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Some preparations for Wii AX (much work remains)

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git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@1101 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
hrydgard
2008-11-09 16:19:30 +00:00
parent 530fb9ba3d
commit 860ffe9541
4 changed files with 341 additions and 295 deletions
Download Patch File Download Diff File Expand all files Collapse all files
Source/Plugins/Plugin_DSP_HLE/Src/Logging/Logging.cpp
+4 -4
View File
@@ -141,8 +141,8 @@ int vectorLength2 = 100; // for console version
// should we worry about the additonal memory these lists require? bool will allocate
// very little memory
std::vector< std::vector<bool> > vector1(64, std::vector<bool>(vectorLength,0));
std::vector< std::vector<bool> > vector2(64, std::vector<bool>(vectorLength2,0));
std::vector< std::vector<bool> > vector1(64, std::vector<bool>(vectorLength, 0));
std::vector< std::vector<bool> > vector2(64, std::vector<bool>(vectorLength2, 0));
std::vector<int> numberRunning(64);
@@ -301,12 +301,12 @@ void CUCode_AX::Logging(short* _pBuffer, int _iSize, int a)
}
else
{
Conditions = (numberRunning.at(i) > 0 || PBs[i].audio_addr.looping);
Conditions = numberRunning.at(i) > 0 || PBs[i].audio_addr.looping;
}
// --------------
if(Conditions)
if (Conditions)
{
// AXPB base
gcoef[i] = PBs[i].unknown1;
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AX.cpp
+270 -282
View File
@@ -28,7 +28,6 @@
#include "UCode_AXStructs.h"
#include "UCode_AX.h"
// ---------------------------------------------------------------------------------------
// Externals
// -----------
@@ -43,7 +42,6 @@ bool gReset = false; // used externally
extern CDebugger* m_frame;
// -----------
CUCode_AX::CUCode_AX(CMailHandler& _rMailHandler, bool wii)
: IUCode(_rMailHandler)
, m_addressPBs(0xFFFFFFFF)
@@ -76,7 +74,7 @@ void CUCode_AX::HandleMail(u32 _uMail)
}
}
s16 CUCode_AX::ADPCM_Step(AXParamBlock& pb, u32& samplePos, u32 newSamplePos, u16 frac)
s16 ADPCM_Step(AXParamBlock& pb, u32& samplePos, u32 newSamplePos, u16 frac)
{
PBADPCMInfo &adpcm = pb.adpcm;
@@ -158,302 +156,294 @@ u16 ADPCM_Vol(u16 vol, u16 delta, u16 mixer_control)
}
// ==============
void MixAddVoice(AXParamBlock &pb, int *templbuffer, int *temprbuffer, int _iSize)
{
#ifdef _WIN32
ratioFactor = 32000.0f / (float)DSound::DSound_GetSampleRate();
#else
ratioFactor = 32000.0f / 44100.0f;
#endif
// 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; pb.src.ratio_hi = 0; pb.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; pb.src.ratio_hi = 0; pb.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;
}
}
void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
{
AXParamBlock PBs[NUMBER_OF_PBS];
// read out pbs
int numberOfPBs = ReadOutPBs(1, 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)
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;
}
MixAddVoice(pb, templbuffer, temprbuffer, _iSize);
}
// write back out pbs
WriteBackPBs(PBs, numberOfPBs);
for (int i = 0; i < _iSize; i++)
{
// Clamp into 16-bit. Maybe we should add a volume compressor here.
@@ -467,9 +457,6 @@ void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
*_pBuffer++ = right;
}
// write back out pbs
WriteBackPBs(PBs, numberOfPBs);
// write logging data to debugger again after the update
if (m_frame)
{
@@ -477,6 +464,7 @@ void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
}
}
void CUCode_AX::Update()
{
// check if we have to sent something
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AX.h
+5 -7
View File
@@ -20,6 +20,11 @@
#include "UCode_AXStructs.h"
enum
{
NUMBER_OF_PBS = 64
};
class CUCode_AX : public IUCode
{
public:
@@ -34,12 +39,6 @@ public:
void Logging(short* _pBuffer, int _iSize, int a);
private:
enum
{
NUMBER_OF_PBS = 64
};
enum
{
MAIL_AX_ALIST = 0xBABE0000,
@@ -64,7 +63,6 @@ private:
void SendMail(u32 _uMail);
int ReadOutPBs(int a, AXParamBlock *_pPBs, int _num);
void WriteBackPBs(AXParamBlock *_pPBs, int _num);
s16 ADPCM_Step(AXParamBlock& pb, u32& samplePos, u32 newSamplePos, u16 frac);
};
#endif // _UCODE_AX
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AXStructs.h
+62 -2
View File
@@ -29,6 +29,17 @@ struct PBMixer
u16 unknown4[6];
};
struct PBMixerWii
{
u16 volume_left;
u16 unknown;
u16 volume_right;
u16 unknown2;
u16 unknown3[12];
u16 unknown4[8];
};
struct PBInitialTimeDelay
{
u16 on;
@@ -51,11 +62,23 @@ struct PBUpdates
u16 data_lo;
};
struct PBUnknown
struct PBUpdatesWii
{
u16 num_updates[3];
u16 data_hi; // These point to main RAM. Not sure about the structure of the data.
u16 data_lo;
};
struct PBDpop
{
s16 unknown[9];
};
struct PBDpopWii
{
s16 unknown[12];
};
struct PBVolumeEnvelope
{
u16 cur_volume;
@@ -121,7 +144,7 @@ struct AXParamBlock
/* 9 */ PBMixer mixer;
/* 27 */ PBInitialTimeDelay initial_time_delay;
/* 34 */ PBUpdates updates;
/* 41 */ PBUnknown unknown2;
/* 41 */ PBDpop dpop;
/* 50 */ PBVolumeEnvelope vol_env;
/* 52 */ PBUnknown2 unknown3;
/* 55 */ PBAudioAddr audio_addr;
@@ -131,6 +154,43 @@ struct AXParamBlock
/* 93 */ u16 unknown_maybe_padding[3];
};
struct PBLpf
{
u16 enabled;
u16 yn1;
u16 a0;
u16 b0;
};
struct AXParamBlockWii
{
u16 next_pb_hi;
u16 next_pb_lo;
u16 this_pb_hi;
u16 this_pb_lo;
u16 src_type; // Type of sample rate converter (none, ?, linear)
u16 coef_select;
u16 mixer_control;
u16 running; // 1=RUN 0=STOP
u16 is_stream; // 1 = stream, 0 = one shot
PBMixerWii mixer;
PBInitialTimeDelay initial_time_delay;
PBUpdatesWii updates;
PBDpopWii dpop;
PBVolumeEnvelope vol_env;
PBUnknown2 unknown3;
PBAudioAddr audio_addr;
PBADPCMInfo adpcm;
PBSampleRateConverter src;
PBADPCMLoopInfo adpcm_loop_info;
PBLpf lpf;
u16 pad[22];
};
enum {
AUDIOFORMAT_ADPCM = 0,
AUDIOFORMAT_PCM8 = 0x19,