C++ soundtouch::SoundTouch类代码示例

void ApplySettings(soundtouch::SoundTouch &sndtouch)
{
    sndtouch.setSetting(SETTING_SEQUENCE_MS, SequenceLenMS);
    sndtouch.setSetting(SETTING_SEEKWINDOW_MS, SeekWindowMS);
    sndtouch.setSetting(SETTING_OVERLAP_MS, OverlapMS);
}

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;
}

void OpenALStream::SoundLoop()
{
  Common::SetCurrentThreadName("Audio thread - openal");

  bool surround_capable = SConfig::GetInstance().bDPL2Decoder;
  bool float32_capable = false;
  bool fixed32_capable = false;

#if defined(__APPLE__)
  surround_capable = false;
#endif

  u32 ulFrequency = m_mixer->GetSampleRate();
  numBuffers = SConfig::GetInstance().iLatency + 2;  // OpenAL requires a minimum of two buffers

  memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
  uiSource = 0;

  if (alIsExtensionPresent("AL_EXT_float32"))
    float32_capable = true;

  // As there is no extension to check for 32-bit fixed point support
  // and we know that only a X-Fi with hardware OpenAL supports it,
  // we just check if one is being used.
  if (strstr(alGetString(AL_RENDERER), "X-Fi"))
    fixed32_capable = true;

  // Clear error state before querying or else we get false positives.
  ALenum err = alGetError();

  // Generate some AL Buffers for streaming
  alGenBuffers(numBuffers, (ALuint*)uiBuffers);
  err = CheckALError("generating buffers");

  // Generate a Source to playback the Buffers
  alGenSources(1, &uiSource);
  err = CheckALError("generating sources");

  // Set the default sound volume as saved in the config file.
  alSourcef(uiSource, AL_GAIN, fVolume);

  // TODO: Error handling
  // ALenum err = alGetError();

  unsigned int nextBuffer = 0;
  unsigned int numBuffersQueued = 0;
  ALint iState = 0;

  soundTouch.setChannels(2);
  soundTouch.setSampleRate(ulFrequency);
  soundTouch.setTempo(1.0);
  soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
  soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
  soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
  soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
  soundTouch.setSetting(SETTING_OVERLAP_MS, 12);

  while (m_run_thread.IsSet())
  {
    // Block until we have a free buffer
    int numBuffersProcessed;
    alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
    if (numBuffers == numBuffersQueued && !numBuffersProcessed)
    {
      soundSyncEvent.Wait();
      continue;
    }

    // Remove the Buffer from the Queue.
    if (numBuffersProcessed)
    {
      ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
      alSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
      err = CheckALError("unqueuing buffers");

      numBuffersQueued -= numBuffersProcessed;
    }

    // num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
    const u32 stereo_16_bit_size = 4;
    const u32 dma_length = 32;
    const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
    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);
//.........这里部分代码省略.........

void OpenALStream::SoundLoop()
{
	Common::SetCurrentThreadName("Audio thread - openal");

	bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
#if defined(__APPLE__)
	bool float32_capable = false;
	const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
	// OSX does not have the alext AL_FORMAT_51CHN32 yet.
	surround_capable = false;
	const ALenum AL_FORMAT_51CHN32 = 0;
#else
	bool float32_capable = true;
#endif

	u32 ulFrequency = m_mixer->GetSampleRate();
	numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers

	memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
	uiSource = 0;

	// Generate some AL Buffers for streaming
	alGenBuffers(numBuffers, (ALuint *)uiBuffers);
	// Generate a Source to playback the Buffers
	alGenSources(1, &uiSource);

	// Short Silence
	memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
	for (int i = 0; i < numBuffers; i++)
	{
		if (surround_capable)
			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
		else
			alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT, ulFrequency);
	}
	alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
	alSourcePlay(uiSource);

	// Set the default sound volume as saved in the config file.
	alSourcef(uiSource, AL_GAIN, fVolume);

	// TODO: Error handling
	//ALenum err = alGetError();

	ALint iBuffersFilled = 0;
	ALint iBuffersProcessed = 0;
	ALint iState = 0;
	ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};

	soundTouch.setChannels(2);
	soundTouch.setSampleRate(ulFrequency);
	soundTouch.setTempo(1.0);
	soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
	soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
	soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
	soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
	soundTouch.setSetting(SETTING_OVERLAP_MS, 12);

	while (!threadData)
	{
		// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
		const u32 stereo_16_bit_size = 4;
		const u32 dma_length = 32;
		const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
		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 << 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)
			{
//.........这里部分代码省略.........