Android Framework学习笔记 -- Audio的播放流程
流程图
这是基于Android5.1分析的，前几版本好像有些不同，6.0没改太多，不过大体思想是一致的
播放就像个排水机，AuidoPolicyService是阀门，AudioFlinger是排水池，PlaybackThread是发动机，Track是源，AudioOutput是排水孔。AudioTrack是水桶
排水首先要凿个孔（openOutput），然后添加发动机（建立PlaybackThread），然后将源接到水桶上（建立Track），选择排水孔（selectOutptu），开启相应的发动机（PlaybackThread从睡眠中唤醒），然后就各自排水了。。。。
AudioTrack服务端的启动及准备
服务端的指的是AudioFlinger跟AudioPolicyService等音频相关的服务，这些服务会在系统开机的时候启动
在系统启动完成后，客户端（一般都是app）就能利用这些服务来使用系统提供的功能
Audio相关的服务启动
开机时系统启动各种服务，AudioFlinger跟AudioPolicyService和一些音频相关的服务会在此启动。
各服务的Instantiate()函数在BinderService.h中定义实现，主要是用于抽象出注册服务的操作。
BinderService是个模板类，服务继承该类后可以直接注册到systemserver。
//--->frameworks/av/media/mediaserver/main_mediaserver.cpp
int main(int argc __unused, char** argv)
{
...
AudioFlinger::instantiate();
MediaPlayerService::instantiate();
AudioPolicyService::instantiate();
...
}
//--->frameworks/native/include/binder/BinderService.h
template
class BinderService
{
public:
static status_t publish(bool allowIsolated = false) {
sp sm(defaultServiceManager());
// 这里用模板生成了具体服务的对象
// new SERVICE()将会调用服务(AudioFlinger，AudioPolicyService等)的构造函数
return sm->addService(
String16(SERVICE::getServiceName()),
new SERVICE(), allowIsolated);
}
...
static void instantiate() { publish(); }
...
};
AudioFlinger的创建
AudioFlinger承担混音工作。（总之很重要啦）
AudioFlinger的构造函数主要是对成员变量和调试工具的初始化。
onFirstRef一般做进一步的初始化工作，AudioFlinger暂时没有在该函数中做重要的工作。
//--->frameworks/av/services/audioflinger.cpp
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mPrimaryHardwareDev(NULL),
mAudioHwDevs(NULL),
mHardwareStatus(AUDIO_HW_IDLE),
mMasterVolume(1.0f),
mMasterMute(false),
mNextUniqueId(1),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false),
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
mGlobalEffectEnableTime(0),
mPrimaryOutputSampleRate(0)
{
...
#ifdef TEE_SINK
....
#endif
...
}
void AudioFlinger::onFirstRef()
{
Mutex::Autolock _l(mLock);
...
mPatchPanel = new PatchPanel(this);
mMode = AUDIO_MODE_NORMAL;
}
AudioPolicyService的创建
AudioPolicyService用于控制音频播放策略（比如插耳机的时候来电用什么设备去播放音乐）、管理音频设备等
AudioPolicyService的构造函数更简单，只是初始化主要成员。
AudioPolicyService会在onFristRef中做比较多的工作，比如创建command线程，初始化重要成员mAudioPolicyManager。
//--->frameworks/av/services/audiopolicy/AudioPolicyService.cpp
AudioPolicyService::AudioPolicyService()
: BnAudioPolicyService(),
mpAudioPolicyDev(NULL),
mpAudioPolicy(NULL),
mAudioPolicyManager(NULL),
mAudioPolicyClient(NULL),
mPhoneState(AUDIO_MODE_INVALID)
{}
void AudioPolicyService::onFirstRef()
{
...
{
Mutex::Autolock _l(mLock);
mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this);
mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this);
mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);
#ifdef USE_LEGACY_AUDIO_POLICY
...(暂时这宏意义不明)
#else
mAudioPolicyClient = new AudioPolicyClient(this);
mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient);
#endif
}
...(效果相关)
}
AudioPolicyManager的创建
AudioPolicyManager作为音频调度策略的实现，在AudioPolicyService关于音频调度的基本都是直接转发给AudioPolicyManager。
（貌似可以重载AudioPolicyManager来改动音频策略的实现,6.0开始可以直接动态选择不同的AudiPolicyManger实现）
在构造函数中，打开了所有能用的音频设备和录音设备，并调用AudioPolicyService创建了相应设备的混音线程。
//--->frameworks/av/services/audiopolicy/AudioPolicyManager.cpp
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:mPrimaryOutput((audio_io_handle_t)0),
...
