Sound System Rework: Phase 2

. Performance boost
  (Completely non-blocking between Sound thread and CPU thread, in the meantime keeping them thread safe)

. Both 32KHz & 48KHz sound can be handled properly now
  (But up-sampling is still not implemented, and I don't think any game requires it.)

. Strategy adjustment
  When your PC is *NOT* capable to run the game at 100%:
  >> DSound    Could yield more fluent sound than OpenAL sometimes, but you will lose the sync between video & audio (since audio is played before video to guarantee fluency)
  >> OpenAL    Ensures video & audio are always sync'ed, but sound could be intermittent(to let slow video catch up)

. Changed default frame limit to: Auto
  (Somehow this can dramatically decrease the chance of wiimote desync in game NSMB)

git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@4724 8ced0084-cf51-0410-be5f-012b33b47a6e

Source/Core/AudioCommon/Src/AudioCommon.h

@@ -30,6 +30,31 @@ extern DSPInitialize g_dspInitialize;
 extern SoundStream *soundStream;
 extern AudioCommonConfig ac_Config;
 
+// UDSPControl
+union UDSPControl
+{
+	u16 Hex;
+	struct
+	{
+		unsigned DSPReset       : 1; // Write 1 to reset and waits for 0
+		unsigned DSPAssertInt   : 1;
+		unsigned DSPHalt        : 1;
+
+		unsigned AI             : 1;
+		unsigned AI_mask        : 1;
+		unsigned ARAM           : 1;
+		unsigned ARAM_mask      : 1;
+		unsigned DSP            : 1;
+		unsigned DSP_mask       : 1;
+
+		unsigned ARAM_DMAState  : 1; // DSPGetDMAStatus() uses this flag
+		unsigned DSPInitCode    : 1;
+		unsigned DSPInit        : 1; // DSPInit() writes to this flag
+		unsigned pad            : 4;
+	};
+	UDSPControl(u16 _Hex = 0) : Hex(_Hex) {}
+};
+
 namespace AudioCommon 
 {
 	SoundStream *InitSoundStream(CMixer *mixer = NULL);

Source/Core/AudioCommon/Src/DSoundStream.cpp

@@ -111,13 +111,11 @@ void DSound::SoundLoop()
 		int numBytesToRender = FIX128(ModBufferSize(currentPos - lastPos));
 		if (numBytesToRender >= 256)
 		{
-			if (numBytesToRender > sizeof(realtimeBuffer))
+			if (numBytesToRender > sizeof(realtimeBuffer) * sizeof(short))
 				PanicAlert("soundThread: too big render call");
-			m_mixer->Mix(realtimeBuffer, numBytesToRender >> 2);
+			m_mixer->Mix(realtimeBuffer, numBytesToRender / 4);
 			WriteDataToBuffer(lastPos, (char*)realtimeBuffer, numBytesToRender);
-			currentPos = ModBufferSize(lastPos + numBytesToRender);
+			lastPos = ModBufferSize(lastPos + numBytesToRender);
-			totalRenderedBytes += numBytesToRender;
-			lastPos = currentPos;
 		}
 		soundCriticalSection.Leave();
 		soundSyncEvent.Wait();
@@ -142,7 +140,6 @@ bool DSound::Start()
 	dsBuffer->Lock(0, bufferSize, (void* *)&p1, &num1, 0, 0, 0);
 	memset(p1, 0, num1);
 	dsBuffer->Unlock(p1, num1, 0, 0);
-	totalRenderedBytes = -bufferSize;
 	thread = new Common::Thread(soundThread, (void *)this);
 	return true;
 }

Source/Core/AudioCommon/Src/DSoundStream.h

@@ -25,8 +25,7 @@
 #include <mmsystem.h>
 #include <dsound.h>
 
-#define BUFSIZE 32768
+#define BUFSIZE (1024 * 8 * 4)
-#define MAXWAIT 70   // miliseconds
 #endif
 
 class DSound : public SoundStream
@@ -41,31 +40,30 @@ class DSound : public SoundStream
     IDirectSoundBuffer* dsBuffer;
     
     int bufferSize;     //i bytes
-    int totalRenderedBytes;
 	int m_volume;
     
     // playback position
     int currentPos;
     int lastPos;
-    short realtimeBuffer[1024 * 1024];
+    short realtimeBuffer[BUFSIZE / sizeof(short)];
     
-    inline int FIX128(int x) {
+    inline int FIX128(int x)
+	{
 		return x & (~127);
     }
 
-    inline int ModBufferSize(int x) {
+    inline int ModBufferSize(int x)
+	{
 		return (x + bufferSize) % bufferSize;
     }
 
     bool CreateBuffer();
-    bool WriteDataToBuffer(DWORD dwOffset, char* soundData,
+    bool WriteDataToBuffer(DWORD dwOffset, char* soundData, DWORD dwSoundBytes);
-			   DWORD dwSoundBytes);
 
 public:
 	DSound(CMixer *mixer, void *hWnd = NULL)
 		: SoundStream(mixer)
 		, bufferSize(0)
-		, totalRenderedBytes(0)
 		, currentPos(0)
 		, lastPos(0)
 		, dsBuffer(0)

Source/Core/AudioCommon/Src/Mixer.cpp

@@ -16,112 +16,65 @@
 // http://code.google.com/p/dolphin-emu/
 
 
-// This queue solution is temporary. I'll implement something more efficient later.
+#include "Atomic.h"
-#include <queue> // System
-
-#include "Thread.h" // Common
 
 #include "Mixer.h"
-#include "FixedSizeQueue.h"
 #include "AudioCommon.h"
 
-int CMixer::Mix(short *samples, int numSamples)
+// Executed from sound stream thread
+unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
 {
-	if (! samples) {
+	if (!samples)
-		Premix(NULL, 0);
 		return 0;
-	}
-	// silence
-	memset(samples, 0, numSamples * 2 * sizeof(short));
 
