一、概述
音频是几乎是任何一个机器都是必备的一项功能，从早起的单纯发声的录音机，到后来的MP3，以及到现在的手机，它一直陪伴在我们的生活中，功能不变，形式却一直在变，包括它的架构也在变化。从早期的OSS到现在的ALSA，这个介绍在上篇文档是有介绍的，这里我们就着重说一下ALSA。首先高通的音频结构分为以下几个部分：
- 应用层，主要使用音频的用户主体
- 架构层(framework)，这一层主要是为应用层提供了相关处理接口，并且链接了HAL层
- 硬件抽象(HAL)层，在音频开发中可能大部分主要逻辑都是放在这个层次来处理，链接了 framework层和kernel层，这里面还包含了ALSA库用于链接ALSA驱动
- 内核(kernel)层，链接硬件的驱动程序
- 硬件，包含MODEM，CODEC，ADSP
接下来分别介绍一下几个层，除了应用层
1.1 framework层
声音的播放方式有两种：MediaPlayer和Audiotrack，MediaPlayer能够播放多种格式的声音文件，比如MP3，AAC，WAV，OGG，MIDI等。MediaPlayer包含了AudioTrack。
AudioTrack仅仅能播放已经解码的PCM流。假设是文件的话仅仅支持wav格式的音频文件，由于wav格式的音频文件大部分都是PCM流。AudioTrack不创建解码器。所以仅仅能播放不须要解码的wav文件。
先从AudioTrack这方面来看，主要以下三个Java类开始：
- AudioTrack.java 用于播放音频
- AudioRecord.java 用于录制音频
- AudioSystem.java 用于控制系统各种音频属性
他们的代码分别调用走向如下;
AuidoTrack.java → android_media_AudioTrack.cpp → AudioTrack.cpp → AudioFlinger.cpp
AudioRecord.java → android_media_AudioRecord.cpp → AudioRecord.cpp → AudioFlinger.cpp
AuidoSystem.java → android_media_AudioSystem.cpp → AudioSystem.cpp → AudioFlinger.cpp
从上面三个类的代码走向可以看出，他们最终都会走向同一个类AudioFlinger，它是音频管理器，我们也可以把它理解为代理，它代理了音频上层来的所有事物。它可以创造真正用于功能的track和recorder，那我们以track为例探究一下他是怎么走通这条路的。
首先了解一件事情，上层是怎么使用audiotrack 进行声音播放的，AudioTrack 有两种数据加载模式MODE_STREAM和MODE_STATIC，分别表示数据流加载和音频流类型，MODE_STREAM是通过不断的往AudioTrack内部buffer拷贝数据，这种方式会有一定的延迟，通常用来播放比较大的音频文件，MODE_STATIC则是把所有数据一次写入AudioTrack的内部缓冲区中，后续不再传入数据，这种模式通常用于占用内存小的文件播放。
- 新建一个Auiotrack对象，设置一些音频的参数
- 调用.write写数据
- .play进行播放
第一步，新建AudioTrack对象时，AudioTrack.java本身有三个重载的构造方法，分别有不同的参数，但是最终多会用到参数最多的那个，看下它的代码做了什么
AudioTrack.java
private AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId, boolean offload, int encapsulationMode,
@Nullable TunerConfiguration tunerConfiguration)
throws IllegalArgumentException {
super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
// mState already == STATE_UNINITIALIZED
......
// native initialization
int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes, ------> 1
sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/,
offload, encapsulationMode, tunerConfiguration,
getCurrentOpPackageName());
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing AudioTrack.");
return; // with mState == STATE_UNINITIALIZED
}
......
baseRegisterPlayer(mSessionId);
native_setPlayerIId(mPlayerIId); // mPlayerIId now ready to send to native AudioTrack. ------> 2
我们着重看下1、2处，分别调用了native层的两个函数，即android_media_AudioTrack.cpp中，1 native_setup做了一些初始化的事情，新建了AudioTrack对象，并且根据不同的加载模式调用了AudioTrack set函数进行了不同的参数设置，2 纯粹设置了PLAYER ID
AudioTrack的构建函数中，并没有做太多事情，反而set函数中做了比较多的事情，看看set做了什么，set函数有两个重载参数差异，我们直接看最后调用的set
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,
int32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
const AttributionSourceState& attributionSource,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
{
mThreadCanCallJava = threadCanCallJava;
mSelectedDeviceId = selectedDeviceId;
mSessionId = sessionId;
......
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
......
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
......
mFormat = format;
......
mChannelMask = channelMask;
channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
......
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
......
mAuxEffectId = 0;
mOrigFlags = mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
{
AutoMutex lock(mLock);
status = createTrack_l();
}
............
}
可以从上述截取的部分代码看出对一些重要的属性进行了设置，并且启动了一个重要线程AudioTrackTread，这个线程的作用后续会讲到。接着调用了createTrack_l()，这个函数主要是调用AudioFlinger.creaTrack()创建了一个Track，并且把它相关的属性保存在这边
status_t AudioTrack::createTrack_l()
{
status_t status;
bool callbackAdded = false;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 获取AudioFlinger
if (audioFlinger == 0) {
ALOGE("%s(%d): Could not get audioflinger",
__func__, mPortId);
status = NO_INIT;
goto exit;
}
......
IAudioFlinger::CreateTrackInput input; 组成input参数，这是要传递到AuioFlinger侧
if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
// Legacy: This is based on original parameters even if the track is recreated.
input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
} else {
input.attr = mAttributes;
}
input.config = AUDIO_CONFIG_INITIALIZER;
......
input.sharedBuffer = mSharedBuffer; 共享内存，这里是int值，表示的是共享内存的地址
input.notificationsPerBuffer = mNotificationsPerBufferReq;
......
media::CreateTrackResponse response;
status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
IAudioFlinger::CreateTrackOutput output{
};
if (status == NO_ERROR) {
output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response)); 通过AIDL跨进程通信方式将参数传递到AudioFlinger并且返回值
}
......
