一、簡介
Audio是多媒體子系統中的一個重要模組，其涉及的內容比較多，有音訊的渲染、音訊的採集、音訊的策略管理等。本文主要針對音訊渲染功能進行詳細地分析，並透過原始碼中提供的例子，對音訊渲染進行流程的梳理。
二、目錄foundation/multimedia/audio_framework
audio_framework
├── frameworks
│ ├── js #js 介面
│ │ └── napi
│ │ └── audio_renderer #audio_renderer NAPI介面
│ │ ├── include
│ │ │ ├── audio_renderer_callback_napi.h
│ │ │ ├── renderer_data_request_callback_napi.h
│ │ │ ├── renderer_period_position_callback_napi.h
│ │ │ └── renderer_position_callback_napi.h
│ │ └── src
│ │ ├── audio_renderer_callback_napi.cpp
│ │ ├── audio_renderer_napi.cpp
│ │ ├── renderer_data_request_callback_napi.cpp
│ │ ├── renderer_period_position_callback_napi.cpp
│ │ └── renderer_position_callback_napi.cpp
│ └── native #native 介面
│ └── audiorenderer
│ ├── BUILD.gn
│ ├── include
│ │ ├── audio_renderer_private.h
│ │ └── audio_renderer_proxy_obj.h
│ ├── src
│ │ ├── audio_renderer.cpp
│ │ └── audio_renderer_proxy_obj.cpp
│ └── test
│ └── example
│ └── audio_renderer_test.cpp
├── interfaces
│ ├── inner_api #native實現的介面
│ │ └── native
│ │ └── audiorenderer #audio渲染本地實現的介面定義
│ │ └── include
│ │ └── audio_renderer.h
│ └── kits #js呼叫的介面
│ └── js
│ └── audio_renderer #audio渲染NAPI介面的定義
│ └── include
│ └── audio_renderer_napi.h
└── services #服務端
└── audio_service
├── BUILD.gn
├── client #IPC呼叫中的proxy端
│ ├── include
│ │ ├── audio_manager_proxy.h
│ │ ├── audio_service_client.h
│ └── src
│ ├── audio_manager_proxy.cpp
│ ├── audio_service_client.cpp
└── server #IPC呼叫中的server端
├── include
│ └── audio_server.h
└── src
├── audio_manager_stub.cpp
└── audio_server.cpp
三、音訊渲染總體流程
四、Native介面使用
在OpenAtom OpenHarmony（以下簡稱“OpenHarmony”）系統中，音訊模組提供了功能測試程式碼，本文選取了其中的音訊渲染例子作為切入點來進行介紹，例子採用的是對wav格式的音訊檔案進行渲染。wav格式的音訊檔案是wav標頭檔案和音訊的原始資料，不需要進行資料解碼，所以音訊渲染直接對原始資料進行操作，檔案路徑為：foundation/multimedia/audio_framework/frameworks/native/audiorenderer/test/example/audio_renderer_test.cpp
bool TestPlayback(int argc, char *argv[]) const
{
FILE* wavFile = fopen(path, "rb");
//讀取wav檔案頭資訊
size_t bytesRead = fread(&wavHeader, 1, headerSize, wavFile);
//設定AudioRenderer引數
AudioRendererOptions rendererOptions = {};
rendererOptions.streamInfo.encoding = AudioEncodingType::ENCODING_PCM;
rendererOptions.streamInfo.samplingRate = static_cast<AudioSamplingRate>(wavHeader.SamplesPerSec);
rendererOptions.streamInfo.format = GetSampleFormat(wavHeader.bitsPerSample);
rendererOptions.streamInfo.channels = static_cast<AudioChannel>(wavHeader.NumOfChan);
rendererOptions.rendererInfo.contentType = contentType;
rendererOptions.rendererInfo.streamUsage = streamUsage;
rendererOptions.rendererInfo.rendererFlags = 0;
//建立AudioRender例項
unique_ptr<AudioRenderer> audioRenderer = AudioRenderer::Create(rendererOptions);
shared_ptr<AudioRendererCallback> cb1 = make_shared<AudioRendererCallbackTestImpl>();
//設定音訊渲染回撥
ret = audioRenderer->SetRendererCallback(cb1);
//InitRender方法主要呼叫了audioRenderer例項的Start方法，啟動音訊渲染
if (!InitRender(audioRenderer)) {
