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OpenALStream: Remove audio stretching

This commit is contained in:
MerryMage 2017-04-10 14:57:24 +01:00
parent 29344cb5ff
commit 6a4945090c
2 changed files with 19 additions and 79 deletions
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84
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -20,8 +20,6 @@
#pragma comment(lib, "openal32.lib") #pragma comment(lib, "openal32.lib")
#endif #endif
static soundtouch::SoundTouch soundTouch;
// //
// AyuanX: Spec says OpenAL1.1 is thread safe already // AyuanX: Spec says OpenAL1.1 is thread safe already
// //
@ -71,7 +69,6 @@ bool OpenALStream::Start()
// Initialize DPL2 parameters // Initialize DPL2 parameters
DPL2Reset(); DPL2Reset();
soundTouch.clear();
return bReturn; return bReturn;
} }
@ -81,8 +78,6 @@ void OpenALStream::Stop()
// kick the thread if it's waiting // kick the thread if it's waiting
soundSyncEvent.Set(); soundSyncEvent.Set();
soundTouch.clear();
thread.join(); thread.join();
alSourceStop(uiSource); alSourceStop(uiSource);
@ -120,7 +115,6 @@ void OpenALStream::Clear(bool mute)
if (m_muted) if (m_muted)
{ {
soundTouch.clear();
alSourceStop(uiSource); alSourceStop(uiSource);
} }
else else
@ -213,15 +207,6 @@ void OpenALStream::SoundLoop()
unsigned int numBuffersQueued = 0; unsigned int numBuffersQueued = 0;
ALint iState = 0; ALint iState = 0;
soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency);
soundTouch.setTempo(1.0);
soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
while (m_run_thread.IsSet()) while (m_run_thread.IsSet())
{ {
// Block until we have a free buffer // Block until we have a free buffer
@ -243,62 +228,29 @@ void OpenALStream::SoundLoop()
numBuffersQueued -= numBuffersProcessed; numBuffersQueued -= numBuffersProcessed;
} }
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD. // DPL2 accepts 240 samples minimum (FWRDURATION)
const u32 stereo_16_bit_size = 4; unsigned int minSamples = surround_capable ? 240 : 0;
const u32 dma_length = 32;
const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
u64 audio_dma_period = SystemTimers::GetTicksPerSecond() /
(AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
u64 num_samples_to_render =
(audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
unsigned int numSamples = (unsigned int)num_samples_to_render; unsigned int numSamples = OAL_MAX_SAMPLES;
unsigned int minSamples = numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
surround_capable ? 240 : 0; // DPL2 accepts 240 samples minimum (FWRDURATION)
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);
// Convert the samples from short to float // Convert the samples from short to float
float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i) for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
dest[i] = (float)realtimeBuffer[i] / (1 << 15); sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15);
soundTouch.putSamples(dest, numSamples); if (numSamples <= minSamples)
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)
{
soundTouch.clear();
}
}
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
if (nSamples <= minSamples)
continue; continue;
if (surround_capable) if (surround_capable)
{ {
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS]; float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
DPL2Decode(sampleBuffer, nSamples, dpl2); DPL2Decode(sampleBuffer, numSamples, dpl2);
// zero-out the subwoofer channel - DPL2Decode generates a pretty // 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 // 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. // AL_FORMAT_50CHN32 to make this super-explicit.
// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
for (u32 i = 0; i < nSamples; ++i) for (u32 i = 0; i < numSamples; ++i)
{ {
dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f; dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
} }
@ -306,13 +258,13 @@ void OpenALStream::SoundLoop()
if (float32_capable) if (float32_capable)
{ {
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2, alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
nSamples * FRAME_SURROUND_FLOAT, ulFrequency); numSamples * FRAME_SURROUND_FLOAT, ulFrequency);
} }
else if (fixed32_capable) else if (fixed32_capable)
{ {
int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS]; int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i) for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
{ {
// For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1. // 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 // Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
@ -327,13 +279,13 @@ void OpenALStream::SoundLoop()
} }
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32, alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32,
nSamples * FRAME_SURROUND_INT32, ulFrequency); numSamples * FRAME_SURROUND_INT32, ulFrequency);
} }
else else
{ {
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS]; short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i) for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
{ {
dpl2[i] = dpl2[i] * (1 << 15); dpl2[i] = dpl2[i] * (1 << 15);
if (dpl2[i] > SHRT_MAX) if (dpl2[i] > SHRT_MAX)
@ -345,7 +297,7 @@ void OpenALStream::SoundLoop()
} }
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short, alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
nSamples * FRAME_SURROUND_SHORT, ulFrequency); numSamples * FRAME_SURROUND_SHORT, ulFrequency);
} }
err = CheckALError("buffering data"); err = CheckALError("buffering data");
@ -362,7 +314,7 @@ void OpenALStream::SoundLoop()
if (float32_capable) if (float32_capable)
{ {
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
nSamples * FRAME_STEREO_FLOAT, ulFrequency); numSamples * FRAME_STEREO_FLOAT, ulFrequency);
err = CheckALError("buffering float32 data"); err = CheckALError("buffering float32 data");
if (err == AL_INVALID_ENUM) if (err == AL_INVALID_ENUM)
@ -374,21 +326,21 @@ void OpenALStream::SoundLoop()
{ {
// Clamping is not necessary here, samples are always between (-1,1) // 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_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i) for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31)); stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32, alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32,
nSamples * FRAME_STEREO_INT32, ulFrequency); numSamples * FRAME_STEREO_INT32, ulFrequency);
} }
else else
{ {
// Convert the samples from float to short // Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS]; short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i) for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15)); stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo, alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
nSamples * FRAME_STEREO_SHORT, ulFrequency); numSamples * FRAME_STEREO_SHORT, ulFrequency);
} }
} }

14
Source/Core/AudioCommon/OpenALStream.h
View File

@ -26,18 +26,6 @@
#include <AL/alext.h> #include <AL/alext.h>
#endif #endif
#ifdef __APPLE__
// Avoid conflict with objc.h (on Windows, ST uses the system BOOL type, so this doesn't work)
#define BOOL SoundTouch_BOOL
#endif
#include <soundtouch/STTypes.h>
#include <soundtouch/SoundTouch.h>
#ifdef __APPLE__
#undef BOOL
#endif
#define SFX_MAX_SOURCE 1 #define SFX_MAX_SOURCE 1
#define OAL_MAX_BUFFERS 32 #define OAL_MAX_BUFFERS 32
#define OAL_MAX_SAMPLES 256 #define OAL_MAX_SAMPLES 256
@ -89,7 +77,7 @@ private:
Common::Event soundSyncEvent; Common::Event soundSyncEvent;
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS]; short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS]; float sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS]; ALuint uiBuffers[OAL_MAX_BUFFERS];
ALuint uiSource; ALuint uiSource;
ALfloat fVolume; ALfloat fVolume;