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fix indendentation

This commit is contained in:
Michael Maltese 2016-10-01 20:56:25 -07:00
parent 567dffc1ee
commit 8fa79f3897
1 changed files with 90 additions and 90 deletions
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180
Source/Core/AudioCommon/OpenALStream.cpp
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@ -225,120 +225,120 @@ void OpenALStream::SoundLoop()
soundTouch.putSamples(dest, numSamples);
double rate = (double)m_mixer->GetCurrentSpeed();
if (rate <= 0)
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)
{
Core::RequestRefreshInfo();
rate = (double)m_mixer->GetCurrentSpeed();
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;
}
// 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)
if (float32_capable)
{
soundTouch.setTempo(rate);
if (rate > 10)
{
soundTouch.clear();
}
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
}
else
{
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
nSamples * FRAME_SURROUND_SHORT, ulFrequency);
}
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
if (nSamples <= minSamples)
continue;
if (surround_capable)
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
{
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
{
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
nSamples * FRAME_SURROUND_SHORT, ulFrequency);
}
// 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);
}
}
else
{
if (float32_capable)
{
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
nSamples * FRAME_STEREO_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;
float32_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
else
{
if (float32_capable)
{
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
nSamples * FRAME_STEREO_FLOAT, ulFrequency);
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
{
float32_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
else
{
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
nSamples * FRAME_STEREO_SHORT, ulFrequency);
}
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
nSamples * FRAME_STEREO_SHORT, ulFrequency);
}
}
alSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
ALenum err = alGetError();
alSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
}
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
}
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
}
}
}