// Copyright (C) 2003-2008 Dolphin Project.

// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0.

// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
// GNU General Public License 2.0 for more details.

// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/

// Official SVN repository and contact information can be found at
// http://code.google.com/p/dolphin-emu/

#include "../Debugger/Debugger.h"
#include "../Logging/Console.h" // for aprintf

#ifdef _WIN32
#include "../PCHW/DSoundStream.h"
#endif
#include "../PCHW/Mixer.h"
#include "../MailHandler.h"

#include "UCodes.h"
#include "UCode_AXStructs.h"
#include "UCode_AX.h"


// ---------------------------------------------------------------------------------------
// Externals
// -----------
extern float ratioFactor;
extern u32 gLastBlock;
bool gSSBM = true; // used externally
bool gSSBMremedy1 = true; // used externally
bool gSSBMremedy2 = true; // used externally
bool gSequenced = true; // used externally
bool gVolume= true; // used externally
bool gReset = false; // used externally
extern CDebugger* m_frame;
// -----------


CUCode_AX::CUCode_AX(CMailHandler& _rMailHandler, bool wii)
	: IUCode(_rMailHandler)
	, m_addressPBs(0xFFFFFFFF)
	, wii_mode(wii)
{
	// we got loaded
	m_rMailHandler.PushMail(0xDCD10000);
	m_rMailHandler.PushMail(0x80000000);  // handshake ??? only (crc == 0xe2136399) needs it ...

	templbuffer = new int[1024 * 1024];
	temprbuffer = new int[1024 * 1024];
}

CUCode_AX::~CUCode_AX()
{
	m_rMailHandler.Clear();
	delete [] templbuffer;
	delete [] temprbuffer;
}

void CUCode_AX::HandleMail(u32 _uMail)
{
	if ((_uMail & 0xFFFF0000) == MAIL_AX_ALIST)
	{
		// a new List
	}
	else
	{
		AXTask(_uMail);
	}
}

s16 CUCode_AX::ADPCM_Step(AXParamBlock& pb, u32& samplePos, u32 newSamplePos, u16 frac)
{
	PBADPCMInfo &adpcm = pb.adpcm;

	while (samplePos < newSamplePos)
	{
		if ((samplePos & 15) == 0)
		{
			adpcm.pred_scale = g_dspInitialize.pARAM_Read_U8((samplePos & ~15) >> 1);
			samplePos    += 2;
			newSamplePos += 2;
		}

		int scale = 1 << (adpcm.pred_scale & 0xF);
		int coef_idx = adpcm.pred_scale >> 4;

		s32 coef1 = adpcm.coefs[coef_idx * 2 + 0];
		s32 coef2 = adpcm.coefs[coef_idx * 2 + 1];

		int temp = (samplePos & 1) ?
			   (g_dspInitialize.pARAM_Read_U8(samplePos >> 1) & 0xF) :
			   (g_dspInitialize.pARAM_Read_U8(samplePos >> 1) >> 4);

		if (temp >= 8)
			temp -= 16;

		// 0x400 = 0.5 in 11-bit fixed point
		int val = (scale * temp) + ((0x400 + coef1 * adpcm.yn1 + coef2 * adpcm.yn2) >> 11);

		if (val > 0x7FFF)
			val = 0x7FFF;
		else if (val < -0x7FFF)
			val = -0x7FFF;
		
		adpcm.yn2 = adpcm.yn1;
		adpcm.yn1 = val;

		samplePos++;
	}

	return adpcm.yn1;
}

void ADPCM_Loop(AXParamBlock& pb)
{
	if (!pb.is_stream)
	{
		pb.adpcm.yn1 = pb.adpcm_loop_info.yn1;
		pb.adpcm.yn2 = pb.adpcm_loop_info.yn2;
		pb.adpcm.pred_scale = pb.adpcm_loop_info.pred_scale;
	}
	//else stream and we should not attempt to replace values
}


// =======================================================================================
// Volume control (ramping)
// --------------
u16 ADPCM_Vol(u16 vol, u16 delta, u16 mixer_control)
{
	int x = vol;
	if (delta && delta < 0x5000)
		x += delta * 20 * 8; // unsure what the right step is
	else if (delta && delta > 0x5000)
		//x -= (0x10000 - delta); // this is to small, it's often 1
		x -= (0x10000 - delta) * 20 * 16; // if this was 20 * 8 the sounds in Fire Emblem and Paper Mario
			// did not have time to go to zero before the were closed

	 // make lower limits
	if (x < 0) x = 0;	
	//if (pb.mixer_control < 1000 && x < pb.mixer_control) x = pb.mixer_control; // does this make
		// any sense?