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
// FIXME compare to AudioRecord
std::optional<media::SharedFileRegion> sfr;
output.audioTrack->getCblk(&sfr); 这里获得的audioTrack其实是个代理，获取共享内存
sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
if (iMem == 0) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
status = NO_INIT;
goto exit;
}
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
void *iMemPointer = iMem->unsecurePointer();
if (iMemPointer == NULL) {
ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
status = NO_INIT;
goto exit;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
mAudioTrack = output.audioTrack; 保存audiotrack和共享内存
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
......
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
// is the value of pointer() for the shared buffer, otherwise buffers points
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
if (mSharedBuffer == 0) {
buffers = cblk + 1;
} else {
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
buffers = mSharedBuffer->unsecurePointer();
if (buffers == NULL) {
ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
status = NO_INIT;
goto exit;
}
}
mAudioTrack->attachAuxEffect(mAuxEffectId, &status); 绑定音效
.......
// update proxy
if (mSharedBuffer == 0) {
根据是否使用共享内存创建了两个不同的client代理，这两个代理会与AudioFlinger中的server相配对
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy = mStaticProxy;
}
mProxy->setVolumeLR(gain_minifloat_pack(
gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
mProxy->setSendLevel(mSendLevel);
const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
mProxy->setSampleRate(effectiveSampleRate);
AudioPlaybackRate playbackRateTemp = mPlaybackRate;
playbackRateTemp.mSpeed = effectiveSpeed;
playbackRateTemp.mPitch = effectivePitch;
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setMinimum(mNotificationFramesAct);
......
mDeathNotifier = new DeathNotifier(this);
IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
......
}
第二步，write 数据，在AudioTrack.java中有好几个write的重载方法，分别用于写不同的数据类型：byte，short，float，long，nativebytes等，到android_media_AudioTrack.cpp中发现除了nativebytes其他都是指向同一个函数android_media_AudioTrack_writeArray
android_media_AudioTrack.cpp
template <typename T>
static jint android_media_AudioTrack_writeArray(JNIEnv *env, jobject thiz,
T javaAudioData,
jint offsetInSamples, jint sizeInSamples,
jint javaAudioFormat,
jboolean isWriteBlocking) {
......
jint samplesWritten = writeToTrack(lpTrack, javaAudioFormat, cAudioData,
offsetInSamples, sizeInSamples, isWriteBlocking == JNI_TRUE /* blocking */);
envReleaseArrayElements(env, javaAudioData, cAudioData, 0);
//ALOGV("write wrote %d (tried %d) samples in the native AudioTrack with offset %d",
// (int)samplesWritten, (int)(sizeInSamples), (int)offsetInSamples);
return samplesWritten;
}
nativbytes指向了函数 android_media_AudioTrack_write_native_bytes
android_media_AudioTrack.cpp
static jint android_media_AudioTrack_write_native_bytes(JNIEnv *env, jobject thiz,
jobject javaByteBuffer, jint byteOffset, jint sizeInBytes,
jint javaAudioFormat, jboolean isWriteBlocking) {
......
if (bytes == NULL) {
ALOGE("Error retrieving source of audio data to play, can't play");
return (jint)AUDIO_JAVA_BAD_VALUE;
}
jint written = writeToTrack(lpTrack, javaAudioFormat, bytes, byteOffset,
sizeInBytes, isWriteBlocking == JNI_TRUE /* blocking */);
return written;
}
可以看出两个函数虽然名字参数不一样但是最终都调用了writeToTrack函数，这个函数会根据AuioTrack.cpp 的shardBuffer，采用不同的方式写数据，shareBuffer是否为零根据上层使用的是MODE_STREAM还是MODE_STATIC，MODE_STATIC是会有sharedBuffer，然后写数据是直接写入sharedBuffer中，而MODE_STEREAM则是调用AudiaoTrack的write方式。
android_media_AudioTrack.cpp
// ----------------------------------------------------------------------------
class AudioTrackJniStorage {
public:
sp<MemoryHeapBase> mMemHeap;
sp<MemoryBase> mMemBase;
audiotrack_callback_cookie mCallbackData{
};
sp<JNIDeviceCallback> mDeviceCallback;
sp<JNIAudioTrackCallback> mAudioTrackCallback;
bool allocSharedMem(int sizeInBytes) {
mMemHeap = new MemoryHeapBase(sizeInBytes, 0, "AudioTrack Heap Base");
if (mMemHeap->getHeapID() < 0) {
return false;
}
mMemBase = new MemoryBase(mMemHeap, 0, sizeInBytes);
return true;
}
};
AudioTrackJniStorage则是持有和管理AudioTrack共享内存的类，是在android_media_AudioTrack中进行初始化申请内存然后通过AudioTrack set() 函数传递，双方进行使用。
AudioTrack.cpp
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
......
