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

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// 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
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
}
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];
// =======================================================================================
/*
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) also had one odd block with a strange high loopPos and strange num_updates values,
the loopPos limit turns those off also. - Please report any side effects.Please report any
side effects.
*/
// ------------
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;
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.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 has ended
could cause loud garbled sound to be played from several blocks. It could be seen as five or six
simultaneous looping blocks that presumable produced garbled music. My guess is that it was sound
effects that were placed in previous music blocks and mutated into these looping noise machines.
*/
if (
// detect blocks that have recently been running that we should reset
pb.running == 0 && pb.audio_addr.looping == 1
// this prevents us from ruining sequenced music blocks
&& !(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])
&& gSSBMremedy2 // let us turn this fix on and off
)
{
// reset all values, or mostly all
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.audio_addr.looping = 0;
pb.adpcm_loop_info.pred_scale = 0;
pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
}
// =============
// =======================================================================================
/*
// Sequenced music fix - Because SSBM type music 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. Because the pred_scale check caused these
effects I'm trying the num_updates check instead. Please report any side effects.
*/
// ------------
//if (!pb.running && pb.adpcm_loop_info.pred_scale)
/**/
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])
)
{
pb.running = 1;
}
// =============
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)?
// ?
// ------------
if(pb.src_type == 2)
{
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.
// ---------------------------------------------------------------------------------------
if(
pb.adpcm_loop_info.pred_scale || pb.adpcm_loop_info.yn1 || pb.adpcm_loop_info.yn2
&& gSSBM
)
{
pb.audio_addr.looping = 1;
}
// =======================================================================================
// =======================================================================================
// Streaming music and volume - The streaming music in Paper Mario use these settings:
// Base settings
// is_stream = 1
// src_type = 0
// PBAudioAddr
// audio_addr.looping = 1 (adpcm_loop_info.pred_scale = value, .yn1 = 0, .yn2 = 0)
/*
However. Some of the ingame music and seemingly randomly some other music incorrectly get
volume = 0 for both left and right. This also affects Fire Emblem. But Starfox Assault
that also use is_stream = 1 has no problem wuth the volume, but its settings are somewhat
different, it uses src_type = 1 and pb.src.ratio_lo (fraction) != 0
*/
// =======================================================================================
// =======================================================================================
// 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;
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
// =======================================================================================
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);
}
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 readed 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;
}
}
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