	// make upper limits
	//if (mixer_control > 1000 && x > mixer_control) x = mixer_control; // maybe mixer_control also
		// has a volume target?
	//if (x >= 0x7fff) x = 0x7fff; // this seems a little high
	if (x >= 0x4e20) x = 0x4e20; // add a definitive limit at 20 000
	return x; // update volume
}
// ==============


void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
{
	AXParamBlock PBs[NUMBER_OF_PBS];

	if (_iSize > 1024 * 1024)
		_iSize = 1024 * 1024;

	memset(templbuffer, 0, _iSize * sizeof(int));
	memset(temprbuffer, 0, _iSize * sizeof(int));

	// read out pbs
	int numberOfPBs = ReadOutPBs(1, PBs, NUMBER_OF_PBS);

#ifdef _WIN32
	ratioFactor = 32000.0f / (float)DSound::DSound_GetSampleRate();
#else
	ratioFactor = 32000.0f / 44100.0f;
#endif

	// write logging data to debugger
	if(m_frame)
	{
		CUCode_AX::Logging(_pBuffer, _iSize, 0);
	}


	for (int i = 0; i < numberOfPBs; i++)
	{
		AXParamBlock& pb = PBs[i];

		// get necessary values
		const u32 sampleEnd = (pb.audio_addr.end_addr_hi << 16) | pb.audio_addr.end_addr_lo;
		const u32 loopPos   = (pb.audio_addr.loop_addr_hi << 16) | pb.audio_addr.loop_addr_lo;
		const u32 updaddr   = (u32)(pb.updates.data_hi << 16) | pb.updates.data_lo;
		const u16 updpar   = Memory_Read_U16(updaddr);
		const u16 upddata   = Memory_Read_U16(updaddr + 2);


		// =======================================================================================
		/*
		Fix problems introduced with the SSBM fix - Sometimes when a music stream ended sampleEnd
		would become extremely high and the game would play random sound data from ARAM resulting in
		a strange noise. This should take care of that. - Some games (Monkey Ball 1 and Tales of
		Symphonia and other) also had one odd last block with a strange high loopPos and strange
		num_updates values, the loopPos limit turns those off also. - Please report any side effects.
		*/
		// ------------
		if (
			(sampleEnd > 0x10000000 || loopPos > 0x10000000)
			&& gSSBMremedy1
			)
		{
			pb.running = 0;

			// also reset all values if it makes any difference
			pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
			pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
			pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;

			pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
			pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;

			pb.audio_addr.looping = 0;
			pb.adpcm_loop_info.pred_scale = 0;
			pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
		}

		/*
		// the fact that no settings are reset (except running) after a SSBM type music stream or another
		looping block (for example in Battle Stadium DON) has ended could cause loud garbled sound to be
		played from one or more blocks. Perhaps it was in conjunction with the old sequenced music fix below,
		I'm not sure. This was an attempt to prevent that anyway by resetting all. But I'm not sure if this
		is needed anymore. Please try to play SSBM without it and see if it works anyway.
		*/
		if (
		// detect blocks that have recently been running that we should reset
		pb.running == 0 && pb.audio_addr.looping == 1
		//pb.running == 0 && pb.adpcm_loop_info.pred_scale

		// this prevents us from ruining sequenced music blocks, may not be needed
		/*
		&& !(pb.updates.num_updates[0] || pb.updates.num_updates[1] || pb.updates.num_updates[2]
			|| pb.updates.num_updates[3] || pb.updates.num_updates[4])
		*/	
		&& !(updpar || upddata)
	
		&& pb.mixer_control == 0 // only use this in SSBM

		&& gSSBMremedy2 // let us turn this fix on and off
		)
		{
			// reset the detection values
			pb.audio_addr.looping = 0;
			pb.adpcm_loop_info.pred_scale = 0;
			pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;

			//pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
			//pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
			//pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;