-	if (g_dspInitialize.pEmulatorState) {
+	if (g_dspInitialize.pEmulatorState)
+	{
 		if (*g_dspInitialize.pEmulatorState != 0)
-			return 0;
+		{
-	}
+			// Silence
-
+			memset(samples, 0, numSamples * 4);
-	// first get the DTK Music
+			return numSamples;
-	if (m_EnableDTKMusic) {
-		g_dspInitialize.pGetAudioStreaming(samples, numSamples);
-	}
-
-	Premix(samples, numSamples);
-	
-	int count = 0;
-
-	push_sync.Enter();
-	while (m_queueSize > queue_minlength && count < numSamples * 2) 
-	{
-		int x = samples[count];
-		x += sample_queue.front();
-		if (x > 32767) x = 32767;
-		if (x < -32767) x = -32767;
-		samples[count++] = x;
-		sample_queue.pop();
-		x = samples[count];
-		x += sample_queue.front();
-		if (x > 32767) x = 32767;
-		if (x < -32767) x = -32767;
-		samples[count++] = x;
-		sample_queue.pop();
-		m_queueSize-=2;
-	}
-	push_sync.Leave();
-
-	return count;
-}
-
-
-void CMixer::PushSamples(short *samples, int num_stereo_samples, int core_sample_rate)
-{
-	push_sync.Enter();
-	if (m_queueSize == 0)
-	{
-		m_queueSize = queue_minlength;
-		for (int i = 0; i < queue_minlength; i++)
-			sample_queue.push((s16)0);
-	}
-	push_sync.Leave();
- 
-#ifdef _WIN32
-	if (GetAsyncKeyState(VK_TAB))
-		return;
-#endif
-
-	// Write Other Audio
-	if (!m_throttle)
-		return;	
-	
-	// -----------------------------------------------------------------------	
-	// The auto throttle function. This loop will put a ceiling on the CPU MHz.
-	// ----------------------------
-	/* This is only needed for non-AX sound, currently directly
-	   streamed and DTK sound. For AX we call SoundStream::Update in
-	   AXTask() for example. */
-	while (m_queueSize > queue_maxlength / 2)
-	{
-		// Urgh.
-		if (g_dspInitialize.pEmulatorState) {
-			if (*g_dspInitialize.pEmulatorState != 0) 
-				return;
 		}
-		soundStream->Update();
-		SLEEP(1);
 	}
-	// -----------------------------------------------------------------------
 
-	push_sync.Enter();
+	unsigned int numLeft = Common::AtomicLoad(m_numSamples);
-	while (num_stereo_samples)
+	numLeft = (numLeft > numSamples) ? numSamples : numLeft;
+
+	// Do re-sampling if needed
+	if (m_sampleRate == m_dspSampleRate)
 	{
-		sample_queue.push(Common::swap16(*samples));
+		for (unsigned int i = 0; i < numLeft * 2; i++)
-		samples++;
+			samples[i] = Common::swap16(m_buffer[(m_indexR + i) & INDEX_MASK]);
-		sample_queue.push(Common::swap16(*samples));
+		m_indexR += numLeft * 2;
-		samples++;
-		m_queueSize += 2;
-		num_stereo_samples--;
 	}
-	push_sync.Leave();
+	else if (m_sampleRate < m_dspSampleRate) // If down-sampling needed
-	return;
+	{
+		_dbg_assert_msg_(DSPHLE, !(numSamples % 2), "Number of Samples: %i must be even!", numSamples);
 
+		short *pDest = samples;
+		int last_l, last_r, cur_l, cur_r;
+
+		for (unsigned int i = 0; i < numLeft * 3 / 2; i++)
+		{
+			cur_l = Common::swap16(m_buffer[(m_indexR + i * 2) & INDEX_MASK]);
+			cur_r = Common::swap16(m_buffer[(m_indexR + i * 2 + 1) & INDEX_MASK]);
+
+			if (i % 3)
+			{
+				*pDest++ = (last_l + cur_r) / 2;
+				*pDest++ = (last_r + cur_r) / 2;
+			}
+
+			last_l = cur_l;
+			last_r = cur_r;
+		}
+
+		m_indexR += numLeft * 2 * 3 / 2;
+	}
+	else if (m_sampleRate > m_dspSampleRate)
+	{
+		// AyuanX: Up-sampling is not implemented yet
+		PanicAlert("Mixer: Up-sampling is not implemented yet!");
 /*
 	static int PV1l=0,PV2l=0,PV3l=0,PV4l=0;
 	static int PV1r=0,PV2r=0,PV3r=0,PV4r=0;
@@ -183,16 +136,93 @@ void CMixer::PushSamples(short *samples, int num_stereo_samples, int core_sample
 		sample_queue.push(r);
 		m_queueSize += 2;
 	}
-	push_sync.Leave();
 */
+	}
 
+	// Padding
+	if (numSamples > numLeft)
+		memset(&samples[numLeft * 2], 0, (numSamples - numLeft) * 4);
+
+	// Add the HLE sound
+	if (m_sampleRate < m_dspSampleRate)
+	{
+		PanicAlert("Mixer: DSPHLE down-sampling is not implemented yet!\n"
+			"Usually no game should require this, please report!");
+	}
+	else
+	{
+		Premix(samples, numSamples, m_sampleRate);
+	}
+
+	// Add the DTK Music
+	if (m_EnableDTKMusic)
+	{
+		// Re-sampling is done inside
+		g_dspInitialize.pGetAudioStreaming(samples, numSamples, m_sampleRate);
+	}
+
+	Common::AtomicAdd(m_numSamples, -(int)numLeft);
+
+	return numSamples;
 }
 