{
mpClientInterface = clientInterface;
...
// 加载音频模块
defaultAudioPolicyConfig();
...
for (size_t i = 0; i < mHwModules.size(); i++) {
...
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
...
status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle, // 打开音频设备
&output,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
}
}
...(打开录音设备)
}
mpClientInterface就是AudioPolicyService，AudioPolicyService最终会调用AudioFlinger的openOutput函数。
//--->frameworks/av/services/audiopolicy/AudioPolicyClientImpl.cpp
status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
sp af = AudioSystem::get_audio_flinger();
if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
return af->openOutput(module, output, config, devices, address, latencyMs, flags);
}
AudioFlinger的openOutput中会针对输出设备的类型创建了一个PlaybackThread。
PlaybackThread在AudioFlinger相当重要，相当于音频系统的发动机
PlaybackThread有几种，比较常见有MixerThread，蓝牙耳机设备需要外放（比如ring类型的流需要同时从耳机与喇叭出来）的时候使用DuplicatingThread。
//--->frameworks/av/services/audioflinger/AudioFlinger.cpp
status_t AudioFlinger::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
uint32_t *latencyMs,
audio_output_flags_t flags){
...
sp thread = openOutput_l(module, output, config, *devices, address, flags);
...
}
sp AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_output_flags_t flags)
{
...
status_t status = hwDevHal->open_output_stream(
hwDevHal,
*output,
devices,
flags,
config,
&outStream,
address.string());
...
// 根据flags 创建相应的thread
AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags);
PlaybackThread *thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, devices);
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, devices);
} else {
thread = new MixerThread(this, outputStream, *output, devices);
}
...
}
PlaybackThread创建完之后在onFirstRef()中会自己启动起来,并会在threadLoop等待一个AudioTrack的连接。
//--->frameworks/av/services/audioflinger/Threads.cpp
AudioFlinger::MixerThread::MixerThread(const sp& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, type_t type)
: PlaybackThread(audioFlinger, output, id, device, type),
mFastMixerFutex(0)
{...(主要初始化了Mixer)}
AudioFlinger::PlaybackThread::PlaybackThread(const sp& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
type_t type)
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
...
{...(初始化了音量相关的参数和获取输出设备的参数)}
// 在此PlaybackThread会跑起来，运行threadLoop()
void AudioFlinger::PlaybackThread::onFirstRef()
{
run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
}
// threadLoop是整个AudioFlinger的核心，混音的工作在此进行
bool AudioFlinger::PlaybackThread::threadLoop()
{
...
cacheParameters_l();
...
checkSilentMode_l();
while (!exitPending())
{
...
processConfigEvents_l();
...
size_t size = mActiveTracks.size();
for (size_t i = 0; i < size; i++) {
sp 到此各种线程服务准备完毕，可以播放AudioTrack了
客户端播放AudioTrack
Android SDK向外提供了MediaPlayer和比较底层的AudioTrack接口，MediaPlayer做一些解码的工作，最终还是会使用到AudioTrack。
AudioTrack的构建
调用从app开始，首先是AudioTrack(java)的构造函数,会调用jni的native_setup
//--->frameworks/base/media/java/android/media/AudioTrack.java
public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId)throws IllegalArgumentException {
...(参数检查)
mStreamType = AudioSystem.STREAM_DEFAULT;
int[] session = new int[1];
session[0] = sessionId;
// native initialization
int initResult = native_setup(new WeakReference(this), mAttributes,
mSampleRate, mChannels, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session);
...
}
jni对应的函数是android_media_AudioTrack_setup，生成一个AudioTrack(c++)并设置其参数
//--->android/frameworks/base/core/jni/android_media_AudioTrack.cpp
static jint android_media_AudioTrack_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jobject jaa,
jint sampleRateInHertz, jint javaChannelMask,
jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession) {
...(检查参数)
sp lpTrack = new AudioTrack();
...
