mirror of
https://github.com/dolphin-emu/dolphin.git
synced 2025-01-26 12:35:27 +00:00
Clarifying the OpenAL loop
because it isn't as clear as it can be
This commit is contained in:
John Peterson
parent
3ddd24872b
commit
adb83cfabe
@ -97,7 +97,7 @@ void OpenALStream::SetVolume(int volume)
|
||||
fVolume = (float)volume / 100.0f;
|
||||
|
||||
if (uiSource)
|
||||
alSourcef(uiSource, AL_GAIN, fVolume);
|
||||
alSourcef(uiSource, AL_GAIN, fVolume);
|
||||
}
|
||||
|
||||
void OpenALStream::Update()
|
||||
@ -124,6 +124,15 @@ void OpenALStream::SoundLoop()
|
||||
{
|
||||
Common::SetCurrentThreadName("Audio thread - openal");
|
||||
|
||||
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
|
||||
#if defined(__APPLE__)
|
||||
bool float32_capable = false;
|
||||
// OSX does not have the alext AL_FORMAT_51CHN32 yet.
|
||||
surround_capable = false;
|
||||
#else
|
||||
bool float32_capable = true;
|
||||
#endif
|
||||
|
||||
u32 ulFrequency = m_mixer->GetSampleRate();
|
||||
numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
|
||||
|
||||
@ -136,22 +145,20 @@ void OpenALStream::SoundLoop()
|
||||
alGenSources(1, &uiSource);
|
||||
|
||||
// Short Silence
|
||||
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
|
||||
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
|
||||
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 !defined(__APPLE__)
|
||||
if (Core::g_CoreStartupParameter.bDPL2Decoder)
|
||||
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
|
||||
if (surround_capable)
|
||||
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
|
||||
else
|
||||
#endif
|
||||
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
|
||||
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);
|
||||
alSourcef(uiSource, AL_GAIN, fVolume);
|
||||
|
||||
// TODO: Error handling
|
||||
//ALenum err = alGetError();
|
||||
@ -170,14 +177,7 @@ void OpenALStream::SoundLoop()
|
||||
soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
|
||||
soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
|
||||
|
||||
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
|
||||
#if defined(__APPLE__)
|
||||
bool float32_capable = false;
|
||||
#else
|
||||
bool float32_capable = true;
|
||||
#endif
|
||||
|
||||
while (!threadData)
|
||||
while (!threadData)
|
||||
{
|
||||
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
|
||||
const u32 stereo_16_bit_size = 4;
|
||||
@ -193,12 +193,9 @@ void OpenALStream::SoundLoop()
|
||||
numSamples = m_mixer->Mix(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);
|
||||
}
|
||||
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);
|
||||
|
||||
@ -230,101 +227,94 @@ void OpenALStream::SoundLoop()
|
||||
}
|
||||
}
|
||||
|
||||
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * OAL_MAX_BUFFERS);
|
||||
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
|
||||
|
||||
if (nSamples > minSamples)
|
||||
if (nSamples < minSamples)
|
||||
continue;
|
||||
|
||||
// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer)
|
||||
if (iBuffersFilled == 0)
|
||||
{
|
||||
// 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 defined(__APPLE__)
|
||||
// OSX does not have the alext AL_FORMAT_51CHN32 yet.
|
||||
surround_capable = false;
|
||||
#else
|
||||
if (surround_capable)
|
||||
{
|
||||
float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
|
||||
dpl2decode(sampleBuffer, nSamples, dpl2);
|
||||
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, 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);
|
||||
}
|
||||
}
|
||||
#endif
|
||||
if (!surround_capable)
|
||||
{
|
||||
#if !defined(__APPLE__)
|
||||
if (float32_capable)
|
||||
{
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, 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);
|
||||
}
|
||||
}
|
||||
#endif
|
||||
if (!float32_capable)
|
||||
{
|
||||
// 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);
|
||||
}
|
||||
}
|
||||
|
||||
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
|
||||
alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
|
||||
ALenum err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
|
||||
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
|
||||
}
|
||||
iBuffersFilled++;
|
||||
}
|
||||
|
||||
if (iBuffersFilled == numBuffers)
|
||||
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)
|
||||
{
|
||||
alSourcePlay(uiSource);
|
||||
err = alGetError();
|
||||
if (err != 0)
|
||||
// 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(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * FRAME_STEREO_FLOAT, ulFrequency);
|
||||
ALenum err = alGetError();
|
||||
if (err == AL_INVALID_ENUM)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
|
||||
float32_capable = false;
|
||||
}
|
||||
else if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
|
||||
}
|
||||
}
|
||||
|
||||
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
|
||||
if (iState != AL_PLAYING)
|
||||
else
|
||||
{
|
||||
// Buffer underrun occurred, resume playback
|
||||
alSourcePlay(uiSource);
|
||||
err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
|
||||
}
|
||||
// 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 << 16));
|
||||
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * FRAME_STEREO_SHORT, ulFrequency);
|
||||
}
|
||||
}
|
||||
|
||||
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
|
||||
ALenum err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
|
||||
}
|
||||
iBuffersFilled++;
|
||||
|
||||
if (iBuffersFilled == numBuffers)
|
||||
{
|
||||
alSourcePlay(uiSource);
|
||||
err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
|
||||
}
|
||||
}
|
||||
|
||||
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);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
@ -30,11 +30,16 @@
|
||||
#include <soundtouch/STTypes.h>
|
||||
|
||||
// 16 bit Stereo
|
||||
#define SFX_MAX_SOURCE 1
|
||||
#define OAL_MAX_BUFFERS 32
|
||||
#define OAL_MAX_SAMPLES 256
|
||||
#define SURROUND_CHANNELS 6 // number of channels in surround mode
|
||||
#define SIZE_FLOAT 4 // size of a float in bytes
|
||||
#define SFX_MAX_SOURCE 1
|
||||
#define OAL_MAX_BUFFERS 32
|
||||
#define OAL_MAX_SAMPLES 256
|
||||
#define STEREO_CHANNELS 2
|
||||
#define SURROUND_CHANNELS 6 // number of channels in surround mode
|
||||
#define SIZE_SHORT 2
|
||||
#define SIZE_FLOAT 4 // size of a float in bytes
|
||||
#define FRAME_STEREO_SHORT STEREO_CHANNELS * SIZE_SHORT
|
||||
#define FRAME_STEREO_FLOAT STEREO_CHANNELS * SIZE_FLOAT
|
||||
#define FRAME_SURROUND_FLOAT SURROUND_CHANNELS * SIZE_FLOAT
|
||||
#endif
|
||||
|
||||
class OpenALStream: public SoundStream
|
||||
@ -61,8 +66,8 @@ private:
|
||||
std::thread thread;
|
||||
Common::Event soundSyncEvent;
|
||||
|
||||
short realtimeBuffer[OAL_MAX_SAMPLES * 2];
|
||||
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
|
||||
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
|
||||
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
|
||||
ALuint uiBuffers[OAL_MAX_BUFFERS];
|
||||
ALuint uiSource;
|
||||
ALfloat fVolume;
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user