C++ SoundTouch::receiveSamples方法代码示例

//.........这里部分代码省略.........

    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)
        {
          // For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1.
          // Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
          // fix the decoder or implement a limiter.

//.........这里部分代码省略.........
		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)
				{
					ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
				}
			}