// 看样子很重要
AudioTrackJniStorage* lpJniStorage = new AudioTrackJniStorage();
// 分不同的模式设置track
switch (memoryMode) {
case MODE_STREAM:
status = lpTrack->set(...);
break;
case MODE_STATIC:
// AudioTrack is using shared memory
status = lpTrack->set(...);
break;
}
...(错误处理)
}
AudioTrack的无参构造函数很简单，主要工作还是放在set里面
//--->frameworks/av/media/libmedia/AudioTrack.cpp
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
mAttributes.flags = 0x0;
strcpy(mAttributes.tags, "");
}
AudioTrack的set的工作很多，服务端的track其实是在此时建立的。
//--->frameworks/av/media/libmedia/AudioTrack.cpp
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
uint32_t notificationFrames,
const sp& sharedBuffer,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes)
{
...(设置参数等)
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
}
status_t status = createTrack_l();
...(错误处理)
}
这里插入一个track选择output的一个过程,在工作中经常碰到这个鬼东西。在创建track的时候其实音频的路由已经定好了，之前还一直以为在start后选，在createTrack之前，会调用getOutputForAttr来获取当前的流对应的output（后面有空会补一下output跟device及stream的乱七八糟的关系）
//--->frameworks/av/media/libmedia/AudioTrack.cpp
status_t AudioTrack::createTrack_l()
{
const sp& audioFlinger = AudioSystem::get_audio_flinger();
...(计算FrameCount，Latency等
// 这里的output即到底层的通路，可以用adb shell dumpsys media.audio_policy 看到所有的output
status_t status = AudioSystem::getOutputForAttr(attr, &output,
(audio_session_t)mSessionId, &streamType, mClientUid,
mSampleRate, mFormat, mChannelMask,
mFlags, mSelectedDeviceId, mOffloadInfo);
sp track = audioFlinger->createTrack(streamType, //创建服务端的track
mSampleRate,
format,
mChannelMask,
&temp,
&trackFlags,
mSharedBuffer,
output,
tid,
&mSessionId,
mClientUid,
&status);
...(debug代码等)
// AudioTrackClientProxy主要实现管理cblk和服务端通信
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
...
}
// AudioSystem转发给AudiopolicyService
//--->frameworks/av/media/libmedia/AudioSystem.cpp
status_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
audio_port_handle_t selectedDeviceId,
const audio_offload_info_t *offloadInfo)
{
const sp& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return NO_INIT;
return aps->getOutputForAttr(attr, output, session, stream, uid,
samplingRate, format, channelMask,
flags, selectedDeviceId, offloadInfo);
}
// AudioPolicyService 转发给AudioPolicyManager
//--->/frameworks/av/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
audio_port_handle_t selectedDeviceId,
const audio_offload_info_t *offloadInfo)
{
...
if (IPCThreadState::self()->getCallingPid() != getpid_cached || uid == (uid_t)-1) {
uid_t newclientUid = IPCThreadState::self()->getCallingUid();
if (uid != (uid_t)-1 && uid != newclientUid) {
ALOGW("%s uid %d tried to pass itself off as %d", __FUNCTION__, newclientUid, uid);
}
uid = newclientUid;
}
return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, uid, samplingRate,
format, channelMask, flags, selectedDeviceId, offloadInfo);
}
// 这里涉及到了流和策略及output的选择，还是比较重要的吧
//--->frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
audio_port_handle_t selectedDeviceId,
const audio_offload_info_t *offloadInfo)
{
...(流类型判断)
// 获取路由策略
routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
// 获取策略对应的设备
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
...(判断flag)
// 获取对应设备的output
*output = getOutputForDevice(device, session, *stream,
samplingRate, format, channelMask,
flags, offloadInfo);
...
return NO_ERROR;
}
接下来的服务端AudioFlinger会调用PlaybackThread的createTrack_l创建Track。
//--->frameworks/av/services/audioflinger/AudioFlinger.cpp
sp AudioFlinger::createTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
const sp& sharedBuffer,
audio_io_handle_t output,
pid_t tid,
int *sessionId,
int clientUid,
status_t *status)
{
...(参数检查)
// 根据output选择对应的thread
PlaybackThread *thread = checkPlaybackThread_l(output);
...
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
...
// return handle to client
trackHandle = new TrackHandle(track);
Exit:
*status = lStatus;
return trackHandle;
}
PlaybackThread会创建Track的实体对象
//--->android/frameworks/av/services/audioflinger/Threads.cpp
sp AudioFlinger::PlaybackThread::createTrack_l(
const sp& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
const sp& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int uid,
status_t *status)
{
...(设置参数)
if (!isTimed) {
track = new Track(this, client, streamType, sampleRate, format,
channelMask, frameCount, NULL, sharedBuffer,
sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
} else {
track = TimedTrack::create(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId, uid);
}
...