AUDIO_ERR_LOG("AudioRendererTest: Init render failed");
fclose(wavFile);
return false;
}
//StartRender方法主要是讀取wavFile檔案的資料，然後透過呼叫audioRenderer例項的Write方法進行播放
if (!StartRender(audioRenderer, wavFile)) {
AUDIO_ERR_LOG("AudioRendererTest: Start render failed");
fclose(wavFile);
return false;
}
//停止渲染
if (!audioRenderer->Stop()) {
AUDIO_ERR_LOG("AudioRendererTest: Stop failed");
}
//釋放渲染
if (!audioRenderer->Release()) {
AUDIO_ERR_LOG("AudioRendererTest: Release failed");
}
//關閉wavFile
fclose(wavFile);
return true;
}
首先讀取wav檔案，透過讀取到wav檔案的頭資訊對AudioRendererOptions相關的引數進行設定，包括編碼格式、取樣率、取樣格式、通道數等。根據AudioRendererOptions設定的引數來建立AudioRenderer例項（實際上是AudioRendererPrivate），後續的音訊渲染主要是透過AudioRenderer例項進行。建立完成後，呼叫AudioRenderer的Start方法，啟動音訊渲染。啟動後，透過AudioRenderer例項的Write方法，將資料寫入，音訊資料會被播放。
五、呼叫流程
- 建立AudioRenderer
std::unique_ptr<AudioRenderer> AudioRenderer::Create(const std::string cachePath,
const AudioRendererOptions &rendererOptions, const AppInfo &appInfo)
{
ContentType contentType = rendererOptions.rendererInfo.contentType;
StreamUsage streamUsage = rendererOptions.rendererInfo.streamUsage;
AudioStreamType audioStreamType = AudioStream::GetStreamType(contentType, streamUsage);
auto audioRenderer = std::make_unique<AudioRendererPrivate>(audioStreamType, appInfo);
if (!cachePath.empty()) {
AUDIO_DEBUG_LOG("Set application cache path");
audioRenderer->SetApplicationCachePath(cachePath);
}
audioRenderer->rendererInfo_.contentType = contentType;
audioRenderer->rendererInfo_.streamUsage = streamUsage;
audioRenderer->rendererInfo_.rendererFlags = rendererOptions.rendererInfo.rendererFlags;
AudioRendererParams params;
params.sampleFormat = rendererOptions.streamInfo.format;
params.sampleRate = rendererOptions.streamInfo.samplingRate;
params.channelCount = rendererOptions.streamInfo.channels;
params.encodingType = rendererOptions.streamInfo.encoding;
if (audioRenderer->SetParams(params) != SUCCESS) {
AUDIO_ERR_LOG("SetParams failed in renderer");
audioRenderer = nullptr;
return nullptr;
}
return audioRenderer;
}
首先透過AudioStream的GetStreamType方法獲取音訊流的型別，根據音訊流型別建立AudioRendererPrivate物件，AudioRendererPrivate是AudioRenderer的子類。緊接著對audioRenderer進行引數設定，其中包括取樣格式、取樣率、通道數、編碼格式。設定完成後返回建立的AudioRendererPrivate例項。
- 設定回撥
int32_t AudioRendererPrivate::SetRendererCallback(const std::shared_ptr<AudioRendererCallback> &callback)
{
RendererState state = GetStatus();
if (state == RENDERER_NEW || state == RENDERER_RELEASED) {
return ERR_ILLEGAL_STATE;
}
if (callback == nullptr) {
return ERR_INVALID_PARAM;
}
// Save reference for interrupt callback
if (audioInterruptCallback_ == nullptr) {
return ERROR;
}
std::shared_ptr<AudioInterruptCallbackImpl> cbInterrupt =
std::static_pointer_cast<AudioInterruptCallbackImpl>(audioInterruptCallback_);
cbInterrupt->SaveCallback(callback);
// Save and Set reference for stream callback. Order is important here.
