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Merge pull request #5631 from LAGonauta/openal-real-latency 

Fixed latency setting and cleaned-up OpenAL backend
This commit is contained in:
shuffle2 2017-06-27 11:31:16 -07:00 committed by GitHub
parent 43dba0cb47 db7bb3bda8
commit ce4d514542
5 changed files with 115 additions and 129 deletions
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208
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -100,27 +100,27 @@ bool OpenALStream::Start()
return false;
}
const char* defaultDeviceName = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
INFO_LOG(AUDIO, "Found OpenAL device %s", defaultDeviceName);
const char* default_device_dame = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
INFO_LOG(AUDIO, "Found OpenAL device %s", default_device_dame);
ALCdevice* pDevice = palcOpenDevice(defaultDeviceName);
if (!pDevice)
ALCdevice* device = palcOpenDevice(default_device_dame);
if (!device)
{
PanicAlertT("OpenAL: can't open device %s", defaultDeviceName);
PanicAlertT("OpenAL: can't open device %s", default_device_dame);
return false;
}
ALCcontext* pContext = palcCreateContext(pDevice, nullptr);
if (!pContext)
ALCcontext* context = palcCreateContext(device, nullptr);
if (!context)
{
palcCloseDevice(pDevice);
PanicAlertT("OpenAL: can't create context for device %s", defaultDeviceName);
palcCloseDevice(device);
PanicAlertT("OpenAL: can't create context for device %s", default_device_dame);
return false;
}
palcMakeContextCurrent(pContext);
palcMakeContextCurrent(context);
m_run_thread.Set();
thread = std::thread(&OpenALStream::SoundLoop, this);
m_thread = std::thread(&OpenALStream::SoundLoop, this);
return true;
}
@ -128,37 +128,37 @@ void OpenALStream::Stop()
{
m_run_thread.Clear();
// kick the thread if it's waiting
soundSyncEvent.Set();
m_sound_sync_event.Set();
thread.join();
m_thread.join();
palSourceStop(uiSource);
palSourcei(uiSource, AL_BUFFER, 0);
palSourceStop(m_source);
palSourcei(m_source, AL_BUFFER, 0);
// Clean up buffers and sources
palDeleteSources(1, &uiSource);
uiSource = 0;
palDeleteBuffers(numBuffers, uiBuffers);
palDeleteSources(1, &m_source);
m_source = 0;
palDeleteBuffers(OAL_BUFFERS, m_buffers.data());
ALCcontext* pContext = palcGetCurrentContext();
ALCdevice* pDevice = palcGetContextsDevice(pContext);
ALCcontext* context = palcGetCurrentContext();
ALCdevice* device = palcGetContextsDevice(context);
palcMakeContextCurrent(nullptr);
palcDestroyContext(pContext);
palcCloseDevice(pDevice);
palcDestroyContext(context);
palcCloseDevice(device);
}
void OpenALStream::SetVolume(int volume)
{
fVolume = (float)volume / 100.0f;
m_volume = (float)volume / 100.0f;
if (uiSource)
palSourcef(uiSource, AL_GAIN, fVolume);
if (m_source)
palSourcef(m_source, AL_GAIN, m_volume);
}
void OpenALStream::Update()
{
soundSyncEvent.Set();
m_sound_sync_event.Set();
}
void OpenALStream::Clear(bool mute)
@ -167,11 +167,11 @@ void OpenALStream::Clear(bool mute)
if (m_muted)
{
palSourceStop(uiSource);
palSourceStop(m_source);
}
else
{
palSourcePlay(uiSource);
palSourcePlay(m_source);
}
}
@ -229,86 +229,109 @@ void OpenALStream::SoundLoop()
// we just check if one is being used.
bool fixed32_capable = IsCreativeXFi();
u32 ulFrequency = m_mixer->GetSampleRate();
numBuffers = SConfig::GetInstance().iLatency + 2; // OpenAL requires a minimum of two buffers
u32 frequency = m_mixer->GetSampleRate();
memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
uiSource = 0;
u32 frames_per_buffer;
// Can't have zero samples per buffer
if (SConfig::GetInstance().iLatency > 0)
{
frames_per_buffer = frequency / 1000 * SConfig::GetInstance().iLatency / OAL_BUFFERS;
}
else
{
frames_per_buffer = frequency / 1000 * 1 / OAL_BUFFERS;
}
if (frames_per_buffer > OAL_MAX_FRAMES)
{
frames_per_buffer = OAL_MAX_FRAMES;
}
// DPL2 needs a minimum number of samples to work (FWRDURATION)
if (use_surround && frames_per_buffer < 240)
{
frames_per_buffer = 240;
}
INFO_LOG(AUDIO, "Using %d buffers, each with %d audio frames for a total of %d.", OAL_BUFFERS,
frames_per_buffer, frames_per_buffer * OAL_BUFFERS);
// Should we make these larger just in case the mixer ever sends more samples
// than what we request?
m_realtime_buffer.resize(frames_per_buffer * STEREO_CHANNELS);
m_source = 0;
// Clear error state before querying or else we get false positives.
ALenum err = palGetError();
// Generate some AL Buffers for streaming
palGenBuffers(numBuffers, (ALuint*)uiBuffers);
palGenBuffers(OAL_BUFFERS, (ALuint*)m_buffers.data());
err = CheckALError("generating buffers");
// Generate a Source to playback the Buffers
palGenSources(1, &uiSource);
palGenSources(1, &m_source);
err = CheckALError("generating sources");
// Set the default sound volume as saved in the config file.
palSourcef(uiSource, AL_GAIN, fVolume);
palSourcef(m_source, AL_GAIN, m_volume);
// TODO: Error handling
// ALenum err = alGetError();
unsigned int nextBuffer = 0;
unsigned int numBuffersQueued = 0;