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;
}
从AudioTrack::write 函数来看，这里是向ClientProxy申请了一个共享的buffer，然后把数据拷贝给这个buffer，可见stream模式最终也是通过共享内存来交换数据，最后在数据拷贝成功后，会唤醒AudioTrackThread 线程，这个线程他会将目前会循环的去获取当前的播放状态和事件，并且返回给上层
第三步，上层开始play
AudioTrack.java
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(); -------- 1
} else {
new Thread() {
public void run() {
try {
Thread.sleep(delay);
} catch (InterruptedException e) {
e.printStackTrace();
}
baseSetStartDelayMs(0);
try {
startImpl(); ------------ 1
} 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();
}
}
private void startImpl() {
------------- 1
synchronized (mRoutingChangeListeners) {
if (!mEnableSelfRoutingMonitor) {
mEnableSelfRoutingMonitor = testEnableNativeRoutingCallbacksLocked();
}
}
synchronized(mPlayStateLock) {
baseStart(0); // unknown device at this point
native_start(); ------------ 2
// FIXME see b/179218630
//baseStart(native_getRoutedDeviceId());
if (mPlayState == PLAYSTATE_PAUSED_STOPPING) {
mPlayState = PLAYSTATE_STOPPING;
} else {
mPlayState = PLAYSTATE_PLAYING;
mOffloadEosPending = false;
}
}
}
在上层AudioTrack.java 调用play()之后，没有做太多的操作就是一个延迟开始和不延迟，实际上用到了native层native_start()，它对应android_media_AudioTrack.cpp中的android_media_AudioTrack_start()函数，我们直接看android_media_AudioTrack_start()
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();
}
直接跨度到了AudioTrack.cpp start()函数，这个start（）函数比较长就不贴出来，主要做的事情就是将动作传递到了AudioFlinger那一侧的Track中，上面三步目前都已经走到了AudioFlinger 这里了，AudioFlinger到底做了什么，怎么来的，接下来就揭开其神秘面纱。
AudioFlinger:
首先从audioserver开始，audioserver是放在framework/av/media下面，它只有三个文件Andoird.bp，audioserver.rc，main_audioserver.cpp，从文件组成来看这个模块是开机就开始运行了，audioserver.rc 在开机的时候会被系统的init进程所加载，接着启动audioserver，也就是main_audioserver.cpp，然后在audioserver的main函数中就启动了AudioFlinger、AudioPolicyService，分别调用了它们的instantiate()函数，
AudioFlinger.cpp
void AudioFlinger::instantiate() {
sp<IServiceManager> sm(defaultServiceManager());
sm->addService(String16(IAudioFlinger::DEFAULT_SERVICE_NAME), -------- 1
new AudioFlingerServerAdapter(new AudioFlinger()), false,
IServiceManager::DUMP_FLAG_PRIORITY_DEFAULT);
}
AudioFlinger::AudioFlinger()
: mMediaLogNotifier(new AudioFlinger::MediaLogNotifier()),
mPrimaryHardwareDev(NULL),
mAudioHwDevs(NULL),
mHardwareStatus(AUDIO_HW_IDLE),
mMasterVolume(1.0f),
mMasterMute(false),
// mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false),
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
mTotalMemory(0),
mClientSharedHeapSize(kMinimumClientSharedHeapSizeBytes),
mGlobalEffectEnableTime(0),
mPatchPanel(this),
mDeviceEffectManager(this),
mSystemReady(false)
{
// Move the audio session unique ID generator start base as time passes to limit risk of
// generating the same ID again after an audioserver restart.
// This is important because clients will reuse previously allocated audio session IDs
// when reconnecting after an audioserver restart and newly allocated IDs may conflict with
// active clients.
// Moving the base by 1 for each elapsed second is a good compromise between avoiding overlap
// between allocation ranges and not reaching wrap around too soon.
timespec ts{
};
clock_gettime(CLOCK_MONOTONIC, &ts);
// zero ID has a special meaning, so start allocation at least at AUDIO_UNIQUE_ID_USE_MAX
uint32_t movingBase = (uint32_t)std::max((long)1, ts.tv_sec);
// unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
mNextUniqueIds[use] =
((use == AUDIO_UNIQUE_ID_USE_SESSION || use == AUDIO_UNIQUE_ID_USE_CLIENT) ?
movingBase : 1) * AUDIO_UNIQUE_ID_USE_MAX;
}
#if 1
// FIXME See bug 165702394 and bug 168511485
const bool doLog = false;
#else
const bool doLog = property_get_bool("ro.test_harness", false);
#endif
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
(void) pthread_once(&sMediaLogOnce, sMediaLogInit);
}
// reset battery stats.
// if the audio service has crashed, battery stats could be left
// in bad state, reset the state upon service start.
