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

Source/Core/AudioCommon/Src/AudioCommon.cpp

@@ -34,19 +34,19 @@ namespace AudioCommon
 			mixer = new CMixer();
 
 		std::string backend = ac_Config.sBackend;
-		if (backend == BACKEND_COREAUDIO   && CoreAudioSound::isValid()) 
+		if (backend == BACKEND_OPENAL           && OpenALStream::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()) 
 			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()) 

Source/Core/AudioCommon/Src/Mixer.cpp

@@ -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()

Source/Core/AudioCommon/Src/Mixer.h

@@ -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);

Source/Core/Core/Src/HW/AudioInterface.cpp

@@ -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;
+			if (g_SampleRate == 48000)
-			*_pDestBuffer++ = (pcm[pos*2+1] * rvolume) >> 8;
-
-			pos++;
-			if (pos == 28) 
 			{
-				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;

Source/Core/Core/Src/HW/DSP.cpp

@@ -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.ReadAddress = g_audioDMA.SourceAddress;
-			{
+			g_audioDMA.BlocksLeft = g_audioDMA.AudioDMAControl.NumBlocks;
-				g_audioDMA.BlocksLeft = g_audioDMA.AudioDMAControl.NumBlocks;
-				g_audioDMA.ReadAddress = g_audioDMA.SourceAddress;
-			}
 			// DEBUG_LOG(DSPLLE, "ADMA read addresses: %08x", g_audioDMA.ReadAddress);
 			GenerateDSPInterrupt(DSP::INT_AID);
 		}

Source/Plugins/Plugin_DSP_HLE/Src/Globals.cpp

@@ -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];
+}

Source/Plugins/Plugin_DSP_HLE/Src/Globals.h

@@ -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;
 

Source/Plugins/Plugin_DSP_HLE/Src/main.cpp

@@ -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);

Source/Plugins/Plugin_DSP_LLE/Src/DSPConfigDlgLLE.cpp

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

Source/Plugins/Plugin_DSP_LLE/Src/DSPConfigDlgLLE.h

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

Source/Plugins/Plugin_DSP_LLE/Src/main.cpp

@@ -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();