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Sound System Rework: Phase 1 

. Fixed Sample Rate for DSP
  (Now if your CPU is capable to run game at 100%, you will get pure sound without buzz or static noise)

. Fixed Sample Rate for AI
  (Now if your CPU is capable to run game at 100%, you will get sync'ed video and audio)

. Fixed Backend list for DSPLLE

. Improved Aduio DMA a bit
  (There might be a completely redesign in following phases)

WARNING: The whole rework will take time to complete.
This commit is compilable, but could be unstable.
So you can try it and test it but don't take it as a release rev!

git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@4717 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
ayuanx 2009-12-22 07:26:30 +00:00
parent a1fefe870c
commit 24d8839793
11 changed files with 125 additions and 47 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

32
Source/Core/AudioCommon/Src/AudioCommon.cpp
View File

@ -34,19 +34,19 @@ namespace AudioCommon
mixer = new CMixer();
std::string backend = ac_Config.sBackend;
if (backend == BACKEND_COREAUDIO && CoreAudioSound::isValid())
soundStream = new CoreAudioSound(mixer);
if (backend == BACKEND_DIRECTSOUND && DSound::isValid())
soundStream = new DSound(mixer, g_dspInitialize.hWnd);
if (backend == BACKEND_AOSOUND && AOSound::isValid())
soundStream = new AOSound(mixer);
if (backend == BACKEND_OPENAL && OpenALStream::isValid())
if (backend == BACKEND_OPENAL && OpenALStream::isValid())
soundStream = new OpenALStream(mixer);
if (backend == BACKEND_ALSA && AlsaSound::isValid())
else if (backend == BACKEND_DIRECTSOUND && DSound::isValid())
soundStream = new DSound(mixer, g_dspInitialize.hWnd);
else if (backend == BACKEND_AOSOUND && AOSound::isValid())
soundStream = new AOSound(mixer);
else if (backend == BACKEND_ALSA && AlsaSound::isValid())
soundStream = new AlsaSound(mixer);
if (backend == BACKEND_PULSEAUDIO && PulseAudio::isValid())
else if (backend == BACKEND_COREAUDIO && CoreAudioSound::isValid())
soundStream = new CoreAudioSound(mixer);
else if (backend == BACKEND_PULSEAUDIO && PulseAudio::isValid())
soundStream = new PulseAudio(mixer);
if (backend == BACKEND_NULL && NullSound::isValid())
else if (backend == BACKEND_NULL && NullSound::isValid())
soundStream = new NullSound(mixer);
if (soundStream != NULL)
@ -60,12 +60,10 @@ namespace AudioCommon
soundStream->StartLogAudio(FULL_DUMP_DIR g_Config.recordFile);
}
*/
return soundStream;
}
PanicAlert("Could not initialize backend %s, falling back to NULL", backend.c_str());
}
PanicAlert("Sound backend %s is not valid, falling back to NULL", backend.c_str());
delete soundStream;
@ -77,7 +75,7 @@ namespace AudioCommon
void ShutdownSoundStream()
{
NOTICE_LOG(DSPHLE, "Shutting down sound stream");
INFO_LOG(DSPHLE, "Shutting down sound stream");
if (soundStream)
{
@ -94,16 +92,16 @@ namespace AudioCommon
{
std::vector<std::string> backends;
if (CoreAudioSound::isValid())
backends.push_back(BACKEND_COREAUDIO);
if (DSound::isValid())
backends.push_back(BACKEND_DIRECTSOUND);
if (AOSound::isValid())
backends.push_back(BACKEND_AOSOUND);
if (OpenALStream::isValid())
backends.push_back(BACKEND_OPENAL);
if (AOSound::isValid())
backends.push_back(BACKEND_AOSOUND);
if (AlsaSound::isValid())
backends.push_back(BACKEND_ALSA);
if (CoreAudioSound::isValid())
backends.push_back(BACKEND_COREAUDIO);
if (PulseAudio::isValid())
backends.push_back(BACKEND_PULSEAUDIO);
if (NullSound::isValid())