BatteryNotifier::getInstance().noteResetAudio();
mDevicesFactoryHal = DevicesFactoryHalInterface::create(); -------- 2
mEffectsFactoryHal = EffectsFactoryHalInterface::create(); -------- 3
mMediaLogNotifier->run("MediaLogNotifier");
std::vector<pid_t> halPids;
mDevicesFactoryHal->getHalPids(&halPids);
TimeCheck::setAudioHalPids(halPids);
// Notify that we have started (also called when audioserver service restarts)
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CTOR)
.record();
}
AuidoFlinger 的初始化，先是把自己包装之后加入到了servicemanager中，这里主要用于native 层通过binder和AudioFlinger进行数据交换，然后是初始化了两个很重要的接口，DevicesFactoryHalInterface和EffectsFactoryHalInterface，这两个接口是用于连接HAL层的主要接口类，负责与HAL层的数据交换，这里与HAL层的数据交换同样使用的Binder机制，学名HIDL一种跨进程通信方式，那先来看下他们做了什么，以DevicesFactoryHalInterface为例，窥探一下DevicesFactoryHalInterface.create后面的代码
DevicesFactoryHalInterface.cpp
// static
sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
这里create 直接链接到下一个类的函数createPreferredImpl,传递了两个参数一个是包名另一个是AIDL 的代理类
return createPreferredImpl<DevicesFactoryHalInterface>(
"android.hardware.audio", "IDevicesFactory");
}
--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
FactoryHalHidl.cpp
void* createPreferredImpl(const std::string& package, const std::string& interface) {
这里显示遍历了一个保存HAL版本的数组，并判断当前处于哪个版本，如果找到有这个版本则创建这个版本的代理接口
for (auto version = detail::sAudioHALVersions; version != nullptr; ++version) {
遍历HAL版本数组
void* rawInterface = nullptr;
if (hasHalService(package, *version, interface) 判断该版本号对应的HAL是否存在
&& createHalService(*version, interface, &rawInterface)) {
创建对应版本的HAL代理接口
return rawInterface;
}
}
return nullptr;
}
bool createHalService(const std::string& version, const std::string& interface,
void** rawInterface) {
const std::string libName = "libaudiohal@" + version + ".so"; 这里组装了一个libaudiohal@x.x.so字符串，一个库，x.x就是刚才遍历到的版本储存在sAudioHALVersions中
const std::string factoryFunctionName = "create" + interface; 刚才穿进来的interface 是IDevicesFactory，这里组装成了createIDevicesFactory，像是一个函数
constexpr int dlMode = RTLD_LAZY;
void* handle = nullptr;
dlerror(); // clear
handle = dlopen(libName.c_str(), dlMode); 加载库libaudiohal@x.x.so
if (handle == nullptr) {
const char* error = dlerror();
ALOGE("Failed to dlopen %s: %s", libName.c_str(),
error != nullptr ? error : "unknown error");
return false;
}
void* (*factoryFunction)();
*(void **)(&factoryFunction) = dlsym(handle, factoryFunctionName.c_str()); 获取库libaudiohal@x.x.so中的createIDevicesFactory函数指针
if (!factoryFunction) {
const char* error = dlerror();
ALOGE("Factory function %s not found in library %s: %s",
factoryFunctionName.c_str(), libName.c_str(),
error != nullptr ? error : "unknown error");
dlclose(handle);
return false;
}
*rawInterface = (*factoryFunction)(); 这里则调用了createIDevicesFactory函数并返回了rawInterface，这个rawInterface最终是要返回到调用DevicesFactoryHalInterface.create的地方
ALOGW_IF(!*rawInterface, "Factory function %s from %s returned nullptr",
factoryFunctionName.c_str(), libName.c_str());
return true;
}
通过上面几段代码的一个跟踪和分析，发现DevicesFactoryHalInterface.create其实只是去打开了一个[email protected]的库，并且调用了它的函数createIDevicesFactory获取了其返回值，通过查找代码发现这个库的源码是在framewrok/av/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
DevicesFactoryHalHybrid.cpp
extern "C" __attribute__((visibility("default"))) void* createIDevicesFactory() {
这个函数的调用实际上是构建了DevicesFactoryHalHybrid并且把它返回给了调用者
auto service = hardware::audio::CPP_VERSION::IDevicesFactory::getService();
return service ? new CPP_VERSION::DevicesFactoryHalHybrid(service) : nullptr;
}
DevicesFactoryHalHybrid::DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory) DevicesFactoryHalHybrid构建初始化了两个类DevicesFactoryHalLocal和DevicesFactoryHalHidl从名字看一个是本地一个是HIDL代理者
: mLocalFactory(new DevicesFactoryHalLocal()),
mHidlFactory(new DevicesFactoryHalHidl(hidlFactory)) {
}
status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
打开HAL层的一些设备，这里设备有几种类型：primary,a2dp,usb,r_submix,stub，我们主要使用的是primary
if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
return mHidlFactory->openDevice(name, device);
}
return mLocalFactory->openDevice(name, device);
}
status_t DevicesFactoryHalHybrid::getHalPids(std::vector<pid_t> *pids) {
if (mHidlFactory != 0) {
return mHidlFactory->getHalPids(pids);
}
return INVALID_OPERATION;
}
status_t DevicesFactoryHalHybrid::setCallbackOnce(sp<DevicesFactoryHalCallback> callback) {
if (mHidlFactory) {
return mHidlFactory->setCallbackOnce(callback);
}
return INVALID_OPERATION;
}
从DevicesFactoryHalHybrid.cpp上面的代码可以看出，它的主要处理是交给了mHidlFactory也就是DevicesFactoryHalHidl。
介绍完了AudioFlinger的初始化，我们再看一下上层调用的三步到达AudioFlinger之后做了什么
第一步，代码到达了AudioFlinger.createTrack
status_t AudioFlinger::createTrack(const media::CreateTrackRequest& _input,
media::CreateTrackResponse& _output)
{
// Local version of VALUE_OR_RETURN, specific to this method's calling conventions.
CreateTrackInput input = VALUE_OR_RETURN_STATUS(CreateTrackInput::fromAidl(_input));
CreateTrackOutput output;
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
status_t lStatus;
audio_stream_type_t streamType;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
std::vector<audio_io_handle_t> secondaryOutputs;
......
output.sessionId = sessionId;
output.outputId = AUDIO_IO_HANDLE_NONE;
output.selectedDeviceId = input.selectedDeviceId;
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
adjAttributionSource, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs); ------------------ 1
......
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output.outputId); 获取playback数组
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output.outputId);
lStatus = BAD_VALUE;
goto Exit;
}
......