-int CMixer::GetNumSamples()
+
+void CMixer::PushSamples(short *samples, unsigned int num_samples, unsigned int sample_rate)
 {
-	return m_queueSize / 2;
+	// The auto throttle function. This loop will put a ceiling on the CPU MHz.
-	//int ret = (m_queueSize - queue_minlength) / 2;
+	if (m_throttle)
-	//ret = (ret > 0) ? ret : 0;
+	{
-	//return ret;
+		// AyuanX: Remember to reserve "num_samples * 1.5" free sample space at least!
+		// Becuse we may do re-sampling later
+		while (Common::AtomicLoad(m_numSamples) >= MAX_SAMPLES - RESERVED_SAMPLES)
+		{
+			if (g_dspInitialize.pEmulatorState)
+			{
+				if (*g_dspInitialize.pEmulatorState != 0) 
+					break;
+			}
+			soundStream->Update();
+			SLEEP(1);
+		}
+	}
+
+	// Check if we have enough free space
+	if (num_samples > MAX_SAMPLES - Common::AtomicLoad(m_numSamples))
+		return;
+
+	// AyuanX: Actual re-sampling work has been moved to sound thread
+	// to alleviates the workload on main thread
+	// and we simply store raw data here to make fast mem copy
+	int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (m_indexW & INDEX_MASK)) * sizeof(short);
+	if (over_bytes > 0)
+	{
+		memcpy(&m_buffer[m_indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
+		memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
+	}
+	else
+	{
+		memcpy(&m_buffer[m_indexW & INDEX_MASK], samples, num_samples * 4);
+	}
+
+	m_indexW += num_samples * 2;
+
+	if (m_sampleRate < m_dspSampleRate)
+	{
+		// This is kind of tricky :P  
+		num_samples = num_samples * 2 / 3;
+	}
+	else if (m_sampleRate > m_dspSampleRate)
+	{
+		PanicAlert("Mixer: Up-sampling is not implemented yet!");
+	}
+
+	Common::AtomicAdd(m_numSamples, num_samples);
+
+	return;
+}
+
+unsigned int CMixer::GetNumSamples()
+{
+	return Common::AtomicLoad(m_numSamples);
 }
 

Source/Core/AudioCommon/Src/Mixer.h

@@ -18,39 +18,38 @@
 #ifndef _MIXER_H_
 #define _MIXER_H_
 
-#include "FixedSizeQueue.h"
+// 16 bit Stereo
-#include "Thread.h"
+#define MAX_SAMPLES			(1024 * 4)
-
+#define INDEX_MASK			(MAX_SAMPLES * 2 - 1)
-// On real hardware, this fifo is much, much smaller. But timing is also
+#define RESERVED_SAMPLES	(MAX_SAMPLES / 8)
-// tighter than under Windows, so...
-#define queue_minlength  1024 * 4
-#define queue_maxlength  1024 * 28
 
 class CMixer {
 	
 public:
-	// AyuanX: Mixer sample rate is fixed to 32khz for now
+	CMixer(unsigned int AISampleRate = 48000, unsigned int DSPSampleRate = 48000)
-	// if any game sets DSP sample rate to 48khz, we are doomed
+		: m_aiSampleRate(AISampleRate)
-	// TODO: Fix this somehow!
+		, m_dspSampleRate(DSPSampleRate)
-	CMixer(unsigned int SampleRate = 32000)
-		: m_sampleRate(SampleRate)
 		, m_bits(16)
 		, m_channels(2)
-		, m_mode(2)
 		, m_HLEready(false)
-		, m_queueSize(0)
+		, m_numSamples(0)
-	{}
+		, m_indexW(0)
+		, m_indexR(0)
+	{
+		// AyuanX: When sample rate differs, we have to do re-sampling
+		// I perfer speed so let's do down-sampling instead of up-sampling
+		// If you like better sound than speed, feel free to implement the up-sampling code
+		m_sampleRate = (m_aiSampleRate < m_dspSampleRate) ? m_aiSampleRate : m_dspSampleRate;
+	}
 
 	// Called from audio threads
-	virtual int Mix(short *sample, int numSamples);
+	virtual unsigned int Mix(short* samples, unsigned int numSamples);
-	virtual int GetNumSamples();
+	virtual void Premix(short *samples, unsigned int numSamples, unsigned int sampleRate) {}
+	unsigned int GetNumSamples();
 
 	// Called from main thread
-	virtual void PushSamples(short* samples, int num_stereo_samples, int core_sample_rate);
+	virtual void PushSamples(short* samples, unsigned int num_samples, unsigned int sample_rate);
-	
+	unsigned int GetSampleRate() {return m_sampleRate;}
-	virtual void Premix(short *samples, int numSamples) {}
-
-	int GetSampleRate() {return m_sampleRate;}
 
 	void SetThrottle(bool use) { m_throttle = use;}
 	void SetDTKMusic(bool use) { m_EnableDTKMusic = use;}
@@ -61,19 +60,23 @@ public:
 	// ---------------------
 
 protected:
-	int m_sampleRate;
+	unsigned int m_sampleRate;
+	unsigned int m_aiSampleRate;
+	unsigned int m_dspSampleRate;
 	int m_bits;
 	int m_channels;
 	
-	int m_mode;
 	bool m_HLEready;
-	int m_queueSize;
 
 	bool m_EnableDTKMusic;
 	bool m_throttle;
+
+	short m_buffer[MAX_SAMPLES * 2];
+	u32 m_indexW;
+	u32 m_indexR;
+	volatile u32 m_numSamples;
+
 private:
-	Common::CriticalSection push_sync;
-	FixedSizeQueue<s16, queue_maxlength> sample_queue;
 
 };
 

Source/Core/AudioCommon/Src/NullSoundStream.h

@@ -22,10 +22,10 @@
 #include "Mixer.h"
 
 class NullMixer : public CMixer {
+
 public:
-	virtual int Mix(short *sample, int numSamples) {return 0;}
+	virtual unsigned int Mix(short *samples, unsigned int numSamples) { return 0; }
-	virtual void PushSamples(short* samples, int num_stereo_samples, 
+	virtual void PushSamples(short* samples, unsigned int num_samples, unsigned int sample_rate) {}
-							 int core_sample_rate) {}
 };
 
 class NullSound : public SoundStream
@@ -35,7 +35,6 @@ public:
 	{
 		delete m_mixer;
 		m_mixer = new NullMixer();
-	   
 	}
     
     virtual ~NullSound() {}
@@ -47,7 +46,7 @@ public:
 	virtual bool Start() { return true; }
 
 	virtual void Update() { 
-		m_mixer->Mix(NULL, 256 >> 2);
+		//m_mixer->Mix(NULL, 256 >> 2);
 		//(*callback)(NULL, 256 >> 2, 16, sampleRate, 2); 
 	}
 };

Source/Core/AudioCommon/Src/OpenALStream.cpp

@@ -138,12 +138,13 @@ void OpenALStream::SoundLoop()
 	// Generate a Source to playback the Buffers
 	alGenSources(1, &uiSource);
 
