summaryrefslogtreecommitdiff
path: root/Source/Core/AudioCommon/OpenALStream.cpp
diff options
context:
space:
mode:
Diffstat (limited to 'Source/Core/AudioCommon/OpenALStream.cpp')
-rw-r--r--Source/Core/AudioCommon/OpenALStream.cpp176
1 files changed, 88 insertions, 88 deletions
diff --git a/Source/Core/AudioCommon/OpenALStream.cpp b/Source/Core/AudioCommon/OpenALStream.cpp
index e5ede582f0..7b63d0a657 100644
--- a/Source/Core/AudioCommon/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/OpenALStream.cpp
@@ -225,120 +225,120 @@ void OpenALStream::SoundLoop()
soundTouch.putSamples(dest, numSamples);
- double rate = (double)m_mixer->GetCurrentSpeed();
- if (rate <= 0)
- {
- Core::RequestRefreshInfo();
- rate = (double)m_mixer->GetCurrentSpeed();
- }
+ 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)
+ // 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.setTempo(rate);
- if (rate > 10)
- {
- soundTouch.clear();
- }
+ soundTouch.clear();
}
+ }
- unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
+ unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
- if (nSamples <= minSamples)
- continue;
+ if (nSamples <= minSamples)
+ continue;
- if (surround_capable)
+ 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)
{
- 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;
- }
+ 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));
+ 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);
- }
+ alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
+ nSamples * FRAME_SURROUND_SHORT, 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);
+ }
+ }
+
+ 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();
- if (err != 0)
- {
- ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
- }
- numBuffersQueued++;
- nextBuffer = (nextBuffer + 1) % numBuffers;
+ 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)
+ alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
+ if (iState != AL_PLAYING)
+ {
+ // Buffer underrun occurred, resume playback
+ alSourcePlay(uiSource);
+ err = alGetError();
+ if (err != 0)
{
- // 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);
}
+ }
}
}
generated by cgit v1.2.3 (git 2.25.1) at 2026-08-05 20:01:50 +0000