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Changed SoundTouch to use float samples, allowing SSE to be used.

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Made the DPL2 decoder disabled by default.
Re-added the audio hack used by the Accurate VBeam emulation option.
This commit is contained in:
skidau
2013-01-15 22:29:26 +11:00
parent 1c462a1eca
commit 6d4a566bc4
6 changed files with 32 additions and 17 deletions
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35
Source/Core/AudioCommon/Src/OpenALStream.cpp
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@ -176,6 +176,7 @@ void OpenALStream::SoundLoop()
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);
@ -197,7 +198,16 @@ void OpenALStream::SoundLoop()
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
soundTouch.putSamples(realtimeBuffer, numSamples);
// Convert the samples from short to float
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
for (u32 i = 0; i < numSamples; ++i)
{
dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16);
dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16);
}
soundTouch.putSamples(dest, numSamples);
if (iBuffersProcessed == iBuffersFilled)
{
@ -241,16 +251,8 @@ void OpenALStream::SoundLoop()
#else
if (surround_capable)
{
// Convert the samples from short to float for the dpl2 decoder
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples; ++i)
{
dest[i * 2 + 0] = (float)sampleBuffer[i * 2 + 0] / (1<<16);
dest[i * 2 + 1] = (float)sampleBuffer[i * 2 + 1] / (1<<16);
}
float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
dpl2decode(dest, nSamples, dpl2);
dpl2decode(sampleBuffer, nSamples, dpl2);
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
ALenum err = alGetError();
@ -268,7 +270,18 @@ void OpenALStream::SoundLoop()
#endif
if (!surround_capable)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, sampleBuffer, nSamples * 2 * 2, ulFrequency);
#if defined(__APPLE__)
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples; ++i)
{
stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16));
stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
}
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency);
#else
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, ulFrequency);
#endif
}
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);