ALint iState = 0;
unsigned int next_buffer = 0;
unsigned int num_buffers_queued = 0;
ALint state = 0;
while (m_run_thread.IsSet())
{
// Block until we have a free buffer
int numBuffersProcessed;
palGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
if (numBuffers == numBuffersQueued && !numBuffersProcessed)
int num_buffers_processed;
palGetSourcei(m_source, AL_BUFFERS_PROCESSED, &num_buffers_processed);
if (num_buffers_queued == OAL_BUFFERS && !num_buffers_processed)
{
soundSyncEvent.Wait();
std::this_thread::sleep_for(std::chrono::milliseconds(1));
continue;
}
// Remove the Buffer from the Queue.
if (numBuffersProcessed)
if (num_buffers_processed)
{
ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
palSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
std::array<ALuint, OAL_BUFFERS> unqueued_buffer_ids;
palSourceUnqueueBuffers(m_source, num_buffers_processed, unqueued_buffer_ids.data());
err = CheckALError("unqueuing buffers");
numBuffersQueued -= numBuffersProcessed;
num_buffers_queued -= num_buffers_processed;
}
unsigned int numSamples = OAL_MAX_SAMPLES;
unsigned int min_frames = frames_per_buffer;
if (use_surround)
{
// DPL2 accepts 240 samples minimum (FWRDURATION)
unsigned int minSamples = 240;
std::array<float, OAL_MAX_FRAMES * SURROUND_CHANNELS> dpl2;
u32 rendered_frames = m_mixer->MixSurround(dpl2.data(), min_frames);
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
numSamples = m_mixer->MixSurround(dpl2, numSamples);
if (numSamples < minSamples)
if (rendered_frames < min_frames)
continue;
// 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 < numSamples; ++i)
for (u32 i = 0; i < rendered_frames; ++i)
{
dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
}
if (float32_capable)
{
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
numSamples * FRAME_SURROUND_FLOAT, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, dpl2.data(),
rendered_frames * FRAME_SURROUND_FLOAT, frequency);
}
else if (fixed32_capable)
{
int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
std::array<int, OAL_MAX_FRAMES * SURROUND_CHANNELS> surround_int32;
for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
for (u32 i = 0; i < rendered_frames * 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
@ -319,17 +342,17 @@ void OpenALStream::SoundLoop()
else if (dpl2[i] < INT_MIN)
surround_int32[i] = INT_MIN;
else
surround_int32[i] = (int)dpl2[i];
surround_int32[i] = static_cast<int>(dpl2[i]);
}
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32,
numSamples * FRAME_SURROUND_INT32, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, surround_int32.data(),
rendered_frames * FRAME_SURROUND_INT32, frequency);
}
else
{
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
std::array<short, OAL_MAX_FRAMES * SURROUND_CHANNELS> surround_short;
for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
{
dpl2[i] = dpl2[i] * (1 << 15);
if (dpl2[i] > SHRT_MAX)
@ -337,11 +360,11 @@ void OpenALStream::SoundLoop()
else if (dpl2[i] < SHRT_MIN)
surround_short[i] = SHRT_MIN;
else
surround_short[i] = (int)dpl2[i];
surround_short[i] = static_cast<int>(dpl2[i]);
}
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
numSamples * FRAME_SURROUND_SHORT, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN16, surround_short.data(),
rendered_frames * FRAME_SURROUND_SHORT, frequency);
}
err = CheckALError("buffering data");
@ -355,59 +378,26 @@ void OpenALStream::SoundLoop()
}
else
{
numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
u32 rendered_frames = m_mixer->Mix(m_realtime_buffer.data(), min_frames);
// Convert the samples from short to float
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15);
if (!numSamples)
if (!rendered_frames)
continue;
if (float32_capable)
{
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
numSamples * FRAME_STEREO_FLOAT, ulFrequency);
err = CheckALError("buffering float32 data");
if (err == AL_INVALID_ENUM)
{
float32_capable = false;
}
}
else if (fixed32_capable)
{
// Clamping is not necessary here, samples are always between (-1,1)
int stereo_int32[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32,
numSamples * FRAME_STEREO_INT32, ulFrequency);
}
else
{
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
numSamples * FRAME_STEREO_SHORT, ulFrequency);
}
palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO16, m_realtime_buffer.data(),
rendered_frames * FRAME_STEREO_SHORT, frequency);
}
palSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
palSourceQueueBuffers(m_source, 1, &m_buffers[next_buffer]);
err = CheckALError("queuing buffers");
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
num_buffers_queued++;
next_buffer = (next_buffer + 1) % OAL_BUFFERS;
palGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
palGetSourcei(m_source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
palSourcePlay(uiSource);
palSourcePlay(m_source);
err = CheckALError("occurred resuming playback");
}
}