-	memset(realtimeBuffer, 0, OAL_BUFFER_SIZE);
+	// Short Silence
+	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
 	for (int i = 0; i < OAL_NUM_BUFFERS; i++)
-		alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, OAL_BUFFER_SIZE, ulFrequency);
+		alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, OAL_MAX_SAMPLES, ulFrequency);
-
 	alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, uiBuffers);
 	alSourcePlay(uiSource);
+
 	err = alGetError();
 	// TODO: Error handling
 
@@ -158,12 +159,12 @@ void OpenALStream::SoundLoop()
 			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
 			iBuffersFilled = 0;
 		}
-		int numSamples = m_mixer->GetNumSamples();
-		numSamples &= ~0x100;
 
-		if (iBuffersProcessed && numSamples)
+		unsigned int numSamples = m_mixer->GetNumSamples();
+
+		if (iBuffersProcessed && (numSamples >= OAL_THRESHOLD))
 		{
-			numSamples = (numSamples > OAL_BUFFER_SIZE / 4) ? OAL_BUFFER_SIZE / 4 : numSamples;
+			numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
 			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
 			if (iBuffersFilled == 0)
 				alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
@@ -176,11 +177,11 @@ void OpenALStream::SoundLoop()
 			if (iBuffersFilled == OAL_NUM_BUFFERS)
 				alSourcePlay(uiSource);
 		}
-		else
+		else if (numSamples >= OAL_THRESHOLD)
 		{
 			ALint state = 0;
 			alGetSourcei(uiSource, AL_SOURCE_STATE, &state);
-			if (state != AL_PLAYING)
+			if (state == AL_STOPPED)
 				alSourcePlay(uiSource);
 		}
 		soundSyncEvent.Wait();

Source/Core/AudioCommon/Src/OpenALStream.h

@@ -33,10 +33,11 @@
 #include "AL/al.h"
 #include "AL/alc.h"
 #endif // WIN32
-// public use
+// 16 bit Stereo
 #define SFX_MAX_SOURCE		1
 #define OAL_NUM_BUFFERS		8
-#define OAL_BUFFER_SIZE		(512 * 4)
+#define OAL_MAX_SAMPLES		512		// AyuanX: Don't make it too large, as larger buffer means longer delay
+#define OAL_THRESHOLD		128
 #endif
 
 class OpenALStream: public SoundStream
@@ -66,7 +67,7 @@ private:
 	Common::CriticalSection soundCriticalSection;
 	Common::Event soundSyncEvent;
 	
-	short realtimeBuffer[OAL_BUFFER_SIZE/sizeof(short)];
+	short realtimeBuffer[OAL_MAX_SAMPLES * 2];
 	ALuint uiBuffers[OAL_NUM_BUFFERS];
 	ALuint uiSource;
 	ALfloat fVolume;

Source/Core/Common/Src/PluginDSP.h

@@ -27,7 +27,7 @@ typedef void (__cdecl* TDSP_WriteMailBox)(bool _CPUMailbox, unsigned short);
 typedef unsigned short (__cdecl* TDSP_ReadMailBox)(bool _CPUMailbox);
 typedef unsigned short (__cdecl* TDSP_ReadControlRegister)();
 typedef unsigned short (__cdecl* TDSP_WriteControlRegister)(unsigned short);
-typedef void (__cdecl *TDSP_SendAIBuffer)(unsigned int address, int sample_rate);
+typedef void (__cdecl *TDSP_SendAIBuffer)(unsigned int address, unsigned int num_samples, unsigned int sample_rate);
 typedef void (__cdecl *TDSP_Update)(int cycles);
 typedef void (__cdecl *TDSP_StopSoundStream)();
 typedef void (__cdecl *TDSP_ClearAudioBuffer)();

Source/Core/Core/Src/ConfigManager.cpp

@@ -234,7 +234,7 @@ void SConfig::LoadSettings()
 		ini.Get("Core", "RunCompareServer",	&m_LocalCoreStartupParameter.bRunCompareServer,	false);
 		ini.Get("Core", "RunCompareClient",	&m_LocalCoreStartupParameter.bRunCompareClient,	false);
 		ini.Get("Core", "TLBHack",			&m_LocalCoreStartupParameter.iTLBHack,			0);
-		ini.Get("Core", "FrameLimit",		&m_Framelimit,									1);
+		ini.Get("Core", "FrameLimit",		&m_Framelimit,									0); // auto frame limit by default
 
 		// Plugins
 		ini.Get("Core", "GFXPlugin",  &m_LocalCoreStartupParameter.m_strVideoPlugin,	m_DefaultGFXPlugin.c_str());

Source/Core/Core/Src/Core.cpp

@@ -386,6 +386,7 @@ THREAD_RETURN EmuThread(void *pArg)
 	dspInit.pDebuggerBreak			= Callback_DebuggerBreak;
 	dspInit.pGenerateDSPInterrupt	= Callback_DSPInterrupt;
 	dspInit.pGetAudioStreaming		= AudioInterface::Callback_GetStreaming;
+	dspInit.pGetSampleRate			= AudioInterface::Callback_GetSampleRate;
 	dspInit.pEmulatorState			= (int *)PowerPC::GetStatePtr();
 	dspInit.bWii					= _CoreParameter.bWii;
 	dspInit.bOnThread				= _CoreParameter.bDSPThread;

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

@@ -54,13 +54,13 @@ union AICR
 	struct 
 	{
 		unsigned PSTAT		: 1;  // sample counter/playback enable
-		unsigned AFR		: 1;  // 0=32khz 1=48khz
+		unsigned AFR		: 1;  // AI Frequency (0=32khz 1=48khz)
 		unsigned AIINTMSK	: 1;  // 0=interrupt masked 1=interrupt enabled
 		unsigned AIINT		: 1;  // audio interrupt status
 		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=48khz 1=32khz) WTF, who designed this?
+        unsigned DSPFR  	: 1;  // DSP Frequency (0=48khz 1=32khz)
 		unsigned			:25;
 	};
 	u32 hex;
@@ -90,8 +90,8 @@ struct SAudioRegister
 // STATE_TO_SAVE
 static SAudioRegister g_AudioRegister;	
 static u64 g_LastCPUTime = 0;
-static int g_SampleRate = 32000;
+static unsigned int g_SampleRate = 32000;
-static int g_DSPSampleRate = 32000;
+static unsigned int g_DSPSampleRate = 32000;
 static u64 g_CPUCyclesPerSample = 0xFFFFFFFFFFFULL;
 
 void DoState(PointerWrap &p)
@@ -264,9 +264,15 @@ void GenerateAudioInterrupt()
 	UpdateInterrupts();
 }
 