22
Source/Core/AudioCommon/Src/Mixer.cpp
View File

@ -81,9 +81,6 @@ void CMixer::PushSamples(short *samples, int num_stereo_samples, int core_sample
sample_queue.push((s16)0);
}
push_sync.Leave();
static int PV1l=0,PV2l=0,PV3l=0,PV4l=0;
static int PV1r=0,PV2r=0,PV3r=0,PV4r=0;
static int acc=0;
#ifdef _WIN32
if (GetAsyncKeyState(VK_TAB))
@ -113,6 +110,23 @@ void CMixer::PushSamples(short *samples, int num_stereo_samples, int core_sample
// -----------------------------------------------------------------------
push_sync.Enter();
while (num_stereo_samples)
{
sample_queue.push(Common::swap16(*samples));
samples++;
sample_queue.push(Common::swap16(*samples));
samples++;
m_queueSize += 2;
num_stereo_samples--;
}
push_sync.Leave();
return;
/*
static int PV1l=0,PV2l=0,PV3l=0,PV4l=0;
static int PV1r=0,PV2r=0,PV3r=0,PV4r=0;
static int acc=0;
while (num_stereo_samples) {
acc += core_sample_rate;
while (num_stereo_samples && (acc >= 48000)) {
@ -170,6 +184,8 @@ void CMixer::PushSamples(short *samples, int num_stereo_samples, int core_sample
m_queueSize += 2;
}
push_sync.Leave();
*/
}
int CMixer::GetNumSamples()

12
Source/Core/AudioCommon/Src/Mixer.h
View File

@ -29,7 +29,17 @@
class CMixer {
public:
CMixer() : m_sampleRate(48000),m_bits(16),m_channels(2), m_mode(2), m_HLEready(false),m_queueSize(0) {}
// AyuanX: Mixer sample rate is fixed to 32khz for now
// if any game sets DSP sample rate to 48khz, we are doomed
// TODO: Fix this somehow!
CMixer(unsigned int SampleRate = 32000)
: m_sampleRate(SampleRate)
, m_bits(16)
, m_channels(2)
, m_mode(2)
, m_HLEready(false)
, m_queueSize(0)
{}
// Called from audio threads
virtual int Mix(short *sample, int numSamples);

41
Source/Core/Core/Src/HW/AudioInterface.cpp
View File

@ -60,7 +60,7 @@ union AICR
unsigned AIINTVLD : 1; // This bit controls whether AIINT is affected by the AIIT register
// matching AISLRCNT. Once set, AIINT will hold
unsigned SCRESET : 1; // write to reset counter
unsigned DSPFR : 1; // DSP Frequency (0=32khz 1=48khz)
unsigned DSPFR : 1; // DSP Frequency (0=48khz 1=32khz) WTF, who designed this?
unsigned :25;
};
u32 hex;
@ -181,9 +181,8 @@ void Write32(const u32 _Value, const u32 _Address)
g_AudioRegister.m_Control.DSPFR = tmpAICtrl.DSPFR;
}
g_SampleRate = tmpAICtrl.AFR ? 32000 : 48000;
g_SampleRate = tmpAICtrl.AFR ? 48000 : 32000;
g_DSPSampleRate = tmpAICtrl.DSPFR ? 32000 : 48000;
// PanicAlert("Sample rate %i %i", g_Aui, g_SampleRate);
g_CPUCyclesPerSample = SystemTimers::GetTicksPerSecond() / g_SampleRate;
@ -276,29 +275,49 @@ u32 Callback_GetStreaming(short* _pDestBuffer, u32 _numSamples)
const int lvolume = g_AudioRegister.m_Volume.leftVolume;
const int rvolume = g_AudioRegister.m_Volume.rightVolume;
// AyuanX: I hate this, but for now we have to do down-sampling to support 48khz
if (g_SampleRate == 48000)
{
_dbg_assert_msg_(AUDIO_INTERFACE, !(_numSamples % 2), "Number of Samples: %i must be even!", _numSamples);
_numSamples = _numSamples * 3 / 2;
}
short pcm_l = 0;
short pcm_r = 0;
for (unsigned int i = 0; i < _numSamples; i++)
{
if (pos == 0)
{
ReadStreamBlock(pcm);
}
*_pDestBuffer++ = (pcm[pos*2] * lvolume) >> 8;
*_pDestBuffer++ = (pcm[pos*2+1] * rvolume) >> 8;
pos++;
if (pos == 28)
if (g_SampleRate == 48000)
{
pos = 0;
if (i % 3)
{
*_pDestBuffer++ = ((pcm_l / 2 + pcm[pos*2] / 2) * lvolume) >> 8;
*_pDestBuffer++ = ((pcm_r / 2 + pcm[pos*2+1] / 2) * rvolume) >> 8;
}
pcm_l = pcm[pos*2];
pcm_r = pcm[pos*2+1];
}
else
{
*_pDestBuffer++ = (pcm[pos*2] * lvolume) >> 8;
*_pDestBuffer++ = (pcm[pos*2+1] * rvolume) >> 8;
}
if (++pos == 28)
pos = 0;
}
}
else
{
// AyuanX: We have already preset those bytes, no need to do this again
/*
for (unsigned int i = 0; i < _numSamples * 2; i++)
{
_pDestBuffer[i] = 0; //silence!
}
*/
}
return _numSamples;