24
Source/Core/AudioCommon/OpenALStream.h
View File

@ -17,17 +17,15 @@
#include <OpenAL/include/alc.h>
#include <OpenAL/include/alext.h>
#define SFX_MAX_SOURCE 1
#define OAL_MAX_BUFFERS 32
#define OAL_MAX_SAMPLES 256
// OpenAL requires a minimum of two buffers, three or more recommended
#define OAL_BUFFERS 3
#define OAL_MAX_FRAMES 4096
#define STEREO_CHANNELS 2
#define SURROUND_CHANNELS 6 // number of channels in surround mode
#define SIZE_SHORT 2
#define SIZE_INT32 4
#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_STEREO_INT32 STEREO_CHANNELS* SIZE_INT32
#define FRAME_SURROUND_FLOAT SURROUND_CHANNELS* SIZE_FLOAT
#define FRAME_SURROUND_SHORT SURROUND_CHANNELS* SIZE_SHORT
#define FRAME_SURROUND_INT32 SURROUND_CHANNELS* SIZE_INT32
@ -56,7 +54,7 @@ class OpenALStream final : public SoundStream
{
#ifdef _WIN32
public:
OpenALStream() : uiSource(0) {}
OpenALStream() : m_source(0) {}
bool Start() override;
void SoundLoop() override;
void SetVolume(int volume) override;
@ -67,17 +65,15 @@ public:
static bool isValid();
private:
std::thread thread;
std::thread m_thread;
Common::Flag m_run_thread;
Common::Event soundSyncEvent;
Common::Event m_sound_sync_event;
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
float sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS];
ALuint uiSource;
ALfloat fVolume;
std::vector<short> m_realtime_buffer;
std::array<ALuint, OAL_BUFFERS> m_buffers;
ALuint m_source;
ALfloat m_volume;
u8 numBuffers;
#endif // _WIN32
};

6
Source/Core/Core/ConfigManager.cpp
View File

@ -254,7 +254,7 @@ void SConfig::SaveCoreSettings(IniFile& ini)
core->Set("SelectedLanguage", SelectedLanguage);
core->Set("OverrideGCLang", bOverrideGCLanguage);
core->Set("DPL2Decoder", bDPL2Decoder);
core->Set("Latency", iLatency);
core->Set("AudioLatency", iLatency);
core->Set("AudioStretch", m_audio_stretch);
core->Set("AudioStretchMaxLatency", m_audio_stretch_max_latency);
core->Set("MemcardAPath", m_strMemoryCardA);
@ -568,7 +568,7 @@ void SConfig::LoadCoreSettings(IniFile& ini)
core->Get("SelectedLanguage", &SelectedLanguage, 0);
core->Get("OverrideGCLang", &bOverrideGCLanguage, false);
core->Get("DPL2Decoder", &bDPL2Decoder, false);
core->Get("Latency", &iLatency, 5);
core->Get("AudioLatency", &iLatency, 20);
core->Get("AudioStretch", &m_audio_stretch, false);
core->Get("AudioStretchMaxLatency", &m_audio_stretch_max_latency, 80);
core->Get("MemcardAPath", &m_strMemoryCardA);
@ -831,7 +831,7 @@ void SConfig::LoadDefaults()
bOverrideGCLanguage = false;
bWii = false;
bDPL2Decoder = false;
iLatency = 14;
iLatency = 20;
m_audio_stretch = false;
m_audio_stretch_max_latency = 80;

2
Source/Core/Core/ConfigManager.h
View File

@ -109,7 +109,7 @@ struct SConfig : NonCopyable
bool bCopyWiiSaveNetplay = true;
bool bDPL2Decoder = false;
int iLatency = 14;
int iLatency = 20;
bool m_audio_stretch = false;
int m_audio_stretch_max_latency = 80;

4
Source/Core/DolphinWX/Config/AudioConfigPane.cpp
View File

@ -45,8 +45,8 @@ void AudioConfigPane::InitializeGUI()
m_audio_backend_choice =
new wxChoice(this, wxID_ANY, wxDefaultPosition, wxDefaultSize, m_audio_backend_strings);
m_audio_latency_spinctrl =
new wxSpinCtrl(this, wxID_ANY, "", wxDefaultPosition, wxDefaultSize, wxSP_ARROW_KEYS, 0, 30);
m_audio_latency_label = new wxStaticText(this, wxID_ANY, _("Latency:"));
new wxSpinCtrl(this, wxID_ANY, "", wxDefaultPosition, wxDefaultSize, wxSP_ARROW_KEYS, 0, 200);
m_audio_latency_label = new wxStaticText(this, wxID_ANY, _("Latency (ms):"));
m_stretch_checkbox = new wxCheckBox(this, wxID_ANY, _("Enable Audio Stretching"));
m_stretch_label = new wxStaticText(this, wxID_ANY, _("Buffer Size:"));