serverBase;
void* buffers; // 指向缓冲区的首地址
uint32_t frameCount; // 缓冲区的总大小，以Frame为单位
uint32_t loopStart; // 设置播放的起点
uint32_t loopEnd; // 设置播放的终点
int loopCount; // 循环播放的次数
volatile union {
uint16_t volume[2];
uint32_t volumeLR;
};
uint32_t sampleRate; // 采样率
uint8_t frameSize; // 一个Frame的大小
uint8_t pad1;
uint16_t bufferTimeoutMs;// Maximum cumulated timeout before restarting audioflinger
uint16_t waitTimeMs; // Cumulated wait time
uint16_t sendLevel;
volatile int32_t flags; // 控制标识，表示underrun或overrun状态
audio_track_cblk_t();
uint32_t stepUser(uint32_t frameCount); // 更新写位置
bool stepServer(uint32_t frameCount); // 更新读位置
void* buffer(uint32_t offset) const; // 返回可写空间的起始位置
uint32_t framesAvailable(); // 还剩多少空间可写
uint32_t framesAvailable_l();
uint32_t framesReady(); // 是否有可读数据
bool tryLock();
};
生产者如何使用缓冲区
- 在“音频数据的处理(pull或push方式)”一节中，都会调用AudioTrack::obtainBuffer方法，以该方法为线索，来分析数据是如何生产的。
-
对于生产者来说，缓冲区的使用主要是以下三个步骤：
-
调用
framesAvailable方法，获取当前可写的空间大小。 -
调用
buffer方法，获取当前可写空间的首地址，写入数据。 -
调用
stepUser方法，更新可写位置(被obtainBuffer调用)。
-
调用
AudioTrack::obtainBuffer
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
AutoMutex lock(mLock);
......
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
// 得到当前可写的空间大小
uint32_t framesAvail = cblk->framesAvailable();
......
if (framesAvail == 0) {
cblk->lock.lock();
goto start_loop_here;
while (framesAvail == 0) {
......
if (!(cblk->flags & CBLK_INVALID_MSK)) {
mLock.unlock();
// 如果没有可写空间，则需要等待
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
cblk->lock.unlock();
mLock.lock();
if (mActive == 0) {
return status_t(STOPPED);
}
cblk->lock.lock();
}
......
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
cblk->lock.unlock();
}
......
// user为可写空间的起始地址
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - u;
}
......
// 调用buffer，得到可写空间的首地址
audioBuffer->raw = (int8_t *)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
AudioTrack::releaseBuffer
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
AutoMutex lock(mLock);
// 更新写位置
mCblk->stepUser(audioBuffer->frameCount);
}
audio_track_cblk_t::framesAvailable
uint32_t audio_track_cblk_t::framesAvailable()
{
Mutex::Autolock _l(lock);
return framesAvailable_l();
}
audio_track_cblk_t::framesAvailable_l
uint32_t audio_track_cblk_t::framesAvailable_l()
{
uint32_t u = this->user; // 当前写者位置
uint32_t s = this->server; // 当前读者位置
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
return limit + frameCount - u;
} else {
return frameCount + u - s;
}
}
audio_track_cblk_t::buffer
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
// buffers是缓冲区的起始位置，offset是对于userBase的偏移，这种方式可将
// 缓冲区当作一个环形缓冲来处理
return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
}
audio_track_cblk_t::stepUser
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
uint32_t u = this->user;
// 加上写入的帧数
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
if (flags & CBLK_DIRECTION_MSK) {
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
} else if (u > this->server) {
u = this->server;
}
// 更新当前写位置的同时，也更新userBase，这样(offset-userBase)就可以回到缓冲区头。
if (u >= userBase + this->frameCount) {
userBase += this->frameCount;
}
// 更新当前写的位置
this->user = u;
return u;
}
消费者如何使用缓冲区
-
在AudioFlinger一节我们介绍了共享内存的创建是在
AudioFlinger::ThreadBase::TrackBase::TrackBase中，而音频数据的消费者实际上是其继承类Track。它主要使用以下三个方法：-
getNextBuffer：通过frameReady得到可读帧数。 -
getBuffer：函数根据可读帧数信息得到可读空间的首地址。 -
releaseBuffer：通过stepServer更新读信息。
-
- 我们以DirectOutputThread的线程处理为线索分析消费者如何使用缓冲区。
AudioFlinger::DirectOutputThread::threadLoop
bool AudioFlinger::DirectOutputThread::threadLoop()
{
......
while (frameCount) {
buffer.frameCount = frameCount;
// 获取可读帧数
activeTrack->getNextBuffer(&buffer);
if (UNLIKELY(buffer.raw == 0)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
frameCount -= buffer.frameCount;
curBuf += buffer.frameCount * mFrameSize;
// 更新读信息
activeTrack->releaseBuffer(&buffer);
}
......
}
AudioFlinger::PlaybackThread::Track::getNextBuffer
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
// 获取audio_track_cblk_t对象
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
// 音频输出缓冲区大小，单位为帧数
uint32_t framesReq = buffer->frameCount;
......
// 计算有多少帧可读
framesReady = cblk->framesReady();
if (LIKELY(framesReady)) {
uint32_t s = cblk->server;
// 可读的最大位置
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
// 根据起始位置得到数据缓冲的起始地址
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
AudioFlinger::ThreadBase::TrackBase::releaseBuffer
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
}
AudioFlinger::ThreadBase::TrackBase::step
bool AudioFlinger::ThreadBase::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
result = cblk->stepServer(mFrameCount);
if (!result) {
mFlags |= STEPSERVER_FAILED;
}
return result;
}
audio_track_cblk_t::stepServer
bool audio_track_cblk_t::stepServer(uint32_t frameCount)
{
if (!tryLock()) {
return false;
}
// 获取当前读位置
uint32_t s = this->server;
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
}
// It is possible that we receive a flush()
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
// we have s > u
if (s > this->user) {
s = this->user;
}
}
// 处理读取到末尾的情况
if (s >= loopEnd) {
s = loopStart;
if (--loopCount == 0) {
loopEnd = UINT_MAX;
loopStart = UINT_MAX;
}
}
if (s >= serverBase + this->frameCount) {
serverBase += this->frameCount;
}
// 更新当前读位置
this->server = s;
if (!(flags & CBLK_INVALID_MSK)) {
cv.signal();
}
lock.unlock();
return true;
}