45
Source/Core/Core/Src/HW/DSP.cpp
View File

@ -63,16 +63,17 @@ enum
AR_SIZE = 0x5012, // These names are a good guess at best
AR_MODE = 0x5016, //
AR_REFRESH = 0x501a, //
AUDIO_DMA_START_HI = 0x5030,
AUDIO_DMA_START_LO = 0x5032,
AUDIO_DMA_CONTROL_LEN = 0x5036,
AUDIO_DMA_BLOCKS_LEFT = 0x503A,
AR_DMA_MMADDR_H = 0x5020,
AR_DMA_MMADDR_L = 0x5022,
AR_DMA_ARADDR_H = 0x5024,
AR_DMA_ARADDR_L = 0x5026,
AR_DMA_CNT_H = 0x5028,
AR_DMA_CNT_L = 0x502A
AR_DMA_CNT_L = 0x502A,
AUDIO_DMA_START_HI = 0x5030,
AUDIO_DMA_START_LO = 0x5032,
AUDIO_DMA_BLOCKS_LENGTH = 0x5034, // Ever used?
AUDIO_DMA_CONTROL_LEN = 0x5036,
AUDIO_DMA_BLOCKS_LEFT = 0x503A,
};
// UARAMCount
@ -128,7 +129,7 @@ union UAudioDMAControl
struct
{
unsigned NumBlocks : 15;
unsigned Reload : 1;
unsigned Enable : 1;
};
UAudioDMAControl(u16 _Hex = 0) : Hex(_Hex)
@ -378,6 +379,27 @@ void Write16(const u16 _Value, const u32 _Address)
g_dspState.DSPControl.DSPHalt = tmpControl.DSPHalt;
g_dspState.DSPControl.DSPInit = tmpControl.DSPInit;
// AyuanX: WTF, sample rate between AI & DSP can be different?
// This is a big problem especially when our mixer is fixed to 32000
// TODO: Try to support these!
// More info: AudioCommon/Mixer.h, HW/AudioInterface.cpp
static bool FirstTimeWarning = false;
if (!FirstTimeWarning)
{
if (!g_dspState.DSPControl.DSPHalt && g_dspState.DSPControl.DSPInit)
{
// It's time to check now, and we do this only once
FirstTimeWarning = true;
if (AudioInterface::GetAISampleRate() != 32000 || AudioInterface::GetDSPSampleRate() != 32000)
{
WARN_LOG(DSPINTERFACE, "Unsupported Sample Rate, AI:%i, DSP:%i", AudioInterface::GetAISampleRate(), AudioInterface::GetDSPSampleRate());
if (AudioInterface::GetDSPSampleRate() != 32000)
PanicAlert("DSPINTERFACE: Unsupported Sample Rate, AI:%i, DSP:%i\n"
"You may get incorrect sound output, please report!", AudioInterface::GetAISampleRate(), AudioInterface::GetDSPSampleRate());
}
}
}
// Interrupt (mask)
g_dspState.DSPControl.AID_mask = tmpControl.AID_mask;
g_dspState.DSPControl.ARAM_mask = tmpControl.ARAM_mask;
@ -457,8 +479,8 @@ void Write16(const u16 _Value, const u32 _Address)
case AUDIO_DMA_CONTROL_LEN: // called by AIStartDMA()
g_audioDMA.AudioDMAControl.Hex = _Value;
g_audioDMA.BlocksLeft = g_audioDMA.AudioDMAControl.NumBlocks;
g_audioDMA.ReadAddress = g_audioDMA.SourceAddress;
g_audioDMA.BlocksLeft = g_audioDMA.AudioDMAControl.NumBlocks;
INFO_LOG(DSPINTERFACE, "AID DMA started - source address %08x, length %i blocks", g_audioDMA.SourceAddress, g_audioDMA.AudioDMAControl.NumBlocks);
break;
@ -475,7 +497,7 @@ void Write16(const u16 _Value, const u32 _Address)
// This happens at 4 khz, since 32 bytes at 4khz = 4 bytes at 32 khz (16bit stereo pcm)
void UpdateAudioDMA()
{
if (g_audioDMA.BlocksLeft)
if (g_audioDMA.AudioDMAControl.Enable && g_audioDMA.BlocksLeft)
{
// Read audio at g_audioDMA.ReadAddress in RAM and push onto an
// external audio fifo in the emulator, to be mixed with the disc
@ -486,11 +508,8 @@ void UpdateAudioDMA()
g_audioDMA.BlocksLeft--;
if (g_audioDMA.BlocksLeft == 0)
{
if (g_audioDMA.AudioDMAControl.Reload)
{
g_audioDMA.BlocksLeft = g_audioDMA.AudioDMAControl.NumBlocks;
g_audioDMA.ReadAddress = g_audioDMA.SourceAddress;
}
g_audioDMA.ReadAddress = g_audioDMA.SourceAddress;
g_audioDMA.BlocksLeft = g_audioDMA.AudioDMAControl.NumBlocks;
// DEBUG_LOG(DSPLLE, "ADMA read addresses: %08x", g_audioDMA.ReadAddress);
GenerateDSPInterrupt(DSP::INT_AID);
}