			//pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
			//pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
		}
		
		// =============


		// =======================================================================================
		// Reset all values
		// ------------
		if (gReset
			&& (pb.running || pb.audio_addr.looping || pb.adpcm_loop_info.pred_scale)
			)	
		{
			pb.running = 0;

			pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
			pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
			pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;

			pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
			pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;

			pb.audio_addr.looping = 0;
			pb.adpcm_loop_info.pred_scale = 0;
			pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
		}
		// =============


		if (pb.running)
		{
			// =======================================================================================
			// Set initial parameters
			// ------------
			//constants						
			const u32 ratio     = (u32)(((pb.src.ratio_hi << 16) + pb.src.ratio_lo) * ratioFactor);

			//variables
			u32 samplePos = (pb.audio_addr.cur_addr_hi << 16) | pb.audio_addr.cur_addr_lo;
			u32 frac = pb.src.cur_addr_frac;
			// =============

			

			// =======================================================================================
			// Handle no-src streams - No src streams have pb.src_type == 2 and have pb.src.ratio_hi = 0
			// and pb.src.ratio_lo = 0. We handle that by setting the sampling ratio integer to 1. This
			// makes samplePos update in the correct way. I'm unsure how we are actually supposed to
			// detect that this setting. Updates did not fix this automatically.
			// ---------------------------------------------------------------------------------------
			// Stream settings
				// src_type = 2 (most other games have src_type = 0)
			// ------------
			// Affected games:
				// Baten Kaitos - Eternal Wings (2003)
				// Baten Kaitos - Origins (2006)?
				// Soul Calibur 2: The movie music use src_type 2 but it needs no adjustment, perhaps
				// the sound format plays in to, Baten use ADPCM SC2 use PCM16
			// ------------
			if(pb.src_type == 2 && (pb.src.ratio_hi == 0 && pb.src.ratio_lo == 0))
			{
				pb.src.ratio_hi = 1;
			}
			// =============


			// =======================================================================================
			// Games that use looping to play non-looping music streams - SSBM has info in all
			// pb.adpcm_loop_info parameters but has pb.audio_addr.looping = 0. If we treat these streams
			// like any other looping streams the music works. I'm unsure how we are actually supposed to
			// detect that these kinds of blocks should be looping. It seems like pb.mixer_control == 0 may
			// identify these types of blocks. Updates did not write any looping values.
			// --------------
			if(
				(pb.adpcm_loop_info.pred_scale || pb.adpcm_loop_info.yn1 || pb.adpcm_loop_info.yn2)
				&& pb.mixer_control == 0
				&& gSSBM
				)
			{
				pb.audio_addr.looping = 1;
			}
			// ==============


			// =======================================================================================
			// Walk through _iSize. _iSize = numSamples. If the game goes slow _iSize will be higher to
			// compensate for that. _iSize can be as low as 100 or as high as 2000 some cases.
			for (int s = 0; s < _iSize; s++)
			{
				int sample = 0;
				frac += ratio;
				u32 newSamplePos = samplePos + (frac >> 16); //whole number of frac


				// =======================================================================================
				// Process sample format
				// --------------
				switch (pb.audio_addr.sample_format)
				{
				    case AUDIOFORMAT_PCM8:
					    pb.adpcm.yn2 = pb.adpcm.yn1; //save last sample
					    pb.adpcm.yn1 = ((s8)g_dspInitialize.pARAM_Read_U8(samplePos)) << 8;

					    if (pb.src_type == SRCTYPE_NEAREST)
					    {
						    sample = pb.adpcm.yn1;
					    }
					    else //linear interpolation
					    {
						    sample = (pb.adpcm.yn1 * (u16)frac + pb.adpcm.yn2 * (u16)(0xFFFF - frac)) >> 16;
					    }

					    samplePos = newSamplePos;
					    break;

				    case AUDIOFORMAT_PCM16:
					    pb.adpcm.yn2 = pb.adpcm.yn1; //save last sample
					    pb.adpcm.yn1 = (s16)(u16)((g_dspInitialize.pARAM_Read_U8(samplePos * 2) << 8) | (g_dspInitialize.pARAM_Read_U8((samplePos * 2 + 1))));
					    if (pb.src_type == SRCTYPE_NEAREST)
						    sample = pb.adpcm.yn1;
					    else //linear interpolation
						    sample = (pb.adpcm.yn1 * (u16)frac + pb.adpcm.yn2 * (u16)(0xFFFF - frac)) >> 16;