+void Callback_GetSampleRate(unsigned int &_AISampleRate, unsigned int &_DSPSampleRate)
+{
+	_AISampleRate = g_SampleRate;
+	_DSPSampleRate = g_DSPSampleRate;
+}
+
 // Callback for the disc streaming
 // WARNING - called from audio thread
-u32 Callback_GetStreaming(short* _pDestBuffer, u32 _numSamples)
+unsigned int Callback_GetStreaming(short* _pDestBuffer, unsigned int _numSamples, unsigned int _sampleRate)
 {
 	if (g_AudioRegister.m_Control.PSTAT && !CCPU::IsStepping())
 	{		
@@ -275,34 +281,60 @@ 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)
+		if (g_SampleRate == 48000 && _sampleRate == 32000)
 		{
 			_dbg_assert_msg_(AUDIO_INTERFACE, !(_numSamples % 2), "Number of Samples: %i must be even!", _numSamples);
 			_numSamples = _numSamples * 3 / 2;
 		}
+		else if (g_SampleRate == 32000 && _sampleRate == 48000)
+		{
+			// AyuanX: Up-sampling is not implemented yet
+			PanicAlert("AUDIO_INTERFACE: Up-sampling is not implemented yet!");
+		}
 
-		short pcm_l = 0;
+		int pcm_l, pcm_r;
-		short pcm_r = 0;
 		for (unsigned int i = 0; i < _numSamples; i++)
 		{
 			if (pos == 0)
 				ReadStreamBlock(pcm);
 
-			if (g_SampleRate == 48000)
+			if (g_SampleRate == 48000 && _sampleRate == 32000)
 			{
 				if (i % 3)
 				{
-					*_pDestBuffer++ = ((pcm_l / 2 + pcm[pos*2] / 2) * lvolume) >> 8;
+					pcm_l = (((pcm_l + (int)pcm[pos*2]) / 2  * lvolume) >> 8) + (int)(*_pDestBuffer);
-					*_pDestBuffer++ = ((pcm_r / 2 + pcm[pos*2+1] / 2) * rvolume) >> 8;
+					if (pcm_l > 32767)
+						pcm_l = 32767;
+					else if (pcm_l < -32767)
+						pcm_l = -32767;
+					*_pDestBuffer++ = pcm_l;
+
+					pcm_r = (((pcm_r + (int)pcm[pos*2+1]) / 2 * rvolume) >> 8) + (int)(*_pDestBuffer);
+ 					if (pcm_r > 32767)
+						pcm_r = 32767;
+					else if (pcm_r < -32767)
+						pcm_r = -32767;
+					*_pDestBuffer++ = pcm_r;
 				}
 				pcm_l = pcm[pos*2];
 				pcm_r = pcm[pos*2+1];
 			}
 			else
 			{
-				*_pDestBuffer++ = (pcm[pos*2] * lvolume) >> 8;
+				pcm_l = (((int)pcm[pos*2] * lvolume) >> 8) + (int)(*_pDestBuffer);
-				*_pDestBuffer++ = (pcm[pos*2+1] * rvolume) >> 8;
+				if (pcm_l > 32767)
+					pcm_l = 32767;
+				else if (pcm_l < -32767)
+					pcm_l = -32767;
+				*_pDestBuffer++ = pcm_l;
+
+				pcm_r = (((int)pcm[pos*2+1] * rvolume) >> 8) + (int)(*_pDestBuffer);
+				if (pcm_r > 32767)
+					pcm_r = 32767;
+				else if (pcm_r < -32767)
+					pcm_r = -32767;
+				*_pDestBuffer++ = pcm_r;
 			}
 
 			if (++pos == 28) 
@@ -311,7 +343,7 @@ u32 Callback_GetStreaming(short* _pDestBuffer, u32 _numSamples)
 	}
 	else
 	{
-		// AyuanX: We have already preset those bytes, no need to do this again
+		// Don't overwrite existed sample data
 		/*
 		for (unsigned int i = 0; i < _numSamples * 2; i++)
 		{
@@ -361,12 +393,12 @@ void IncreaseSampleCount(const u32 _iAmount)
 	}
 }
 
-u32 GetAISampleRate()
+unsigned int GetAISampleRate()
 {
 	return g_SampleRate;
 }
 
-u32 GetDSPSampleRate()
+unsigned int GetDSPSampleRate()
 {
 	return g_DSPSampleRate;
 }

Source/Core/Core/Src/HW/AudioInterface.h

@@ -34,14 +34,15 @@ void DoState(PointerWrap &p);
 void Update();
 
 // Called by DSP plugin
-u32 Callback_GetStreaming(short* _pDestBuffer, u32 _numSamples);
+void Callback_GetSampleRate(unsigned int &_AISampleRate, unsigned int &_DSPSampleRate);
+unsigned int Callback_GetStreaming(short* _pDestBuffer, unsigned int _numSamples, unsigned int _sampleRate = 48000);
 
 void Read32(u32& _uReturnValue, const u32 _iAddress);
 void Write32(const u32 _iValue, const u32 _iAddress);
 
 // Get the audio rates (48000 or 32000 only)
-u32 GetAISampleRate();
+unsigned int GetAISampleRate();
-u32 GetDSPSampleRate();
+unsigned int GetDSPSampleRate();
 
 }  // namespace
 

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

@@ -197,7 +197,6 @@ static u16 g_AR_SIZE;
 static u16 g_AR_MODE;
 static u16 g_AR_REFRESH;
 
-
 Common::PluginDSP *dsp_plugin;
 
