分析承接<7-13 打开播放设备>

H:\webrtc-20210315\webrtc-20210315\webrtc\webrtc-checkout\src\modules\audio_device\win\audio_device_core_win.cc

AudioDeviceWindowsCore::InitPlayout

int32_t AudioDeviceWindowsCore::InitPlayout() {
  MutexLock lock(&mutex_);
  if (_playing) {
    return -1;
  }
  if (_playIsInitialized) {
    return 0;
  }
  if (_ptrDeviceOut == NULL) {
    return -1;
  }
  // Initialize the speaker (devices might have been added or removed)
  if (InitSpeakerLocked() == -1) {
    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
  }
  // Ensure that the updated rendering endpoint device is valid
  if (_ptrDeviceOut == NULL) {
    return -1;
  }
  if (_builtInAecEnabled && _recIsInitialized) {
    // Ensure the correct render device is configured in case
    // InitRecording() was called before InitPlayout().
    if (SetDMOProperties() == -1) {
      return -1;
    }
  }
  HRESULT hr = S_OK;
  WAVEFORMATEX* pWfxOut = NULL;
  WAVEFORMATEX Wfx = WAVEFORMATEX();
  WAVEFORMATEX* pWfxClosestMatch = NULL;
  // Create COM object with IAudioClient interface.
  SAFE_RELEASE(_ptrClientOut);
  hr = _ptrDeviceOut->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL,
                               (void**)&_ptrClientOut);
  EXIT_ON_ERROR(hr);
    // 获取扬声器支持的参数
  // Retrieve the stream format that the audio engine uses for its internal
  // processing (mixing) of shared-mode streams.
  hr = _ptrClientOut->GetMixFormat(&pWfxOut);
  if (SUCCEEDED(hr)) {
***
  }
    // 构造我们想输出的参数
  // Set wave format
  Wfx.wFormatTag = WAVE_FORMAT_PCM;
  Wfx.wBitsPerSample = 16;
  Wfx.cbSize = 0;
  const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
  hr = S_FALSE;
    // 获取一个支持共享模式下的freqs值
  // Iterate over frequencies and channels, in order of priority
  for (unsigned int freq = 0; freq < sizeof(freqs) / sizeof(freqs[0]); freq++) {
      // _playChannelsPrioList两个成员，一个单声道，一个双声道
    for (unsigned int chan = 0; chan < sizeof(_playChannelsPrioList) /
                                           sizeof(_playChannelsPrioList[0]);
         chan++) {
      Wfx.nChannels = _playChannelsPrioList[chan];
      Wfx.nSamplesPerSec = freqs[freq];
      Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
      Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
      // If the method succeeds and the audio endpoint device supports the
      // specified stream format, it returns S_OK. If the method succeeds and
      // provides a closest match to the specified format, it returns S_FALSE.
        // 判断该系统是否在该采样率下支持共享模式
      hr = _ptrClientOut->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &Wfx,
                                            &pWfxClosestMatch);
      if (hr == S_OK) {
        break;
      } else {
      ****
      }
    }
    if (hr == S_OK)
      break;
  }
  // TODO(andrew): what happens in the event of failure in the above loop?
  //   Is _ptrClientOut->Initialize expected to fail?
  //   Same in InitRecording().
  if (hr == S_OK) {
    _playAudioFrameSize = Wfx.nBlockAlign;
    // Block size is the number of samples each channel in 10ms.
    _playBlockSize = Wfx.nSamplesPerSec / 100;
    _playSampleRate = Wfx.nSamplesPerSec;
    _devicePlaySampleRate =
        Wfx.nSamplesPerSec;  // The device itself continues to run at 44.1 kHz.
    _devicePlayBlockSize = Wfx.nSamplesPerSec / 100;
    _playChannels = Wfx.nChannels;
***
  }
  // Create a rendering stream.
  //
  // ****************************************************************************
  // For a shared-mode stream that uses event-driven buffering, the caller must
  // set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
  // determines how large a buffer to allocate based on the scheduling period
  // of the audio engine. Although the client's buffer processing thread is
  // event driven, the basic buffer management process, as described previously,
  // is unaltered.
  // Each time the thread awakens, it should call
  // IAudioClient::GetCurrentPadding to determine how much data to write to a
  // rendering buffer or read from a capture buffer. In contrast to the two
  // buffers that the Initialize method allocates for an exclusive-mode stream
  // that uses event-driven buffering, a shared-mode stream requires a single
  // buffer.
  // ****************************************************************************
  // 指明扬声器的buffer的最小值
  REFERENCE_TIME hnsBufferDuration =
      0;  // ask for minimum buffer size (default)
  if (_devicePlaySampleRate == 44100) {
    // Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
    // There seems to be a larger risk of underruns for 44.1 compared
    // with the default rate (48kHz). When using default, we set the requested
    // buffer duration to 0, which sets the buffer to the minimum size
    // required by the engine thread. The actual buffer size can then be
    // read by GetBufferSize() and it is 20ms on most machines.
    hnsBufferDuration = 30 * 10000;
  }
  hr = _ptrClientOut->Initialize(
      AUDCLNT_SHAREMODE_SHARED,  // share Audio Engine with other applications
      AUDCLNT_STREAMFLAGS_EVENTCALLBACK,  // processing of the audio buffer by
                                          // the client will be event driven
      hnsBufferDuration,  // requested buffer capacity as a time value (in
                          // 100-nanosecond units)
      0,                  // periodicity
      &Wfx,               // selected wave format
      NULL);              // session GUID
  if (FAILED(hr)) {
    RTC_LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
  }
  EXIT_ON_ERROR(hr);
    // 扬声器播放的声音放到_ptrAudioBuffer
  if (_ptrAudioBuffer) {
    // Update the audio buffer with the selected parameters
    _ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
  } else {
***
  }
  // Get the actual size of the shared (endpoint buffer).
  // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
  UINT bufferFrameCount(0);
  hr = _ptrClientOut->GetBufferSize(&bufferFrameCount);
  if (SUCCEEDED(hr)) {
    RTC_LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
                        << bufferFrameCount << " (<=> "
                        << bufferFrameCount * _playAudioFrameSize << " bytes)";
  }
  // Set the event handle that the system signals when an audio buffer is ready
  // to be processed by the client.
  hr = _ptrClientOut->SetEventHandle(_hRenderSamplesReadyEvent);
  EXIT_ON_ERROR(hr);
  // Get an IAudioRenderClient interface.
  SAFE_RELEASE(_ptrRenderClient);
  hr = _ptrClientOut->GetService(__uuidof(IAudioRenderClient),
                                 (void**)&_ptrRenderClient);
  EXIT_ON_ERROR(hr);
  // Mark playout side as initialized
  _playIsInitialized = true;
*****
}