					    samplePos = newSamplePos;
					    break;

				    case AUDIOFORMAT_ADPCM:
					    sample = ADPCM_Step(pb, samplePos, newSamplePos, frac);
					    break;

				    default:
					    break;
				}
				// ================


				// =======================================================================================
				// Volume control
				frac &= 0xffff;

				int vol = pb.vol_env.cur_volume >> 9;
				sample = sample * vol >> 8;

				if (pb.mixer_control & MIXCONTROL_RAMPING)
				{
					int x = pb.vol_env.cur_volume;
					x += pb.vol_env.cur_volume_delta; // I'm not sure about this, can anybody find a game
					// that use this? Or how does it work?
					if (x < 0) x = 0;
					if (x >= 0x7fff) x = 0x7fff;
					pb.vol_env.cur_volume = x; // maybe not per sample?? :P
				}

				int leftmix  = pb.mixer.volume_left >> 5;
				int rightmix = pb.mixer.volume_right >> 5;
				// ===============


				int left  = sample * leftmix >> 8;
				int right = sample * rightmix >> 8;

				//adpcm has to walk from oldSamplePos to samplePos here
				templbuffer[s] += left;
				temprbuffer[s] += right;

				if (samplePos >= sampleEnd)
				{
					if (pb.audio_addr.looping == 1)
					{
						samplePos = loopPos;
						if (pb.audio_addr.sample_format == AUDIOFORMAT_ADPCM)
							ADPCM_Loop(pb);
					}
					else
					{
						pb.running = 0;
						break;
					}
				}
			} // end of the _iSize loop
			// ============

			if(gVolume) // allow us to turn this off in the debugger
			{
				pb.mixer.volume_left = ADPCM_Vol(pb.mixer.volume_left, pb.mixer.unknown, pb.mixer_control);
				pb.mixer.volume_right = ADPCM_Vol(pb.mixer.volume_right, pb.mixer.unknown2, pb.mixer_control);
			}

			pb.src.cur_addr_frac = (u16)frac;
			pb.audio_addr.cur_addr_hi = samplePos >> 16;
			pb.audio_addr.cur_addr_lo = (u16)samplePos;
		}
	}

	for (int i = 0; i < _iSize; i++)
	{
		// Clamp into 16-bit. Maybe we should add a volume compressor here.
		int left  = templbuffer[i];
		int right = temprbuffer[i];
		if (left < -32767)  left = -32767;
		if (left > 32767)   left = 32767;
		if (right < -32767) right = -32767;
		if (right >  32767) right = 32767;
		*_pBuffer++ += left;
		*_pBuffer++ += right;
	}

	// write back out pbs
	WriteBackPBs(PBs, numberOfPBs);

	// write logging data to debugger again after the update
	if(m_frame)
	{
		CUCode_AX::Logging(_pBuffer, _iSize, 1);
	}
}

void CUCode_AX::Update()
{
	// check if we have to sent something
	if (!m_rMailHandler.IsEmpty())
	{
		g_dspInitialize.pGenerateDSPInterrupt();
	}
}

// AX seems to bootup one task only and waits for resume-callbacks
// everytime the DSP has "spare time" it sends a resume-mail to the CPU
// and the __DSPHandler calls a AX-Callback which generates a new AXFrame
bool CUCode_AX::AXTask(u32& _uMail)
{
	u32 uAddress = _uMail;
	DebugLog("AXTask - AXCommandList-Addr: 0x%08x", uAddress);

	u32 Addr__AXStudio;
	u32 Addr__AXOutSBuffer;
	u32 Addr__AXOutSBuffer_1;
	u32 Addr__AXOutSBuffer_2;
	u32 Addr__A;
	u32 Addr__12;
	u32 Addr__4_1;
	u32 Addr__4_2;
	u32 Addr__5_1;
	u32 Addr__5_2;
	u32 Addr__6;
	u32 Addr__9;

	bool bExecuteList = true;