 
@@ -379,27 +378,6 @@ 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;
@@ -503,12 +481,17 @@ void UpdateAudioDMA()
 		// external audio fifo in the emulator, to be mixed with the disc
 		// streaming output. If that audio queue fills up, we delay the
 		// emulator.
-		dsp_plugin->DSP_SendAIBuffer(g_audioDMA.ReadAddress, AudioInterface::GetDSPSampleRate());
+
-		g_audioDMA.ReadAddress += 32;
+		// AyuanX: let's do it in a bundle to speed up
+		if (g_audioDMA.BlocksLeft == g_audioDMA.AudioDMAControl.NumBlocks)
+			dsp_plugin->DSP_SendAIBuffer(g_audioDMA.SourceAddress, g_audioDMA.AudioDMAControl.NumBlocks * 8, AudioInterface::GetDSPSampleRate());
+
+//		g_audioDMA.ReadAddress += 32;
 		g_audioDMA.BlocksLeft--;
+
 		if (g_audioDMA.BlocksLeft == 0)
 		{
-			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);

Source/Core/Core/Src/HW/MemmapFunctions.cpp

@@ -590,7 +590,7 @@ void GenerateISIException()
 	//	segment (N bit set in segment descriptor), or to guarded memory
 	//	when MSR[IR] = 1. Otherwise, cleared.
 	// Bit 4: Set if a memory access is not permitted by the page or IBAT protection
-	//	mechanism, described in Chapter 7, “Memory Management”; otherwise cleared.
+	//	mechanism, described in Chapter 7, "Memory Management" otherwise cleared.
 	// Only one of 1,3, or 4 may be set at a time
 
 	// For now let's just say that hash lookup failed

Source/PluginSpecs/pluginspecs_dsp.h

@@ -16,7 +16,8 @@ typedef void            (*TLogv)(const char* _szMessage, int _v);
 typedef const char*     (*TName)(void);
 typedef void            (*TDebuggerBreak)(void);
 typedef void            (*TGenerateDSPInt)(void);
-typedef unsigned int    (*TAudioGetStreaming)(short* _pDestBuffer, unsigned int _numSamples);
+typedef unsigned int    (*TAudioGetStreaming)(short* _pDestBuffer, unsigned int _numSamples, unsigned int _sampleRate);
+typedef void			(*TGetSampleRate)(unsigned int &AISampleRate, unsigned int &DSPSampleRate);
 
 typedef struct
 {
@@ -30,6 +31,7 @@ typedef struct
 	TDebuggerBreak			pDebuggerBreak;
 	TGenerateDSPInt			pGenerateDSPInterrupt;
 	TAudioGetStreaming		pGetAudioStreaming;
+	TGetSampleRate			pGetSampleRate;
 	int                    *pEmulatorState;
 	bool					bWii;
 	bool                    bOnThread;
@@ -95,9 +97,10 @@ EXPORT void CALL DSP_Update(int cycles);
 // Function: DSP_SendAIBuffer
 // Purpose:  This function sends the current AI Buffer to the DSP plugin
 // input:    _Address : Memory-Address
-// input:    _Size : Size of the Buffer (always 32)
+// input:    _Number  : Number of the Samples
+// input:    _Rate    : Sample Rate 32000/48000
 // 
-EXPORT void CALL DSP_SendAIBuffer(unsigned int address, int sample_rate);
+EXPORT void CALL DSP_SendAIBuffer(unsigned int address, unsigned int num_samples, unsigned int sample_rate);
 
 // __________________________________________________________________________________________________
 // Function: DSP_StopSoundStream

Source/Plugins/Plugin_DSP_HLE/Src/DSPHandler.h

@@ -19,6 +19,7 @@
 #define _DSPHANDLER_H
 
 #include "Common.h"
+#include "AudioCommon.h"
 #include "MailHandler.h"
 #include "UCodes/UCodes.h"
 
@@ -57,34 +58,6 @@ private:
 	CDSPHandler();
 	~CDSPHandler();
 
-	// UDSPControl
-	union UDSPControl
-	{
-		u16 Hex;
-		struct
-		{
-			unsigned DSPReset       : 1; // Write 1 to reset and waits for 0
-			unsigned DSPAssertInt   : 1;
-			unsigned DSPHalt        : 1;
-
-			unsigned AI             : 1;
-			unsigned AI_mask        : 1;
-			unsigned ARAM           : 1;
-			unsigned ARAM_mask      : 1;
-			unsigned DSP            : 1;
-			unsigned DSP_mask       : 1;
-
-			unsigned ARAM_DMAState  : 1; // DSPGetDMAStatus() uses this flag
-			unsigned DSPInitCode    : 1;
-			unsigned DSPInit        : 1; // DSPInit() writes to this flag
-			unsigned pad            : 4;
-		};
-
-		UDSPControl(u16 _Hex = 0)
-			: Hex(_Hex)
-		{}
-	};
-
 	// singleton instance
 	static CDSPHandler* m_pInstance;
 

Source/Plugins/Plugin_DSP_HLE/Src/HLEMixer.cpp

@@ -20,23 +20,14 @@
 #include "DSPHandler.h"
 #include "HLEMixer.h"
 
-void HLEMixer::MixUCode(short *samples, int numSamples) {
+void HLEMixer::Premix(short *samples, unsigned int numSamples, unsigned int sampleRate)
-	// if this was called directly from the HLE, and not by timeout
+{
-	if (g_Config.m_EnableHLEAudio && IsHLEReady()) {
+	// if this was called directly from the HLE
+	if (g_Config.m_EnableHLEAudio && IsHLEReady())
+	{
 		IUCode *pUCode = CDSPHandler::GetInstance().GetUCode();
 		if (pUCode && samples)
 			pUCode->MixAdd(samples, numSamples);
 	}
 }
 
-void HLEMixer::Premix(short *samples, int numSamples) {
-	
-	// first get the DTK Music
-	//	if (g_Config.m_EnableDTKMusic) {
-	//		g_dspInitialize.pGetAudioStreaming(samples, numSamples);
-	//	}
-
-	MixUCode(samples, numSamples);
-}
-
-

Source/Plugins/Plugin_DSP_HLE/Src/HLEMixer.h

@@ -6,9 +6,10 @@
 class HLEMixer : public CMixer 
 {
 public:
-	void MixUCode(short *samples, int numSamples);
+	HLEMixer(unsigned int AISampleRate = 48000, unsigned int DSPSampleRate = 48000)
+		: CMixer(AISampleRate, DSPSampleRate) {};
 	