if (audioStreamCallback_ == nullptr) {
audioStreamCallback_ = std::make_shared<AudioStreamCallbackRenderer>();
if (audioStreamCallback_ == nullptr) {
return ERROR;
}
}
std::shared_ptr<AudioStreamCallbackRenderer> cbStream =
std::static_pointer_cast<AudioStreamCallbackRenderer>(audioStreamCallback_);
cbStream->SaveCallback(callback);
(void)audioStream_->SetStreamCallback(audioStreamCallback_);
return SUCCESS;
}
引數傳入的回撥主要涉及到兩個方面：一方面是AudioInterruptCallbackImpl中設定了我們傳入的渲染回撥，另一方面是AudioStreamCallbackRenderer中也設定了渲染回撥。
- 啟動渲染
bool AudioRendererPrivate::Start(StateChangeCmdType cmdType) const
{
AUDIO_INFO_LOG("AudioRenderer::Start");
RendererState state = GetStatus();
AudioInterrupt audioInterrupt;
switch (mode_) {
case InterruptMode::SHARE_MODE:
audioInterrupt = sharedInterrupt_;
break;
case InterruptMode::INDEPENDENT_MODE:
audioInterrupt = audioInterrupt_;
break;
default:
break;
}
AUDIO_INFO_LOG("AudioRenderer::Start::interruptMode: %{public}d, streamType: %{public}d, sessionID: %{public}d",
mode_, audioInterrupt.streamType, audioInterrupt.sessionID);
if (audioInterrupt.streamType == STREAM_DEFAULT || audioInterrupt.sessionID == INVALID_SESSION_ID) {
return false;
}
int32_t ret = AudioPolicyManager::GetInstance().ActivateAudioInterrupt(audioInterrupt);
if (ret != 0) {
AUDIO_ERR_LOG("AudioRendererPrivate::ActivateAudioInterrupt Failed");
return false;
}
return audioStream_->StartAudioStream(cmdType);
}
AudioPolicyManager::GetInstance().ActivateAudioInterrupt這個操作主要是根據AudioInterrupt來進行音訊中斷的啟用，這裡涉及了音訊策略相關的內容，後續會專門出關於音訊策略的文章進行分析。這個方法的核心是透過呼叫AudioStream的StartAudioStream方法來啟動音訊流。
bool AudioStream::StartAudioStream(StateChangeCmdType cmdType)
{
int32_t ret = StartStream(cmdType);
resetTime_ = true;
int32_t retCode = clock_gettime(CLOCK_MONOTONIC, &baseTimestamp_);
if (renderMode_ == RENDER_MODE_CALLBACK) {
isReadyToWrite_ = true;
writeThread_ = std::make_unique<std::thread>(&AudioStream::WriteCbTheadLoop, this);
} else if (captureMode_ == CAPTURE_MODE_CALLBACK) {
isReadyToRead_ = true;
readThread_ = std::make_unique<std::thread>(&AudioStream::ReadCbThreadLoop, this);
}
isFirstRead_ = true;
isFirstWrite_ = true;
state_ = RUNNING;
AUDIO_INFO_LOG("StartAudioStream SUCCESS");
if (audioStreamTracker_) {
AUDIO_DEBUG_LOG("AudioStream:Calling Update tracker for Running");
audioStreamTracker_->UpdateTracker(sessionId_, state_, rendererInfo_, capturerInfo_);
}
return true;
}
AudioStream的StartAudioStream主要的工作是呼叫StartStream方法，StartStream方法是AudioServiceClient類中的方法。AudioServiceClient類是AudioStream的父類。接下來看一下AudioServiceClient的StartStream方法。
int32_t AudioServiceClient::StartStream(StateChangeCmdType cmdType)
{
int error;
lock_guard<mutex> lockdata(dataMutex);
pa_operation *operation = nullptr;
pa_threaded_mainloop_lock(mainLoop);
pa_stream_state_t state = pa_stream_get_state(paStream);
streamCmdStatus = 0;
stateChangeCmdType_ = cmdType;
operation = pa_stream_cork(paStream, 0, PAStreamStartSuccessCb, (void *)this);
while (pa_operation_get_state(operation) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait(mainLoop);
}
pa_operation_unref(operation);
pa_threaded_mainloop_unlock(mainLoop);