	while (bExecuteList)
	{
		static int last_valid_command = 0;
		u16 iCommand = Memory_Read_U16(uAddress);
		uAddress += 2;
		switch (iCommand)
		{
	    case AXLIST_STUDIOADDR: //00
		    Addr__AXStudio = Memory_Read_U32(uAddress);
		    uAddress += 4;
			if (wii_mode)
				uAddress += 6;
		    DebugLog("AXLIST studio address: %08x", Addr__AXStudio);
		    break;

	    case 0x001:
	    {
		    u32 address = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    u16 param1 = Memory_Read_U16(uAddress);
		    uAddress += 2;
		    u16 param2 = Memory_Read_U16(uAddress);
		    uAddress += 2;
		    u16 param3 = Memory_Read_U16(uAddress);
		    uAddress += 2;
		    DebugLog("AXLIST 1: %08x, %04x, %04x, %04x", address, param1, param2, param3);
	    }
		    break;

	    //
	    // Somewhere we should be getting a bitmask of AX_SYNC values
	    // that tells us what has been updated
	    // Dunno if important
	    //
	    case AXLIST_PBADDR: //02
		    {
		    m_addressPBs = Memory_Read_U32(uAddress);
		    uAddress += 4;

		    mixer_HLEready = true;
		    DebugLog("AXLIST PB address: %08x", m_addressPBs);
#ifdef _WIN32
		    DebugLog("Update the SoundThread to be in sync");
		    DSound::DSound_UpdateSound(); //do it in this thread to avoid sync problems
#endif
		    }
		    break;

	    case 0x0003:
		    DebugLog("AXLIST command 0x0003 ????");
		    break;

	    case 0x0004:
		    Addr__4_1 = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    Addr__4_2 = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    DebugLog("AXLIST 4_1 4_2 addresses: %08x %08x", Addr__4_1, Addr__4_2);
		    break;

	    case 0x0005:
		    Addr__5_1 = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    Addr__5_2 = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    DebugLog("AXLIST 5_1 5_2 addresses: %08x %08x", Addr__5_1, Addr__5_2);
		    break;

	    case 0x0006:
		    Addr__6   = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    DebugLog("AXLIST 6 address: %08x", Addr__6);
		    break;

	    case AXLIST_SBUFFER:
			// Hopefully this is where in main ram to write.
		    Addr__AXOutSBuffer = Memory_Read_U32(uAddress);
		    uAddress += 4;
			if (wii_mode) {
				uAddress += 12;
			}
		    DebugLog("AXLIST OutSBuffer address: %08x", Addr__AXOutSBuffer);
		    break;

	    case 0x0009:
		    Addr__9   = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    DebugLog("AXLIST 6 address: %08x", Addr__9);
		    break;

	    case AXLIST_COMPRESSORTABLE:  // 0xa
		    Addr__A   = Memory_Read_U32(uAddress);
		    uAddress += 4;
			if (wii_mode) {
				// There's one more here.
//				uAddress += 4;
			}
		    DebugLog("AXLIST CompressorTable address: %08x", Addr__A);
		    break;

	    case 0x000e:
		    Addr__AXOutSBuffer_1 = Memory_Read_U32(uAddress);
		    uAddress += 4;

			// Addr__AXOutSBuffer_2 is the address in RAM that we are supposed to mix to.
			// Although we don't, currently.
		    Addr__AXOutSBuffer_2 = Memory_Read_U32(uAddress);
		    uAddress += 4;
		    DebugLog("AXLIST sbuf2 addresses: %08x %08x", Addr__AXOutSBuffer_1, Addr__AXOutSBuffer_2);
		    break;

	    case AXLIST_END:
		    bExecuteList = false;
		    DebugLog("AXLIST end");
		    break;

	    case 0x0010:  //Super Monkey Ball 2
		    DebugLog("AXLIST unknown");
		    //should probably read/skip stuff here
		    uAddress += 8;
		    break;

	    case 0x0011:
		    uAddress += 4;
		    break;

	    case 0x0012:
		    Addr__12  = Memory_Read_U16(uAddress);
		    uAddress += 2;
		    break;

	    case 0x0013:
		    uAddress += 6 * 4; // 6 Addresses.
		    break;

		case 0x000d:
			if (wii_mode) {
				uAddress += 4 * 4;  // 4 addresses. another aux?
				break;
			}
			// non-wii : fall through