-	virtual void Premix(short *samples, int numSamples);	
+	virtual void Premix(short *samples, unsigned int numSamples, unsigned int sampleRate);	
 };
 
 #endif // HLEMIXER_H

Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AX.cpp

@@ -498,7 +498,7 @@ bool CUCode_AX::AXTask(u32& _uMail)
 		    SaveLog("%08x : AXLIST PB address: %08x", uAddress, m_addressPBs);
 
 		    SaveLog("Update the SoundThread to be in sync");
-            soundStream->Update(); //do it in this thread to avoid sync problems
+//          soundStream->Update(); //do it in this thread to avoid sync problems
 
 		    }
 		    break;

Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AXStructs.h

@@ -156,7 +156,7 @@ struct AXParamBlock
 /* 63 */	PBADPCMInfo adpcm;
 /* 83 */    PBSampleRateConverter src;
 /* 90 */	PBADPCMLoopInfo adpcm_loop_info;
-/* 93 */	u16 unknown_maybe_padding[3];
+/* 93 */	//u16 unknown_maybe_padding[3];		// Comment this out to get some speedup
 };
 
 struct PBLpf
@@ -201,7 +201,7 @@ struct AXParamBlockWii
 /*  95 */	PBADPCMLoopInfo adpcm_loop_info;
 /*  98 */	PBLpf lpf;
 /* 102 */	PBHpf hpf;
-/* 106 */	u16 pad[22];
+/* 106 */	//u16 pad[22];	// Comment this out to get some speedup
 };
 
 struct AXParamBlockWii_ // new CRC version
@@ -226,7 +226,7 @@ struct AXParamBlockWii_ // new CRC version
 /* 0x0B4 */	PBADPCMLoopInfo adpcm_loop_info;
 /* 0x0BA */	PBLpf lpf;
 /* 0x0C2 */	PBHpf hpf;
-/* 0x0CA */	u16 pad[27];
+/* 0x0CA */	//u16 pad[27];	// Comment this out to get some speedup
 /* 0x100 */
 };
 

Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AXWii.cpp

@@ -289,7 +289,7 @@ bool CUCode_AXWii::AXTask(u32& _uMail)
 			m_addressPBs = Memory_Read_U32(uAddress);
 			lCUCode_AX->m_addressPBs = m_addressPBs; // for the sake of logging
 			soundStream->GetMixer()->SetHLEReady(true);
-			soundStream->Update();
+//			soundStream->Update();
 			uAddress += 4;
 		    break;
 

Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_Zelda.cpp

@@ -147,7 +147,7 @@ void CUCode_Zelda::HandleMail_LightVersion(u32 _uMail)
 
 			soundStream->GetMixer()->SetHLEReady(true);
 			DEBUG_LOG(DSPHLE, "Update the SoundThread to be in sync");
-			soundStream->Update(); //do it in this thread to avoid sync problems
+//			soundStream->Update(); //do it in this thread to avoid sync problems
 
 			m_bSyncCmdPending = false;
 		}
@@ -216,7 +216,7 @@ void CUCode_Zelda::HandleMail_SMSVersion(u32 _uMail)
 
 					soundStream->GetMixer()->SetHLEReady(true);
 					DEBUG_LOG(DSPHLE, "Update the SoundThread to be in sync");
-					soundStream->Update(); //do it in this thread to avoid sync problems
+//					soundStream->Update(); //do it in this thread to avoid sync problems
 
 					m_bSyncCmdPending = false;
 				}
@@ -339,7 +339,7 @@ void CUCode_Zelda::HandleMail_NormalVersion(u32 _uMail)
 
 					soundStream->GetMixer()->SetHLEReady(true);
 					DEBUG_LOG(DSPHLE, "Update the SoundThread to be in sync");
-					soundStream->Update(); //do it in this thread to avoid sync problems
+//					soundStream->Update(); //do it in this thread to avoid sync problems
 
 					m_bSyncCmdPending = false;
 				}

Source/Plugins/Plugin_DSP_HLE/Src/main.cpp

@@ -33,7 +33,6 @@ DSPDebuggerHLE* m_DebuggerFrame = NULL;
 #include "Config.h"
 #include "Setup.h"
 #include "StringUtil.h"
-#include "AudioCommon.h"
 #include "LogManager.h"
 
 
@@ -42,8 +41,8 @@ PLUGIN_GLOBALS* globals = NULL;
 DSPInitialize g_dspInitialize;
 u8* g_pMemory;
 extern std::vector<std::string> sMailLog, sMailTime;
-bool g_bMuted = false;
 
+bool g_InitMixer = false;
 SoundStream *soundStream = NULL;
 
 // Mailbox utility
@@ -203,6 +202,7 @@ void DllConfig(HWND _hParent)
 
 void Initialize(void *init)
 {
+	g_InitMixer = false;
 	g_dspInitialize = *(DSPInitialize*)init;
 
 	g_Config.Load();
@@ -211,10 +211,6 @@ void Initialize(void *init)
 	g_dspState.Reset();
 
 	CDSPHandler::CreateInstance();
-
-	soundStream = AudioCommon::InitSoundStream(new HLEMixer()); 
-        if(!soundStream)
-            PanicAlert("Error starting up sound stream");
 }
 
 void DSP_StopSoundStream()
@@ -245,6 +241,7 @@ void Shutdown()
 void DoState(unsigned char **ptr, int mode)
 {
 	PointerWrap p(ptr, mode);
+	p.Do(g_InitMixer);
 	CDSPHandler::GetInstance().GetUCode()->DoState(p);
 }
 
@@ -305,6 +302,19 @@ void DSP_WriteMailboxLow(bool _CPUMailbox, unsigned short _Value)
 // Other DSP fuctions
 unsigned short DSP_WriteControlRegister(unsigned short _Value)
 {
+	UDSPControl Temp(_Value);
+	if (!g_InitMixer)
+	{
+		if (!Temp.DSPHalt && Temp.DSPInit)
+		{
+			unsigned int AISampleRate, DSPSampleRate;
+			g_dspInitialize.pGetSampleRate(AISampleRate, DSPSampleRate);
+			soundStream = AudioCommon::InitSoundStream(new HLEMixer(AISampleRate, DSPSampleRate)); 
+			if(!soundStream) PanicAlert("Error starting up sound stream");
+			// Mixer is initialized
+			g_InitMixer = true;
+		}
+	}
 	return CDSPHandler::GetInstance().WriteControlRegister(_Value);
 }
 