ALOGV("createTrack() sessionId: %d", sessionId);
output.sampleRate = input.config.sample_rate;
output.frameCount = input.frameCount;
output.notificationFrameCount = input.notificationFrameCount;
output.flags = input.flags;
output.streamType = streamType;
track = thread->createTrack_l(client, streamType, localAttr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, &output.notificationFrameCount,
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
callingPid, adjAttributionSource, input.clientInfo.clientTid,
&lStatus, portId, input.audioTrackCallback);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
output.afFrameCount = thread->frameCount();
output.afSampleRate = thread->sampleRate();
output.afLatencyMs = thread->latency();
output.portId = portId;
......
setAudioHwSyncForSession_l(thread, sessionId);
}
......
output.audioTrack = new TrackHandle(track);
_output = VALUE_OR_FATAL(output.toAidl());
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(portId);
}
return lStatus;
}
整个createTrack代码很长总结来说就是先获取传过来的参数input ，根据input在调用PlaybackThread创建Track，然后返回output，并且返回的audioTrack并不是真的Track而是TrackHandle代理，从名字看这是一个回放的线程，它的代码非常的长，主要目的就是创建Track，并且保存在了一个数组中，在上面代码中标记1处，这个函数是获取我们的Output，它首先是通过AudioSystem中转调用AudioPolicyInterfaceImpl中的getOutputForAttr，然后再到AudioPolicyManager的getOutputForAttrInt，直接看AudioPolicyManager
status_t AudioPolicyManager::getOutputForAttrInt(
audio_attributes_t *resultAttr,
audio_io_handle_t *output,
audio_session_t session,
const audio_attributes_t *attr,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
bool *isRequestedDeviceForExclusiveUse,
std::vector<sp<AudioPolicyMix>> *secondaryMixes,
output_type_t *outputType)
{
DeviceVector outputDevices;
const audio_port_handle_t requestedPortId = *selectedDeviceId;
DeviceVector msdDevices = getMsdAudioOutDevices();
const sp<DeviceDescriptor> requestedDevice =
mAvailableOutputDevices.getDeviceFromId(requestedPortId);
*outputType = API_OUTPUT_INVALID;
status_t status = getAudioAttributes(resultAttr, attr, *stream);
if (status != NO_ERROR) {
return status;
}
if (auto it = mAllowedCapturePolicies.find(uid); it != end(mAllowedCapturePolicies)) {
resultAttr->flags = static_cast<audio_flags_mask_t>(resultAttr->flags | it->second);
}
*stream = mEngine->getStreamTypeForAttributes(*resultAttr);
ALOGV("%s() attributes=%s stream=%s session %d selectedDeviceId %d", __func__,
toString(*resultAttr).c_str(), toString(*stream).c_str(), session, requestedPortId);
// The primary output is the explicit routing (eg. setPreferredDevice) if specified,
// otherwise, fallback to the dynamic policies, if none match, query the engine.
// Secondary outputs are always found by dynamic policies as the engine do not support them
sp<AudioPolicyMix> primaryMix;
status = mPolicyMixes.getOutputForAttr(*resultAttr, uid, *flags, primaryMix, secondaryMixes);
if (status != OK) {
return status;
}
// Explicit routing is higher priority then any dynamic policy primary output
bool usePrimaryOutputFromPolicyMixes = requestedDevice == nullptr && primaryMix != nullptr;
// FIXME: in case of RENDER policy, the output capabilities should be checked
if ((usePrimaryOutputFromPolicyMixes
|| (secondaryMixes != nullptr && !secondaryMixes->empty()))
&& !audio_is_linear_pcm(config->format)) {
ALOGD("%s: rejecting request as dynamic audio policy only support pcm", __func__);
return BAD_VALUE;
}
if (usePrimaryOutputFromPolicyMixes) {
sp<DeviceDescriptor> deviceDesc =
mAvailableOutputDevices.getDevice(primaryMix->mDeviceType,
primaryMix->mDeviceAddress,
AUDIO_FORMAT_DEFAULT);
sp<SwAudioOutputDescriptor> policyDesc = primaryMix->getOutput();
if (deviceDesc != nullptr
&& (policyDesc == nullptr || (policyDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT))) {
audio_io_handle_t newOutput;
status = openDirectOutput( --------------------- 1
*stream, session, config,
(audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT),
DeviceVector(deviceDesc), &newOutput);
if (status != NO_ERROR) {
policyDesc = nullptr;
} else {
policyDesc = mOutputs.valueFor(newOutput);
primaryMix->setOutput(policyDesc);
}
}
if (policyDesc != nullptr) {
policyDesc->mPolicyMix = primaryMix;
*output = policyDesc->mIoHandle;
*selectedDeviceId = deviceDesc != 0 ? deviceDesc->getId() : AUDIO_PORT_HANDLE_NONE;
ALOGV("getOutputForAttr() returns output %d", *output);
if (resultAttr->usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
*outputType = API_OUT_MIX_PLAYBACK;
} else {
*outputType = API_OUTPUT_LEGACY;
}
return NO_ERROR;
}
}
// Virtual sources must always be dynamicaly or explicitly routed
if (resultAttr->usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
return BAD_VALUE;
}
// explicit routing managed by getDeviceForStrategy in APM is now handled by engine
// in order to let the choice of the order to future vendor engine
outputDevices = mEngine->getOutputDevicesForAttributes(*resultAttr, requestedDevice, false);
if ((resultAttr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
*flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
// Set incall music only if device was explicitly set, and fallback to the device which is
// chosen by the engine if not.
// FIXME: provide a more generic approach which is not device specific and move this back
// to getOutputForDevice.
// TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side.
if (outputDevices.onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_TELEPHONY_TX) &&
(*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
audio_is_linear_pcm(config->format) &&
isCallAudioAccessible()) {
if (requestedPortId != AUDIO_PORT_HANDLE_NONE) {
*flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
*isRequestedDeviceForExclusiveUse = true;
}
}
ALOGV("%s() device %s, sampling rate %d, format %#x, channel mask %#x, flags %#x stream %s",
__func__, outputDevices.toString().c_str(), config->sample_rate, config->format,
config->channel_mask, *flags, toString(*stream).c_str());
*output = AUDIO_IO_HANDLE_NONE;
if (!msdDevices.isEmpty()) {
*output = getOutputForDevices(msdDevices, session, *stream, config, flags); ------------ 2
if (*output != AUDIO_IO_HANDLE_NONE && setMsdOutputPatches(&outputDevices) == NO_ERROR) {
ALOGV("%s() Using MSD devices %s instead of devices %s",
__func__, msdDevices.toString().c_str(), outputDevices.toString().c_str());
} else {
*output = AUDIO_IO_HANDLE_NONE;
}
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = getOutputForDevices(outputDevices, session, *stream, config, ------------- 3
flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
}
if (*output == AUDIO_IO_HANDLE_NONE) {
return INVALID_OPERATION;
}
*selectedDeviceId = getFirstDeviceId(outputDevices);
for (auto &outputDevice : outputDevices) {
if (outputDevice->getId() == getConfig().getDefaultOutputDevice()->getId()) {
*selectedDeviceId = outputDevice->getId();
break;
}
}
if (outputDevices.onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_TELEPHONY_TX)) {
*outputType = API_OUTPUT_TELEPHONY_TX;
} else {
*outputType = API_OUTPUT_LEGACY;
}
ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId);
return NO_ERROR;
}
整个这些代码目的就一个获取output，上面代码标记的三处分别对应的不同情况，最终都是要用到openDirectOutput，看一下openDirectOutput
status_t AudioPolicyManager::openDirectOutput(audio_stream_type_t stream,
audio_session_t session,
const audio_config_t *config,
audio_output_flags_t flags,
const DeviceVector &devices,
audio_io_handle_t *output) {
*output = AUDIO_IO_HANDLE_NONE;
// skip direct output selection if the request can obviously be attached to a mixed output
// and not explicitly requested
if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX &&
audio_channel_count_from_out_mask(config->channel_mask) <= 2) {
return NAME_NOT_FOUND;
}
// Do not allow offloading or direct if one non offloadable effect is enabled or
// MasterMono is enabled. This prevents creating an offloaded or direct track
// and tearing it down immediately after start when audioflinger detects there
// is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
sp<IOProfile> profile;
if (((flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT)) == 0) ||
!(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
profile = getProfileForOutput(
devices, config->sample_rate, config->format, config->channel_mask,
flags, true /* directOnly */);
}
if (profile == nullptr) {
return NAME_NOT_FOUND;
}
if (!(flags & AUDIO_OUTPUT_FLAG_DIRECT) &&
(profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
ALOGI("%s rejecting direct profile as was not requested ", __func__);
profile = nullptr;
return NAME_NOT_FOUND;
}
sp<SwAudioOutputDescriptor> outputDesc = nullptr;
// check if direct output for pcm/track offload or compress offload already exist
bool directSessionInUse = false;
bool offloadSessionInUse = false;
// exclusive outputs for MMAP and Offload are enforced by different session ids.
if (!(property_get_bool("vendor.audio.offload.multiple.enabled", false) &&
((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0)) {
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
outputDesc = desc;
// reuse direct output if currently open by the same client
// and configured with same parameters
if ((config->sample_rate == desc->getSamplingRate()) &&
(config->format == desc->getFormat()) &&
(config->channel_mask == desc->getChannelMask()) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
ALOGI("%s reusing direct output %d for session %d", __func__,
mOutputs.keyAt(i), session);
*output = mOutputs.keyAt(i);
return NO_ERROR;
}
if (desc->mFlags == AUDIO_OUTPUT_FLAG_DIRECT) {
directSessionInUse = true;
ALOGV("%s Direct PCM already in use", __func__);
}
if (desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
offloadSessionInUse = true;
ALOGV("%s Compress Offload already in use", __func__);
}
}
}
if (outputDesc != nullptr &&
((flags == AUDIO_OUTPUT_FLAG_DIRECT && directSessionInUse) ||
((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && offloadSessionInUse))) {
if (session != outputDesc->mDirectClientSession) {
ALOGV("getOutput() do not reuse direct pcm output because current client (%d) "
"is not the same as requesting client (%d) for different output conf",
outputDesc->mDirectClientSession, session);
return NAME_NOT_FOUND;
} else {
ALOGV("%s close previous output on same client session %d ", __func__, session);
closeOutput(outputDesc->mIoHandle);
}
}
}
if (!profile->canOpenNewIo()) {
return NAME_NOT_FOUND;
} 上面是处理一些找不到的情况，进行一个过滤
outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface); ---------- 新建一个SwAudioOutputDescriptor，这个是包装output的类
// An MSD patch may be using the only output stream that can service this request. Release
// all MSD patches to prioritize this request over any active output on MSD.
