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

// 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"

#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 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 < 0x4000)
		x += delta; // unsure what the right step is
		//x ++;
		//x += 8; //?
	else if (delta && delta > 0x4000)
		//x -= (0x8000 - pb.mixer.unknown); // this didn't work
		x--;
	if (x < 0) x = 0; // make limits
	// does this make any sense?
	//if (pb.mixer_control < 1000 && x < pb.mixer_control) x = pb.mixer_control;
	//if (x >= 0x7fff) xl = 0x7fff; // this seems to high
	if (mixer_control > 1000 && x > mixer_control) x = mixer_control;	
	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(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 u32 upddata   = Memory_Read_U32(updaddr);


		// =======================================================================================
		/*
		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])
		*/	
		&& !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;
		}
		
		// =============


		// =======================================================================================
		/*
		Sequenced music fix - Because SSBM type music and other (for example Battle Stadium DON) looping
		blocks did no have its pred_scale (or any other parameter except running) turned off after a song
		was stopped a pred_scale check here had the effect of turning those blocks on immediately after
		the stopped. One way to easily test this is to start a game in BS DON, wait for the Fight message
		so that the loops begin, and then exit the game and you can see that the sound effects will
		continue to play in the menus. That's not good. Because the pred_scale check caused these effects
		I'm trying an update data check instead, it relieas on the assumption that all games that don't
		use sequencing have blank memory at the update address so that upddata = 0 in those cases. That
		turned out to now hold, many games that don't use sequencing still had update_addr pointing to
		some memory location with data, either inside the parameter block space or close before or after
		it.
		*/
		// ------------
		//if (!pb.running && pb.adpcm_loop_info.pred_scale)	
		//if (!pb.running && pb.audio_addr.looping)		
		if (!pb.running && upddata)
		/*
		if (!pb.running && 
			(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])			
			&& gSequenced
			)
		*/
		{
			pb.running = 1;
		}
		// =============


		// =======================================================================================
		// 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.
			// ---------------------------------------------------------------------------------------
			// 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. It seems like pb.mixer_control == 0 may
			// identify these types of blocks.
			// --------------
			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
			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?
					if (x < 0) x = 0;
					if (x >= 0x7fff) x = 0x7fff;
					pb.vol_env.cur_volume = x; // maybe not per sample?? :P

					// strange way to not use this in Skies where it didn't work well
					if(pb.mixer_control != 9 && pb.mixer_control != 123)
					{
						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);
					}
				}

				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
			// ============


			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(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
			}
			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;
	}
}