		case 0x000b:
			if (wii_mode) {
				uAddress += 2;  // one 0x8000 in rabbids
				uAddress += 4 * 2; // then two RAM addressses
				break;
			}
			// non-wii : fall through

	    default:
			{
		    static bool bFirst = true;
		    if (bFirst == true)
		    {
			    char szTemp[2048];
				sprintf(szTemp, "Unknown AX-Command 0x%x (address: 0x%08x). Last valid: %02x\n",
					    iCommand, uAddress - 2, last_valid_command);
			    int num = -32;
			    while (num < 64+32)
			    {
				    char szTemp2[128] = "";
					sprintf(szTemp2, "%s0x%04x\n", num == 0 ? ">>" : "  ", Memory_Read_U16(uAddress + num));
				    strcat(szTemp, szTemp2);
				    num += 2;
			    }

			    PanicAlert(szTemp);
			    bFirst = false;
		    }

		    // unknown command so stop the execution of this TaskList
		    bExecuteList = false;
			}
			break;
		}
		if (bExecuteList)
			last_valid_command = iCommand;
	}
	DebugLog("AXTask - done, send resume");

	// i hope resume is okay AX
	m_rMailHandler.PushMail(0xDCD10001);
	return true;
}

int CUCode_AX::ReadOutPBs(int a, AXParamBlock* _pPBs, int _num)
{
	int count = 0;
	u32 blockAddr = m_addressPBs;

	// reading and 'halfword' swap
	for (int i = 0; i < _num; i++)
	{
		const short *pSrc = (const short *)g_dspInitialize.pGetMemoryPointer(blockAddr);
		if (pSrc != NULL)
		{
			short *pDest = (short *)&_pPBs[i];
			for (size_t p = 0; p < sizeof(AXParamBlock) / 2; p++)
			{
				pDest[p] = Common::swap16(pSrc[p]);

				// To avoid a performance drop in the Release build I place this in the debug
				// build only
				#if defined(_DEBUG) || defined(DEBUGFAST)
					gLastBlock = blockAddr + p*2 + 2;  // save last block location
				#endif
			}
			// ---------------------------------------------------------------------------------------
			// Make the updates we are told to do
			// ------------
			if(a) // only do this once every 5 ms
			{
				u16 upd0 = pDest[34]; u16 upd1 = pDest[35]; u16 upd2 = pDest[36]; // num_updates
				u16 upd3 = pDest[37]; u16 upd4 = pDest[38];
				u16 upd_hi = pDest[39]; // update addr
				u16	upd_lo = pDest[40];
				const u32 updaddr   = (u32)(upd_hi << 16) | upd_lo;
				const u16 updpar   = Memory_Read_U16(updaddr);
				const u16 upddata   = Memory_Read_U16(updaddr + 2);
				// some safety checks, I hope it's enough, how long does the memory go?
				if(updaddr > 0x80000000 && updaddr < 0x82000000
					&& updpar < 63 && updpar > 3 && upddata >= 0 // updpar > 3 because we don't want to change
					// 0-3, those are important
					&& (upd0 || upd1 || upd2 || upd3 || upd4) // We should use these in some way to I think
					// but I don't know how or when
					&& gSequenced) // on and off option
				{
					pDest[updpar] = upddata;
				}
			}
			//aprintf(1, "%08x %04x %04x\n", updaddr, updpar, upddata);
			// ------------

			blockAddr = (_pPBs[i].next_pb_hi << 16) | _pPBs[i].next_pb_lo;
			count++;			
		}
		else
			break;
	}

	// return the number of read PBs
	return count;
}

void CUCode_AX::WriteBackPBs(AXParamBlock* _pPBs, int _num)
{
	u32 blockAddr = m_addressPBs;

	// write back and 'halfword'swap
	for (int i = 0; i < _num; i++)
	{
		short* pSrc  = (short*)&_pPBs[i];
		short* pDest = (short*)g_dspInitialize.pGetMemoryPointer(blockAddr);
		for (size_t p = 0; p < sizeof(AXParamBlock) / 2; p++)
		{
			pDest[p] = Common::swap16(pSrc[p]);
		}

		// next block
		blockAddr = (_pPBs[i].next_pb_hi << 16) | _pPBs[i].next_pb_lo;
	}
}