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Changed OpenAL latency setting to really reflect how much time it is. 

Before these changes each value of latency were actually 5ms, with a
minimum latency of ~10 ms. If it was set to 4 ms on the UI, the actual
latency was 10 + 5 * 4 = 30 ms.
Now 30 ms on the UI means 30 ms on the backend.
This commit is contained in:
LAGonauta 2017-06-17 07:43:37 -03:00 committed by lfsafady
parent 94ba78d717
commit 8fd1af6783
5 changed files with 54 additions and 28 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

57
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -138,7 +138,7 @@ void OpenALStream::Stop()
// Clean up buffers and sources
palDeleteSources(1, &uiSource);
uiSource = 0;
palDeleteBuffers(numBuffers, uiBuffers);
palDeleteBuffers(OAL_BUFFERS, uiBuffers.data());
ALCcontext* pContext = palcGetCurrentContext();
ALCdevice* pDevice = palcGetContextsDevice(pContext);
@ -230,16 +230,43 @@ void OpenALStream::SoundLoop()
bool fixed32_capable = IsCreativeXFi();
u32 ulFrequency = m_mixer->GetSampleRate();
numBuffers = SConfig::GetInstance().iLatency + 2; // OpenAL requires a minimum of two buffers
memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
u32 frames_per_buffer;
// Can't have zero samples per buffer
if (SConfig::GetInstance().iLatency > 0)
{
frames_per_buffer = ulFrequency / 1000 * SConfig::GetInstance().iLatency / OAL_BUFFERS;
}
else
{
frames_per_buffer = ulFrequency / 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?
realtimeBuffer.resize(frames_per_buffer * STEREO_CHANNELS);
sampleBuffer.resize(frames_per_buffer * STEREO_CHANNELS);
uiSource = 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*)uiBuffers.data());
err = CheckALError("generating buffers");
// Generate a Source to playback the Buffers
@ -261,7 +288,7 @@ void OpenALStream::SoundLoop()
// Block until we have a free buffer
int numBuffersProcessed;
palGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
if (numBuffers == numBuffersQueued && !numBuffersProcessed)
if (OAL_BUFFERS == numBuffersQueued && !numBuffersProcessed)
{
soundSyncEvent.Wait();
continue;
@ -270,21 +297,21 @@ void OpenALStream::SoundLoop()
// Remove the Buffer from the Queue.
if (numBuffersProcessed)
{
ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
ALuint unqueuedBufferIds[OAL_BUFFERS];
palSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
err = CheckALError("unqueuing buffers");
numBuffersQueued -= numBuffersProcessed;
}
unsigned int numSamples = OAL_MAX_SAMPLES;
unsigned int numSamples = frames_per_buffer;
if (use_surround)
{
// DPL2 accepts 240 samples minimum (FWRDURATION)
unsigned int minSamples = 240;
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
float dpl2[OAL_MAX_FRAMES * SURROUND_CHANNELS];
numSamples = m_mixer->MixSurround(dpl2, numSamples);
if (numSamples < minSamples)
@ -306,7 +333,7 @@ void OpenALStream::SoundLoop()
}
else if (fixed32_capable)
{
int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
int surround_int32[OAL_MAX_FRAMES * SURROUND_CHANNELS];
for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
{
@ -327,7 +354,7 @@ void OpenALStream::SoundLoop()
}
else
{
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
short surround_short[OAL_MAX_FRAMES * SURROUND_CHANNELS];
for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
{
@ -355,7 +382,7 @@ void OpenALStream::SoundLoop()
}
else
{
numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
numSamples = m_mixer->Mix(realtimeBuffer.data(), numSamples);
// Convert the samples from short to float
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
@ -366,7 +393,7 @@ void OpenALStream::SoundLoop()
if (float32_capable)
{
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer.data(),
numSamples * FRAME_STEREO_FLOAT, ulFrequency);
err = CheckALError("buffering float32 data");
@ -378,7 +405,7 @@ void OpenALStream::SoundLoop()
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];
int stereo_int32[OAL_MAX_FRAMES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
@ -388,7 +415,7 @@ void OpenALStream::SoundLoop()
else
{
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
short stereo[OAL_MAX_FRAMES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
@ -401,7 +428,7 @@ void OpenALStream::SoundLoop()
err = CheckALError("queuing buffers");
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
nextBuffer = (nextBuffer + 1) % OAL_BUFFERS;
palGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)

13
Source/Core/AudioCommon/OpenALStream.h
View File

@ -17,9 +17,9 @@
#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
@ -72,12 +72,11 @@ private:
Common::Event soundSyncEvent;
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
float sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS];
std::vector<short> realtimeBuffer;
std::vector<float> sampleBuffer;
std::array<ALuint, OAL_BUFFERS> uiBuffers;
ALuint uiSource;
ALfloat fVolume;
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:"));