if (!streamCmdStatus) {
AUDIO_ERR_LOG("Stream Start Failed");
ResetPAAudioClient();
return AUDIO_CLIENT_START_STREAM_ERR;
} else {
AUDIO_INFO_LOG("Stream Started Successfully");
return AUDIO_CLIENT_SUCCESS;
}
}
StartStream方法中主要是呼叫了pulseaudio庫的pa_stream_cork方法進行流啟動，後續就呼叫到了pulseaudio庫中了。pulseaudio庫我們暫且不分析。
- 寫入資料
int32_t AudioRendererPrivate::Write(uint8_t *buffer, size_t bufferSize)
{
return audioStream_->Write(buffer, bufferSize);
}
透過呼叫AudioStream的Write方式實現功能，接下來看一下AudioStream的Write方法。
size_t AudioStream::Write(uint8_t *buffer, size_t buffer_size)
{
int32_t writeError;
StreamBuffer stream;
stream.buffer = buffer;
stream.bufferLen = buffer_size;
isWriteInProgress_ = true;
if (isFirstWrite_) {
if (RenderPrebuf(stream.bufferLen)) {
return ERR_WRITE_FAILED;
}
isFirstWrite_ = false;
}
size_t bytesWritten = WriteStream(stream, writeError);
isWriteInProgress_ = false;
if (writeError != 0) {
AUDIO_ERR_LOG("WriteStream fail,writeError:%{public}d", writeError);
return ERR_WRITE_FAILED;
}
return bytesWritten;
}
Write方法中分成兩個階段，首次寫資料，先呼叫RenderPrebuf方法，將preBuf_的資料寫入後再呼叫WriteStream進行音訊資料的寫入。
size_t AudioServiceClient::WriteStream(const StreamBuffer &stream, int32_t &pError)
{
size_t cachedLen = WriteToAudioCache(stream);
if (!acache.isFull) {
pError = error;
return cachedLen;
}
pa_threaded_mainloop_lock(mainLoop);
const uint8_t *buffer = acache.buffer.get();
size_t length = acache.totalCacheSize;
error = PaWriteStream(buffer, length);
acache.readIndex += acache.totalCacheSize;
acache.isFull = false;
if (!error && (length >= 0) && !acache.isFull) {
uint8_t *cacheBuffer = acache.buffer.get();
uint32_t offset = acache.readIndex;
uint32_t size = (acache.writeIndex - acache.readIndex);
if (size > 0) {
if (memcpy_s(cacheBuffer, acache.totalCacheSize, cacheBuffer + offset, size)) {
AUDIO_ERR_LOG("Update cache failed");
pa_threaded_mainloop_unlock(mainLoop);
pError = AUDIO_CLIENT_WRITE_STREAM_ERR;
return cachedLen;
}
AUDIO_INFO_LOG("rearranging the audio cache");
}
acache.readIndex = 0;
acache.writeIndex = 0;
if (cachedLen < stream.bufferLen) {
StreamBuffer str;
str.buffer = stream.buffer + cachedLen;
str.bufferLen = stream.bufferLen - cachedLen;
AUDIO_DEBUG_LOG("writing pending data to audio cache: %{public}d", str.bufferLen);
cachedLen += WriteToAudioCache(str);
}
}
pa_threaded_mainloop_unlock(mainLoop);
pError = error;
return cachedLen;
}
WriteStream方法不是直接呼叫pulseaudio庫的寫入方法，而是透過WriteToAudioCache方法將資料寫入快取中，如果快取沒有寫滿則直接返回，不會進入下面的流程，只有當快取寫滿後，才會呼叫下面的PaWriteStream方法。該方法涉及對pulseaudio庫寫入操作的呼叫，所以快取的目的是避免對pulseaudio庫頻繁地做IO操作，提高了效率。
六、總結
本文主要對OpenHarmony 3.2 Beta多媒體子系統的音訊渲染模組進行介紹，首先梳理了Audio Render的整體流程，然後對幾個核心的方法進行程式碼的分析。整體的流程主要透過pulseaudio庫啟動流，然後透過pulseaudio庫的pa_stream_write方法進行資料的寫入，最後播放出音訊資料。
音訊渲染主要分為以下幾個層次：
（1）AudioRenderer的建立，實際建立的是它的子類AudioRendererPrivate例項。
（2）透過AudioRendererPrivate設定渲染的回撥。
（3）啟動渲染，這一部分程式碼最終會呼叫到pulseaudio庫中，相當於啟動了pulseaudio的流。
（4）透過pulseaudio庫的pa_stream_write方法將資料寫入裝置，進行播放。
對OpenHarmony 3.2 Beta多媒體系列開發感興趣的讀者，也可以閱讀我之前寫過幾篇文章：
《OpenHarmony 3.2 Beta多媒體系列——影片錄製》
《OpenHarmony 3.2 Beta原始碼分析之MediaLibrary》
《OpenHarmony 3.2 Beta多媒體系列——音影片播放框架》
《OpenHarmony 3.2 Beta多媒體系列——音影片播放gstreamer》。
