C++ SoundTouch::clear方法代码示例

void OpenALStream::Stop()
{
  m_run_thread.Clear();
  // kick the thread if it's waiting
  soundSyncEvent.Set();

  soundTouch.clear();

  thread.join();

  alSourceStop(uiSource);
  alSourcei(uiSource, AL_BUFFER, 0);

  // Clean up buffers and sources
  alDeleteSources(1, &uiSource);
  uiSource = 0;
  alDeleteBuffers(numBuffers, uiBuffers);

  ALCcontext* pContext = alcGetCurrentContext();
  ALCdevice* pDevice = alcGetContextsDevice(pContext);

  alcMakeContextCurrent(nullptr);
  alcDestroyContext(pContext);
  alcCloseDevice(pDevice);
}

void OpenALStream::Clear(bool mute)
{
  m_muted = mute;

  if (m_muted)
  {
    soundTouch.clear();
    alSourceStop(uiSource);
  }
  else
  {
    alSourcePlay(uiSource);
  }
}

//
// AyuanX: Spec says OpenAL1.1 is thread safe already
//
bool OpenALStream::Start()
{
  m_run_thread.Set();
  bool bReturn = false;

  ALDeviceList pDeviceList;
  if (pDeviceList.GetNumDevices())
  {
    char* defDevName = pDeviceList.GetDeviceName(pDeviceList.GetDefaultDevice());

    INFO_LOG(AUDIO, "Found OpenAL device %s", defDevName);

    ALCdevice* pDevice = alcOpenDevice(defDevName);
    if (pDevice)
    {
      ALCcontext* pContext = alcCreateContext(pDevice, nullptr);
      if (pContext)
      {
        // Used to determine an appropriate period size (2x period = total buffer size)
        // ALCint refresh;
        // alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
        // period_size_in_millisec = 1000 / refresh;

        alcMakeContextCurrent(pContext);
        thread = std::thread(&OpenALStream::SoundLoop, this);
        bReturn = true;
      }
      else
      {
        alcCloseDevice(pDevice);
        PanicAlertT("OpenAL: can't create context for device %s", defDevName);
      }
    }
    else
    {
      PanicAlertT("OpenAL: can't open device %s", defDevName);
    }
  }
  else
  {
    PanicAlertT("OpenAL: can't find sound devices");
  }

  // Initialize DPL2 parameters
  DPL2Reset();

  soundTouch.clear();
  return bReturn;
}

//.........这里部分代码省略.........
    u64 audio_dma_period = SystemTimers::GetTicksPerSecond() /
                           (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
    u64 num_samples_to_render =
        (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();

    unsigned int numSamples = (unsigned int)num_samples_to_render;
    unsigned int minSamples =
        surround_capable ? 240 : 0;  // DPL2 accepts 240 samples minimum (FWRDURATION)

    numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
    numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);

    // Convert the samples from short to float
    float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
    for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
      dest[i] = (float)realtimeBuffer[i] / (1 << 15);

    soundTouch.putSamples(dest, numSamples);

    double rate = (double)m_mixer->GetCurrentSpeed();
    if (rate <= 0)
    {
      Core::RequestRefreshInfo();
      rate = (double)m_mixer->GetCurrentSpeed();
    }

    // Place a lower limit of 10% speed.  When a game boots up, there will be
    // many silence samples.  These do not need to be timestretched.
    if (rate > 0.10)
    {
      soundTouch.setTempo(rate);
      if (rate > 10)
      {
        soundTouch.clear();
      }
    }

    unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);

    if (nSamples <= minSamples)
      continue;

    if (surround_capable)
    {
      float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
      DPL2Decode(sampleBuffer, nSamples, dpl2);

      // zero-out the subwoofer channel - DPL2Decode generates a pretty
      // good 5.0 but not a good 5.1 output.  Sadly there is not a 5.0
      // AL_FORMAT_50CHN32 to make this super-explicit.
      // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
      for (u32 i = 0; i < nSamples; ++i)
      {
        dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
      }

      if (float32_capable)
      {
        alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
                     nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
      }
      else if (fixed32_capable)
      {
        int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];

        for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)

//.........这里部分代码省略.........
		numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);

		// Convert the samples from short to float
		float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
		for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
			dest[i] = (float)realtimeBuffer[i] / (1 << 16);

		soundTouch.putSamples(dest, numSamples);

		if (iBuffersProcessed == iBuffersFilled)
		{
			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
			iBuffersFilled = 0;
		}

		if (iBuffersProcessed)
		{
			float rate = m_mixer->GetCurrentSpeed();
			if (rate <= 0)
			{
				Core::RequestRefreshInfo();
				rate = m_mixer->GetCurrentSpeed();
			}

			// Place a lower limit of 10% speed.  When a game boots up, there will be
			// many silence samples.  These do not need to be timestretched.
			if (rate > 0.10)
			{
				// Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio
				soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate)));
				soundTouch.setTempo(rate);
				if (rate > 10)
				{
					soundTouch.clear();
				}
			}

			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);

			if (nSamples <= minSamples)
				continue;

			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
			if (iBuffersFilled == 0)
			{
				alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
				ALenum err = alGetError();
				if (err != 0)
				{
					ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
				}
			}

			if (surround_capable)
			{
				float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
				dpl2decode(sampleBuffer, nSamples, dpl2);
				alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
				ALenum err = alGetError();
				if (err == AL_INVALID_ENUM)
				{
					// 5.1 is not supported by the host, fallback to stereo
					WARN_LOG(AUDIO, "Unable to set 5.1 surround mode.  Updating OpenAL Soft might fix this issue.");
					surround_capable = false;
				}
				else if (err != 0)