}
(TODO这样应有Track构建的分析)
到这一步，客户端跟服务端的track都创建好了，就等着播放了。
AudioTrack的播放
AudioTrack(java)会调用play进行音频播放的准备。
//--->frameworks/base/media/java/android/media/AudioTrack.java
public void play()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("play() called on uninitialized AudioTrack.");
}
if (isRestricted()) {
setVolume(0);
}
synchronized(mPlayStateLock) {
native_start();
mPlayState = PLAYSTATE_PLAYING;
}
}
play()会调用jni的native_start(),对应的函数是android_media_AudioTrack_start(), android_media_AudioTrack_start()只是做了转发，最后会调用 AudioTrack(c++)的start(), AudioTrack的start又会调用TrackHandle(服务端的Track代理) 的start，最后会调用到服务端Track的start。
//--->frameworks/av/media/libmedia/AudioTrack.cpp
status_t AudioTrack::start()
{
...(标记位和参数的检测)
status = mAudioTrack->start();
...
}
TrackHandle仅仅做过转发,最后会触发PlaybackThread的addTrack_l()
//--->frameworks/av/services/audioflinger/Tracks.cpp
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
int triggerSession __unused)
{
...
status = playbackThread->addTrack_l(this);
...
}
PlaybackTread的addTrack_l主要工作是添加track到mActiveTracks,并激活沉睡的PlaybackTread。
//--->frameworks/av/services/audioflinger/Threads.cpp
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp接下来就是写入音频数据 AudioTrack.java会调用write写入音频数据(播放声音)
//--->frameworks/base/media/java/android/media/AudioTrack.java
public int write(byte[] audioData, int offsetInBytes, int sizeInBytes) {
int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat,
true /*isBlocking*/);
}
native_write_byte()会调用jni的android_media_AudioTrack_write_byte，会调用jni的android_media_AudioTrack_write_byte，
主要是获取java传下来的数据， 并调用writeToTrack来向共享内存写入数据，writeToTrack又分track是否为stream或static来做不同的处理。
//--->android/frameworks/base/core/jni/android_media_AudioTrack.cpp
static jint android_media_AudioTrack_write_byte(JNIEnv *env, jobject thiz,
jbyteArray javaAudioData,
jint offsetInBytes, jint sizeInBytes,
jint javaAudioFormat,
jboolean isWriteBlocking)
{
...
cAudioData = (jbyte *)env->GetByteArrayElements(javaAudioData, NULL);
...
jint written = writeToTrack(lpTrack, javaAudioFormat, cAudioData, offsetInBytes, sizeInBytes,
isWriteBlocking == JNI_TRUE /* blocking */);
...
}
jint writeToTrack(const sp& track, jint audioFormat, const jbyte* data,
jint offsetInBytes, jint sizeInBytes, bool blocking = true) {
if (track->sharedBuffer() == 0) {
written = track->write(data + offsetInBytes, sizeInBytes, blocking);
} else {
...
switch (format) {
default:
case AUDIO_FORMAT_PCM_FLOAT:
case AUDIO_FORMAT_PCM_16_BIT: {
...
memcpy(track->sharedBuffer()->pointer(), data + offsetInBytes, sizeInBytes);
...
} break;
case AUDIO_FORMAT_PCM_8_BIT: {
...
memcpy_to_i16_from_u8(dst, src, count);
...
} break;
}
}
return written;
}
其中stream类型的track会调用AudioTrack(c++)的write,AudioTrack会使用obtainBuffer获取一块共享内存，
并写入数据，写完后用releaseBuffer释放共享内存。（就可以给AudioFlingr使用了）
//--->frameworks/av/media/libmedia/AudioTrack.cpp
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
...(参数检查)
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
status_t err = obtainBuffer(&audioBuffer,
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
...(错误处理)
...(memcpy buffer -> audioBuffer);
...(计算剩余数据)
releaseBuffer(&audioBuffer);
}
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
...（参数转换跟计算)
return obtainBuffer(audioBuffer, requested);
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed, size_t *nonContig)
{
...（参数转换）
status = proxy->obtainBuffer(&buffer, requested, elapsed);
...(结果的填充)
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
...
mProxy->releaseBuffer(&buffer);
...