@@ -324,44 +334,26 @@ void DSP_Update(int cycles)
    inside Mixer_PushSamples(), the reason that we don't disable this entire
    function when Other Audio is disabled is that then we can't turn it back on
    again once the game has started. */
-void DSP_SendAIBuffer(unsigned int address, int sample_rate)
+void DSP_SendAIBuffer(unsigned int address, unsigned int num_samples, unsigned int sample_rate)
 {
-	// TODO: This is not yet fully threadsafe.
+	if (!soundStream)
-	if (!soundStream) {
 		return;
-	}
 
 	CMixer* pMixer = soundStream->GetMixer();
+
 	if (pMixer && address)
 	{
 		short* samples = (short*)Memory_Get_Pointer(address);
-/*
+		// sample_rate could be 32khz/48khz here,
-		short samples[16] = {0};  // interleaved stereo
-		if (address)
-		{
-			for (int i = 0; i < 16; i++)
-			{
-				samples[i] = Memory_Read_U16(address + i * 2);
-			}
-
-			// FIXME: Write the audio to a file
-			//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);
+		pMixer->PushSamples(samples, num_samples, sample_rate);
+
+		// FIXME: Write the audio to a file
+		//if (log_ai)
+		//	g_wave_writer.AddStereoSamples(samples, 8);
 	}
 
-	// SoundStream is updated only when necessary (there is no 70 ms limit
+	soundStream->Update();
-	// so each sample now triggers the sound stream)
-	
-	// TODO: think about this.
-	static int counter = 0;
-	counter++;
-	if ((counter & 31) == 0 && soundStream)
-		soundStream->Update();
 }
 
 void DSP_ClearAudioBuffer()

Source/Plugins/Plugin_DSP_LLE/Src/Globals.cpp

@@ -40,6 +40,11 @@ u32 Memory_Read_U32(u32 _uAddress)
 	return Common::swap32(*(u32*)&g_dsp.cpu_ram[_uAddress & RAM_MASK]);
 }
 
+void* Memory_Get_Pointer(u32 _uAddress)
+{
+	return &g_dsp.cpu_ram[_uAddress & RAM_MASK];
+}
+
 #if PROFILE
 
 #define PROFILE_MAP_SIZE 0x10000

Source/Plugins/Plugin_DSP_LLE/Src/Globals.h

@@ -26,6 +26,7 @@
 
 u16 Memory_Read_U16(u32 _uAddress); // For PB address detection
 u32 Memory_Read_U32(u32 _uAddress);
+void* Memory_Get_Pointer(u32 _uAddress);
 
 #if PROFILE	
 	void ProfilerDump(u64 _count);

Source/Plugins/Plugin_DSP_LLE/Src/main.cpp

@@ -47,6 +47,7 @@ PLUGIN_GLOBALS* globals = NULL;
 DSPInitialize g_dspInitialize;
 Common::Thread *g_hDSPThread = NULL;
 SoundStream *soundStream = NULL;
+bool g_InitMixer = false;
 
 bool bIsRunning = false;
 
@@ -166,6 +167,7 @@ void DllConfig(HWND _hParent)
 void DoState(unsigned char **ptr, int mode)
 {
 	PointerWrap p(ptr, mode);
+	p.Do(g_InitMixer);
 }
 
 void DllDebugger(HWND _hParent, bool Show)
@@ -202,6 +204,7 @@ void DSP_DebugBreak()
 
 void Initialize(void *init)
 {
+	g_InitMixer = false;
     bool bCanWork = true;
     g_dspInitialize = *(DSPInitialize*)init;
 
@@ -229,7 +232,6 @@ void Initialize(void *init)
 	{
 		g_hDSPThread = new Common::Thread(dsp_thread, NULL);
 	}
-	soundStream = AudioCommon::InitSoundStream();
 
 #if defined(HAVE_WX) && HAVE_WX
 	if (m_DebuggerFrame)
@@ -256,6 +258,19 @@ void Shutdown()
 
 u16 DSP_WriteControlRegister(u16 _uFlag)
 {
+	UDSPControl Temp(_uFlag);
+	if (!g_InitMixer)
+	{
+		if (!Temp.DSPHalt && Temp.DSPInit)
+		{
+			unsigned int AISampleRate, DSPSampleRate;
+			g_dspInitialize.pGetSampleRate(AISampleRate, DSPSampleRate);
+			soundStream = AudioCommon::InitSoundStream(new CMixer(AISampleRate, DSPSampleRate)); 
+			if(!soundStream) PanicAlert("Error starting up sound stream");
+			// Mixer is initialized
+			g_InitMixer = true;
+		}
+	}
 	DSPInterpreter::WriteCR(_uFlag);
 	return DSPInterpreter::ReadCR();
 }
@@ -343,29 +358,21 @@ void DSP_Update(int cycles)
 	}
 }
 
-void DSP_SendAIBuffer(unsigned int address, int sample_rate)
+void DSP_SendAIBuffer(unsigned int address, unsigned int num_samples, unsigned int sample_rate)
 {
-	if (soundStream->GetMixer())
+	if (!soundStream)
+		return;
+
+	CMixer* pMixer = soundStream->GetMixer();
+
+	if (pMixer && address)
 	{
-		short samples[16] = {0};  // interleaved stereo
+		short* samples = (short*)Memory_Get_Pointer(address);
-		if (address)
+
-		{
+		pMixer->PushSamples(samples, num_samples, sample_rate);
-			for (int i = 0; i < 16; i++)
-			{
-				samples[i] = Memory_Read_U16(address + i * 2);
-			}
-		}
-		soundStream->GetMixer()->PushSamples(samples, 32 / 4, sample_rate);
 	}
 
-	// SoundStream is updated only when necessary (there is no 70 ms limit
+	soundStream->Update();
-	// so each sample now triggers the sound stream)
-	
-	// TODO: think about this.
-	// static int counter = 0;
-	// counter++;
-	if (/*(counter & 31) == 0 &&*/ soundStream)
-		soundStream->Update();
 }
 
 void DSP_ClearAudioBuffer()