OpenHarmony 3.2 Beta Audio：音頻渲染

一、簡介

Audio是多媒體子系統(tǒng)中的一個重要模塊，其涉及的內(nèi)容比較多，有音頻的渲染、音頻的采集、音頻的策略管理等。本文主要針對音頻渲染功能進(jìn)行詳細(xì)地分析，并通過源碼中提供的例子，對音頻渲染進(jìn)行流程的梳理。

二、目錄

foundation/multimedia/audio_framework

audio_framework
├── frameworks
│  ├── js             #js 接口
│  │  └── napi
│  │    └── audio_renderer   #audio_renderer NAPI接口
│  │      ├── 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 接口
│    └── 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(shí)現(xiàn)的接口
│  │  └── native
│  │    └── audiorenderer    #audio渲染本地實(shí)現(xiàn)的接口定義
│  │      └── include
│  │        └── audio_renderer.h
│  └── kits            #js調(diào)用的接口
│    └── js
│      └── audio_renderer   #audio渲染NAPI接口的定義
│        └── include
│          └── audio_renderer_napi.h
└── services            #服務(wù)端
  └── audio_service
    ├── BUILD.gn
    ├── client         #IPC調(diào)用中的proxy端
    │  ├── include
    │  │  ├── audio_manager_proxy.h
    │  │  ├── audio_service_client.h
    │  └── src
    │    ├── audio_manager_proxy.cpp
    │    ├── audio_service_client.cpp
    └── server         #IPC調(diào)用中的server端
      ├── include
      │  └── audio_server.h
      └── src
        ├── audio_manager_stub.cpp
        └── audio_server.cpp

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三、音頻渲染總體流程

四、Native接口使用

在OpenAtom OpenHarmony（以下簡稱“OpenHarmony”）系統(tǒng)中，音頻模塊提供了功能測試代碼，本文選取了其中的音頻渲染例子作為切入點(diǎn)來進(jìn)行介紹，例子采用的是對wav格式的音頻文件進(jìn)行渲染。wav格式的音頻文件是wav頭文件和音頻的原始數(shù)據(jù)，不需要進(jìn)行數(shù)據(jù)解碼，所以音頻渲染直接對原始數(shù)據(jù)進(jìn)行操作，文件路徑為：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文件頭信息
    size_t bytesRead = fread(&wavHeader, 1, headerSize, wavFile);


    //設(shè)置AudioRenderer參數(shù)
    AudioRendererOptions rendererOptions = {};
    rendererOptions.streamInfo.encoding = AudioEncodingType::ENCODING_PCM;
    rendererOptions.streamInfo.samplingRate = static_cast(wavHeader.SamplesPerSec);
    rendererOptions.streamInfo.format = GetSampleFormat(wavHeader.bitsPerSample);
    rendererOptions.streamInfo.channels = static_cast(wavHeader.NumOfChan);
    rendererOptions.rendererInfo.contentType = contentType;
    rendererOptions.rendererInfo.streamUsage = streamUsage;
    rendererOptions.rendererInfo.rendererFlags = 0;


    //創(chuàng)建AudioRender實(shí)例
    unique_ptr audioRenderer = AudioRenderer::Create(rendererOptions);


    shared_ptr cb1 = make_shared();
    //設(shè)置音頻渲染回調(diào)
    ret = audioRenderer->SetRendererCallback(cb1);


    //InitRender方法主要調(diào)用了audioRenderer實(shí)例的Start方法，啟動音頻渲染
if(!InitRender(audioRenderer)) {
AUDIO_ERR_LOG("AudioRendererTest: Init render failed");
fclose(wavFile);
      return false;
}


    //StartRender方法主要是讀取wavFile文件的數(shù)據(jù)，然后通過調(diào)用audioRenderer實(shí)例的Write方法進(jìn)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(guān)閉wavFile
fclose(wavFile);
    return true;
}

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首先讀取wav文件，通過讀取到wav文件的頭信息對AudioRendererOptions相關(guān)的參數(shù)進(jìn)行設(shè)置，包括編碼格式、采樣率、采樣格式、通道數(shù)等。根據(jù)AudioRendererOptions設(shè)置的參數(shù)來創(chuàng)建AudioRenderer實(shí)例（實(shí)際上是AudioRendererPrivate），后續(xù)的音頻渲染主要是通過AudioRenderer實(shí)例進(jìn)行。創(chuàng)建完成后，調(diào)用AudioRenderer的Start方法，啟動音頻渲染。啟動后，通過AudioRenderer實(shí)例的Write方法，將數(shù)據(jù)寫入，音頻數(shù)據(jù)會被播放。

五、調(diào)用流程

1、創(chuàng)建AudioRenderer

std::unique_ptrAudioRenderer::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(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;
}