}
obtainBuffer跟releaseBuffer的具体实现交给了AudioTrackClientProxy来实现，主要是管理cblk对象与共享内存。应该深入研究一下。
服务端读取共享内存的音频数据是在PlaybackThread的threadLoop()中进行的，MixerThread也使用该函数，
不过重写了threadLoop_mix()等关键函数(典型的多态)。
//--->frameworks/av/services/audioflinger/Threads.cpp
bool AudioFlinger::PlaybackThread::threadLoop()
{
...
cacheParameters_l();
...
acquireWakeLock();
...
checkSilentMode_l();
while (!exitPending()){
...(lock)
processConfigEvents_l();
...
saveOutputTracks();
...(wakelock wait,not understand,**mark**)
...
threadLoop_standby(); // 准备音频设备??
...(参数检测)
prepareTracks_l(&tracksToRemove);
...
if (mBytesRemaining == 0) {
mCurrentWriteLength = 0;
if (mMixerStatus == MIXER_TRACKS_READY) {
// threadLoop_mix() sets mCurrentWriteLength
threadLoop_mix(); // 混音
} ...(其他情况处理)
...(音效处理)
}
if (mBytesRemaining) {
ssize_t ret = threadLoop_write(); // 写到音频设备
if (ret < 0) {
mBytesRemaining = 0;
} else {
mBytesWritten += ret;
mBytesRemaining -= ret;
}
}...(其他情况处理)
// 播放完成，删除已经播放的tracks
threadLoop_removeTracks(tracksToRemove);
tracksToRemove.clear();
clearOutputTracks();
effectChains.clear();
}
threadLoop_exit();
...
releaseWakeLock();
mWakeLockUids.clear();
mActiveTracksGeneration++;
}
}
这里重点看一下 MixerThread的threadLoop_mix 和 threadLoop_write
threadLoop_mix调用了AudioMixer的process,threadLoop_write最终调用mOutput->stream->write写到驱动里去了
//--->frameworks/av/services/audioflinger/Threads.cpp
void AudioFlinger::MixerThread::threadLoop_mix()
{
...
// mix buffers...
mAudioMixer->process(pts);
...
}
ssize_t AudioFlinger::MixerThread::threadLoop_write()
{
...(处理一些fastmix的情况)
return PlaybackThread::threadLoop_write();
}
ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
...(一些Sink操作)
bytesWritten = mOutput->stream->write(mOutput->stream,
(char *)mSinkBuffer + offset, mBytesRemaining);
...
}
AudioMixer的process是一个hook函数，根据不同的情况会调用不同的函数。具体的调用会调用到AudioMixer中以process开头的一组函数。如
- void AudioMixer::process__validate(state_t* state, int64_t pts)
- void AudioMixer::process__nop(state_t* state, int64_t pts)
- void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
- void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
- void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,int64_t pts)
- void AudioMixer::process__OneTrack24BitsStereoNoResampling(state_t* state,int64_t pts)
- void AudioMixer::process_NoResampleOneTrack(state_t* state, int64_t pts)
这里以process__OneTrack16BitsStereoNoResampling为例子,在获取buffer的时候使用了Track的getNextBuffer和releaseBuffer。
//--->frameworks/av/services/audioflinger/AudioMixer.cpp
void AudioMixer::process(int64_t pts)
{
mState.hook(&mState, pts);
}
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts)
{
const int i = 31 - __builtin_clz(state->enabledTracks);
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);
...(音量设置)
while (numFrames) {
...(Frame的计算)
t.bufferProvider->getNextBuffer(&b, outputPTS); // 获取客户端写入的buffer
...(错误处理)
// 混音
switch (t.mMixerFormat) {
case AUDIO_FORMAT_PCM_FLOAT: ... break;
case AUDIO_FORMAT_PCM_16_BIT: ... break;
default:LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
}
...
t.bufferProvider->releaseBuffer(&b);
}
其他问题
这里只是简单的过了一下AudioTrack播放音频的流程，其他的地方还有很多东西要看。
- 共享内存的同步(AudioTrackClientProxy的obtainBuffer和releaseBuffer和bufferProvider的getNextBuffer和releaseBuffer);
- mOutput->stream->write最终的去向。
- AudioPolicyService的一些音频策略
- 上层的Mediaplayer的一些工作