releaseMsdOutputPatches(devices);
status_t status = outputDesc->open(config, devices, stream, flags, output); ----------这里才是真正的打开output
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
(config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
(config->channel_mask != 0 && config->channel_mask != outputDesc->getChannelMask())) {
ALOGV("%s failed opening direct output: output %d sample rate %d %d,"
"format %d %d, channel mask %04x %04x", __func__, *output, config->sample_rate,
outputDesc->getSamplingRate(), config->format, outputDesc->getFormat(),
config->channel_mask, outputDesc->getChannelMask());
if (*output != AUDIO_IO_HANDLE_NONE) {
outputDesc->close();
}
// fall back to mixer output if possible when the direct output could not be open
if (audio_is_linear_pcm(config->format) &&
config->sample_rate <= SAMPLE_RATE_HZ_MAX) {
return NAME_NOT_FOUND;
}
*output = AUDIO_IO_HANDLE_NONE;
return BAD_VALUE;
}
outputDesc->mDirectOpenCount = 1;
outputDesc->mDirectClientSession = session;
addOutput(*output, outputDesc);
mPreviousOutputs = mOutputs;
ALOGV("%s returns new direct output %d", __func__, *output);
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
}
上面新建的SwAudioOutputDescriptor调用open之后会经过AudioPolicyClientImpl到达Audioflinger的openoutput
status_t AudioFlinger::openOutput(const media::OpenOutputRequest& request,
media::OpenOutputResponse* response)
{
audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_module_handle_t(request.module));
audio_config_t config = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfig_audio_config_t(request.config));
sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
aidl2legacy_DeviceDescriptorBase(request.device));
audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_output_flags_t_mask(request.flags));
audio_io_handle_t output;
uint32_t latencyMs;
ALOGI("openOutput() this %p, module %d Device %s, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
this, module,
device->toString().c_str(),
config.sample_rate,
config.format,
config.channel_mask,
flags);
audio_devices_t deviceType = device->type();
const String8 address = String8(device->address().c_str());
if (deviceType == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags); openoutput核心处理在这里
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
latencyMs = playbackThread->latency();
// notify client processes of the new output creation
playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
// the first primary output opened designates the primary hw device if no HW module
// named "primary" was already loaded.
AutoMutex lock(mHardwareLock);
if ((mPrimaryHardwareDev == nullptr) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d as the primary audio interface", module);
mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev;
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
MmapThread *mmapThread = (MmapThread *)thread.get();
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
response->flags = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
return NO_ERROR;
}
return NO_INIT;
}
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags)
{
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, deviceType); 寻找并加载对应的HAL层module，也就是链接HAL侧接口
if (outHwDev == NULL) {
return 0;
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
} else {
// Audio Policy does not currently request a specific output handle.
// If this is ever needed, see openInput_l() for example code.
ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
return 0;
}
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
// FOR TESTING ONLY:
// This if statement allows overriding the audio policy settings
// and forcing a specific format or channel mask to the HAL/Sink device for testing.
if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
//config->format = AUDIO_FORMAT_PCM_FLOAT;
//config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
//config->format = AUDIO_FORMAT_PCM_32_BIT;
//config->format = AUDIO_FORMAT_PCM_8_24_BIT;
// ALOGV("openOutput_l() upgrading format to %#08x", config->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
//config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
//config->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
AudioStreamOut *outputStream = NULL;
status_t status = outHwDev->openOutputStream(
&outputStream,
*output,
deviceType,
flags,
config,
address.string()); 这里调用HAL层的openoutputStream正式打通和HAL层的通道
mHardwareStatus = AUDIO_HW_IDLE;
if (status == NO_ERROR) {
下面就对应几种情况分别建立了playbackThread
if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
sp<MmapPlaybackThread> thread =
new MmapPlaybackThread(this, *output, outHwDev, outputStream, mSystemReady); ------------- 1
mMmapThreads.add(*output, thread);
ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p",
*output, thread.get());
return thread;
} else {
sp<PlaybackThread> thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, mSystemReady); -------------- 2
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, mSystemReady); --------------- 3
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
} else {
thread = new MixerThread(this, outputStream, *output, mSystemReady); --------------- 4
ALOGV("openOutput_l() created mixer output: ID %d thread %p",
*output, thread.get());
}
mPlaybackThreads.add(*output, thread); 添加到了数组中
struct audio_patch patch;
mPatchPanel.notifyStreamOpened(outHwDev, *output, &patch);
if (thread->isMsdDevice()) {
thread->setDownStreamPatch(&patch);
}
return thread;
}
}
return 0;
}
至此，mPlaybackThreads的神秘面纱就揭开了，它所保存的几种thread 分别有不用的场景应用，具体如下图
看完playbackthread接着往下看Track的创建
AudioFlinger::PlaybackThread::Track::Track(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
audio_output_flags_t flags,
track_type type,
audio_port_handle_t portId,
size_t frameCountToBeReady,
float speed)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
(sharedBuffer != 0) ? sharedBuffer->unsecurePointer() : buffer,
(sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
sessionId, creatorPid,
VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid)), true /*isOut*/,
(type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
type,
portId,
std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(portId)), 这个地方调用了父类的构造函数，利用父类做了一些事情
mFillingUpStatus(FS_INVALID),
// mRetryCount initialized later when needed
mSharedBuffer(sharedBuffer),
mStreamType(streamType),
mMainBuffer(thread->sinkBuffer()),
mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false),
mFrameMap(16 /* sink-frame-to-track-frame map memory */),
mVolumeHandler(new media::VolumeHandler(sampleRate)),
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(attributionSource, attr, id(),
streamType)),
// mSinkTimestamp
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
* When the track starts playing, its volume will be computed. */
mFinalVolume(0.f),
mResumeToStopping(false),
mFlushHwPending(false),
mFlags(flags),
mSpeed(speed)
{
持有了各种状态属性
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
ALOGV_IF(sharedBuffer != 0, "%s(%d): sharedBuffer: %p, size: %zu",
__func__, mId, sharedBuffer->unsecurePointer(), sharedBuffer->size());
if (mCblk == NULL) {
return;
}
uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
ALOGE("%s(%d): no more tracks available", __func__, mId);
releaseCblk(); // this makes the track invalid.
return;
}
if (sharedBuffer == 0) {
根据是否有共享内存创建了两个server代理，和AudioTrack中的client相配对
mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack(), sampleRate);
} else {
mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, sampleRate);
}
mServerProxy = mAudioTrackServerProxy;
mServerProxy->setStartThresholdInFrames(frameCountToBeReady); // update the Cblk value
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
// FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
// race with setSyncEvent(). However, if we call it, we cannot properly start
// static fast tracks (SoundPool) immediately after stopping.
//mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
int i = __builtin_ctz(thread->mFastTrackAvailMask);
ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
// FIXME This is too eager. We allocate a fast track index before the
// fast track becomes active. Since fast tracks are a scarce resource,
// this means we are potentially denying other more important fast tracks from
// being created. It would be better to allocate the index dynamically.
mFastIndex = i;
thread->mFastTrackAvailMask &= ~(1 << i);
}
mServerLatencySupported = thread->type() == ThreadBase::MIXER
|| thread->type() == ThreadBase::DUPLICATING;
#ifdef TEE_SINK
mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+ "_" + std::to_string(mId) + "_T");
#endif
if (thread->supportsHapticPlayback()) {
// If the track is attached to haptic playback thread, it is potentially to have
// HapticGenerator effect, which will generate haptic data, on the track. In that case,
// external vibration is always created for all tracks attached to haptic playback thread.
mAudioVibrationController = new AudioVibrationController(this);
std::string packageName = attributionSource.packageName.has_value() ?
attributionSource.packageName.value() : "";
mExternalVibration = new os::ExternalVibration(
mUid, packageName, mAttr, mAudioVibrationController);
}
// Once this item is logged by the server, the client can add properties.
const char * const traits = sharedBuffer == 0 ? "" : "static";
mTrackMetrics.logConstructor(creatorPid, uid, id(), traits, streamType);
}
第二步，写数据，这里是将上层要播放的数据写进了共享内存里，这里的关键方法是obtainBuffer，这个函数是用来获取共享内存的，没有太多操作不细讲
第三步，play进行到了AudioTrak.start 调用了AudioFlinger侧的TrackHandle start，这只是一个传递代码真正是到了Track start()，然后看看Track start 做了什么
status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
audio_session_t triggerSession __unused)
{
status_t status = NO_ERROR;
ALOGV("%s(%d): calling pid %d session %d",
__func__, mId, IPCThreadState::self()->getCallingPid(), mSessionId);
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;
// here the track could be either new, or restarted
// in both cases "unstop" the track
// initial state-stopping. next state-pausing.
// What if resume is called ?
if (state == FLUSHED) {
// avoid underrun glitches when starting after flush
reset();
}
// 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;
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(); 获取playbackThread
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); 这里是监测underrun状态
}
status = playbackThread->addTrack_l(this); 将当前track加入到playbackThread中
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();
}
// 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;
}
上面的代码我们主要看playbackThread，我们播放音频的发动机，主要核心线程，playbackThread是保存在mPlaybackThreads数组中，再看看playbackThread→addTrack_l
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
if (mActiveTracks.indexOf(track) < 0) {
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
if (track->isExternalTrack()) {
TrackBase::track_state state = track->mState;
mLock.unlock();
status = AudioSystem::startOutput(track->portId()); 这边实际上是调用到了AudioOutputDescriptor中，增加了mPorfile->curActiveCount的计数
mLock.lock();
// abort track was stopped/paused while we released the lock
if (state != track->mState) {
if (status == NO_ERROR) {
mLock.unlock();
AudioSystem::stopOutput(track->portId());
mLock.lock();
}
return INVALID_OPERATION;
}
// abort if start is rejected by audio policy manager
if (status != NO_ERROR) {
return PERMISSION_DENIED;
}
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
#endif
sendIoConfigEvent_l(AUDIO_CLIENT_STARTED, track->creatorPid(), track->portId());
}
// set retry count for buffer fill
if (track->isOffloaded()) {
if (track->isStopping_1()) {
track->mRetryCount = kMaxTrackStopRetriesOffload;
} else {
track->mRetryCount = kMaxTrackStartupRetriesOffload;
}
track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
} else {
track->mRetryCount = kMaxTrackStartupRetries;
track->mFillingUpStatus =
track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
}
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (mHapticChannelMask != AUDIO_CHANNEL_NONE
&& ((track->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
|| (chain != nullptr && chain->containsHapticGeneratingEffect_l()))) {
// Unlock due to VibratorService will lock for this call and will
// call Tracks.mute/unmute which also require thread's lock.
mLock.unlock();
const int intensity = AudioFlinger::onExternalVibrationStart(
track->getExternalVibration());
mLock.lock();
track->setHapticIntensity(static_cast<os::HapticScale>(intensity));
// Haptic playback should be enabled by vibrator service.
if (track->getHapticPlaybackEnabled()) {
// Disable haptic playback of all active track to ensure only
// one track playing haptic if current track should play haptic.
for (const auto &t : mActiveTracks) {
t->setHapticPlaybackEnabled(false);
}
}
// Set haptic intensity for effect
if (chain != nullptr) {
chain->setHapticIntensity_l(track->id(), intensity);
}
}
track->mResetDone = false;
track->resetPresentationComplete();
mActiveTracks.add(track);
if (chain != 0) {
ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
track->sessionId());
chain->incActiveTrackCnt();
}
track->logBeginInterval(patchSinksToString(&mPatch)); // log to MediaMetrics
status = NO_ERROR;
}
onAddNewTrack_l(); ----------- 1
return status;
}
void AudioFlinger::PlaybackThread::onAddNewTrack_l() 1 ------------- 这里是对我们playbackThread进行了唤醒
{
ALOGV("signal playback thread");
broadcast_l();
}
addTrack_l之后playbackThread会被唤醒，然后就会开始它自己的工作，thread_loop会动起来，这里面包含集合函数:threadloop_mix、threadloop_write、threadloop_standby，分别是对音频进行混音、通过数据流将数据写到HAL层、暂停，上层play之后就会开始往HAL层写数据，然后就放出了了我们听到的音乐，这个track在framework层的使命就算是成功完成了，然后处理的事情就交给了HAL层和kernel及ALSA