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首先通過AudioStream的GetStreamType方法獲取音頻流的類型，根據(jù)音頻流類型創(chuàng)建AudioRendererPrivate對象，AudioRendererPrivate是AudioRenderer的子類。緊接著對audioRenderer進(jìn)行參數(shù)設(shè)置，其中包括采樣格式、采樣率、通道數(shù)、編碼格式。設(shè)置完成后返回創(chuàng)建的AudioRendererPrivate實(shí)例。

2、設(shè)置回調(diào)

int32_t AudioRendererPrivate::SetRendererCallback(const std::shared_ptr &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 cbInterrupt =
    std::static_pointer_cast(audioInterruptCallback_);
  cbInterrupt->SaveCallback(callback);


  // Save and Set reference for stream callback. Order is important here.
  if (audioStreamCallback_ == nullptr) {
    audioStreamCallback_ = std::make_shared();
    if (audioStreamCallback_ == nullptr) {
      return ERROR;
    }
  }
  std::shared_ptr cbStream =
std::static_pointer_cast(audioStreamCallback_);
  cbStream->SaveCallback(callback);
  (void)audioStream_->SetStreamCallback(audioStreamCallback_);


  return SUCCESS;
}

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參數(shù)傳入的回調(diào)主要涉及到兩個方面：一方面是AudioInterruptCallbackImpl中設(shè)置了我們傳入的渲染回調(diào)，另一方面是AudioStreamCallbackRenderer中也設(shè)置了渲染回調(diào)。

3、啟動渲染

bool AudioRendererPrivate::Start(StateChangeCmdType cmdType) const
{
  AUDIO_INFO_LOG("AudioRenderer::Start");
  RendererState state = GetStatus();


  AudioInterrupt audioInterrupt;
  switch (mode_) {
    case InterruptMode:
      audioInterrupt = sharedInterrupt_;
      break;
    case InterruptMode:
      audioInterrupt = audioInterrupt_;
      break;
    default:
      break;
  }
  AUDIO_INFO_LOG("AudioRenderer: %{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);
}

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AudioPolicyManager::GetInstance().ActivateAudioInterrupt這個操作主要是根據(jù)AudioInterrupt來進(jìn)行音頻中斷的激活，這里涉及了音頻策略相關(guān)的內(nèi)容，后續(xù)會專門出關(guān)于音頻策略的文章進(jìn)行分析。這個方法的核心是通過調(diào)用AudioStream的StartAudioStream方法來啟動音頻流。

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;
}

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AudioStream的StartAudioStream主要的工作是調(diào)用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_guardlockdata(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;
  }
}

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StartStream方法中主要是調(diào)用了pulseaudio庫的pa_stream_cork方法進(jìn)行流啟動，后續(xù)就調(diào)用到了pulseaudio庫中了。pulseaudio庫我們暫且不分析。

4、寫入數(shù)據(jù)

int32_t AudioRendererPrivate::Write(uint8_t *buffer, size_t bufferSize)
{
  return audioStream_->Write(buffer, bufferSize);
}

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通過調(diào)用AudioStream的Write方式實(shí)現(xiàn)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;
}

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Write方法中分成兩個階段，首次寫數(shù)據(jù)，先調(diào)用RenderPrebuf方法，將preBuf_的數(shù)據(jù)寫入后再調(diào)用WriteStream進(jìn)行音頻數(shù)據(jù)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;
}

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WriteStream方法不是直接調(diào)用pulseaudio庫的寫入方法，而是通過WriteToAudioCache方法將數(shù)據(jù)寫入緩存中，如果緩存沒有寫滿則直接返回，不會進(jìn)入下面的流程，只有當(dāng)緩存寫滿后，才會調(diào)用下面的PaWriteStream方法。該方法涉及對pulseaudio庫寫入操作的調(diào)用，所以緩存的目的是避免對pulseaudio庫頻繁地做IO操作，提高了效率。

六、總結(jié)

本文主要對OpenHarmony 3.2 Beta多媒體子系統(tǒng)的音頻渲染模塊進(jìn)行介紹，首先梳理了Audio Render的整體流程，然后對幾個核心的方法進(jìn)行代碼的分析。整體的流程主要通過pulseaudio庫啟動流，然后通過pulseaudio庫的pa_stream_write方法進(jìn)行數(shù)據(jù)的寫入，最后播放出音頻數(shù)據(jù)。

音頻渲染主要分為以下幾個層次：

1. AudioRenderer的創(chuàng)建，實(shí)際創(chuàng)建的是它的子類AudioRendererPrivate實(shí)例。

2. 通過AudioRendererPrivate設(shè)置渲染的回調(diào)。

3. 啟動渲染，這一部分代碼最終會調(diào)用到pulseaudio庫中，相當(dāng)于啟動了pulseaudio的流。

4. 通過pulseaudio庫的pa_stream_write方法將數(shù)據(jù)寫入設(shè)備，進(jìn)行播放。

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