6
Source/Plugins/Plugin_DSP_HLE/Src/Globals.cpp
View File

@ -58,3 +58,9 @@ float Memory_Read_Float(u32 _uAddress)
u32 uTemp = Memory_Read_U32(_uAddress);
return *(float*)&uTemp;
}
void* Memory_Get_Pointer(u32 _uAddress)
{
_uAddress &= RAM_MASK;
return &g_pMemory[_uAddress];
}

1
Source/Plugins/Plugin_DSP_HLE/Src/Globals.h
View File

@ -43,6 +43,7 @@ u8 Memory_Read_U8(u32 _uAddress);
u16 Memory_Read_U16(u32 _uAddress);
u32 Memory_Read_U32(u32 _uAddress);
float Memory_Read_Float(u32 _uAddress);
void* Memory_Get_Pointer(u32 _uAddress);
extern PLUGIN_GLOBALS* globals;

6
Source/Plugins/Plugin_DSP_HLE/Src/main.cpp
View File

@ -332,8 +332,10 @@ void DSP_SendAIBuffer(unsigned int address, int sample_rate)
}
CMixer* pMixer = soundStream->GetMixer();
if (pMixer)
if (pMixer && address)
{
short* samples = (short*)Memory_Get_Pointer(address);
/*
short samples[16] = {0}; // interleaved stereo
if (address)
{
@ -346,7 +348,7 @@ void DSP_SendAIBuffer(unsigned int address, int sample_rate)
//if (log_ai)
// g_wave_writer.AddStereoSamples(samples, 8);
}
*/
// sample_rate is usually 32k here,
// see Core/DSP/DSP.cpp for better information
pMixer->PushSamples(samples, 32 / 4, sample_rate);

5
Source/Plugins/Plugin_DSP_LLE/Src/DSPConfigDlgLLE.cpp
View File

@ -105,6 +105,11 @@ void DSPConfigDialogLLE::AddBackend(const char* backend)
#endif
}
void DSPConfigDialogLLE::ClearBackends()
{
m_BackendSelection->Clear();
}
DSPConfigDialogLLE::~DSPConfigDialogLLE()
{
}

1
Source/Plugins/Plugin_DSP_LLE/Src/DSPConfigDlgLLE.h
View File

@ -35,6 +35,7 @@ public:
long style = wxDEFAULT_DIALOG_STYLE);
virtual ~DSPConfigDialogLLE();
void AddBackend(const char *backend);
void ClearBackends();
private:
DECLARE_EVENT_TABLE();

1
Source/Plugins/Plugin_DSP_LLE/Src/main.cpp
View File

@ -145,6 +145,7 @@ void DllConfig(HWND _hParent)
else if (!m_ConfigFrame->GetParent()->IsShown())
m_ConfigFrame->Close(true);
m_ConfigFrame->ClearBackends();
// add backends
std::vector<std::string> backends = AudioCommon::GetSoundBackends();