DSB2 IRQ 2 now fires at 1KHz rather than once per frame, improving

music timing in Daytona USA 2 and Sega Rally 2. DSB1 CPU timing 
increased from 1MHz to 4MHz, improving music fade timing in Scud Race. 
Thanks to gm_matthew for these discoveries.
This commit is contained in:
SpinDizzy 2021-03-03 15:30:19 +00:00
parent e3374256ff
commit 46b1de2238
2 changed files with 212 additions and 180 deletions

View file

@ -1,12 +1,13 @@
/**
** Supermodel
** A Sega Model 3 Arcade Emulator.
** Copyright 2011 Bart Trzynadlowski, Nik Henson
** Copyright 2011-2021 Bart Trzynadlowski, Nik Henson, Ian Curtis,
** Harry Tuttle, and Spindizzi
**
** This file is part of Supermodel.
**
** Supermodel 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
** the terms of the GNU General Public License as published by the Free
** Software Foundation, either version 3 of the License, or (at your option)
** any later version.
**
@ -18,12 +19,12 @@
** You should have received a copy of the GNU General Public License along
** with Supermodel. If not, see <http://www.gnu.org/licenses/>.
**/
/*
* DSB.cpp
*
*
* Sega Digital Sound Board (MPEG audio). Implementation of the CDSB1 and CDSB2
* classes. Based on code donated by R. Belmont. Many Bothans died to bring us
* classes. Based on code donated by R. Belmont. Many Bothans died to bring us
* this emulation.
*
* TODO List
@ -43,61 +44,61 @@
/******************************************************************************
Resampler
MPEG Layer 2 audio can be 32, 44.1, or 48 KHz. Here, an up-sampling algorithm
is provided, which should work for any frequency less than 44.1 KHz and an
output frequency of 44.1 KHz. Down-sampling is not yet implemented, but would
is provided, which should work for any frequency less than 44.1 KHz and an
output frequency of 44.1 KHz. Down-sampling is not yet implemented, but would
work in a similar fashion. The chief difference is that the input index would
sometimes advance by more than one for a single output sample and the
fractions, nFrac and pFrac, would sometimes exceed 1.0.
Up-Sampling Description
-----------------------
Linear interpolation is used to up-sample. Not as accurate as the Shannon
reconstruction equation but it seems to work quite well.
1. Linear Interpolation
Input samples for a given frame (here, this means 1/60Hz, not to be confused
with an MPEG frame, which is shorter) are numbered 0 ... L-1 (L samples in
with an MPEG frame, which is shorter) are numbered 0 ... L-1 (L samples in
total). Output samples are 0 ... M-1.
For two adjacent input samples at times p ("previous") and n ("next"), in[p]
and in[n], and output out[t] at time t, linear interpolation yields:
out[t] = (n-t)/(n-p) * in[p] + (t-p)/(n-p) * in[n]
Note that (n-p) = 1/fin (fin being the input sampling frequency).
Let pFrac = (n-t)/(n-p) and nFrac = (t-p)/(n-p). As t moves from p to n, pFrac
moves from 1 to 0 and nFrac from 0 to 1, as we expect.
If we proceed one output sample at a time, we must add the time difference
If we proceed one output sample at a time, we must add the time difference
between output samples, 1/fout, to t. Call this delta_t. If we divide delta_t
by (n-p), we can add it directly to nFrac and subtract from pFrac. Therefore:
delta = (1/fout)/(n-p) = fin/fout
What happens when nFrac exceeds 1.0 or pFrac goes below 0.0? That can't
be allowed to happen -- it means that we've actually moved along the line into
the region between the next set of samples. We use pFrac < 0 as the condition
to update the input samples.
It so happens that when fin < fout, pFrac and nFrac will never exceed 1.0. So
there is no need to check or mask the fixed point values when using them to
interpolate samples.
2. Input Buffer Overflows
For some low sampling rates, particularly those that are a factor of 2 or 4
smaller, it is possible that the very last sample or two needed from the input
stream will be beyond the end. Fetching two extra samples (which can introduce
an update lag of two samples -- imperceptible and inconsequential) fixes this,
and so we do it.
3. Continuity Between Frames
The very last output sample will typically sit somewhere between two input
samples. It is wrong to start the next frame by assuming everything is lined
up again. The first sample of the next frame will often have to be interpol-
@ -105,17 +106,17 @@
to see how many input samples remain unprocessed when up-sampling is finished,
and then copy those to the beginning of the buffer. We then return the number
of samples so that the buffer update function will know to skip them.
We also must maintain the state of pFrac and nFrac to resume interpolation
We also must maintain the state of pFrac and nFrac to resume interpolation
correctly. Therefore, these variables are persistent.
4. Fixed Point Arithmetic
Fixed point arithmetic is used to track fractions. For such numbers, the low
8 bits represent a fraction (0x100 would be 1.0, 0x080 would be 0.5, etc.)
and the upper bits are the integral portion.
******************************************************************************/
void CDSBResampler::Reset(void)
{
// Initial state of fractions (24.8 fixed point)
@ -142,48 +143,48 @@ int CDSBResampler::UpSampleAndMix(INT16 *outL, INT16 *outR, INT16 *inL, INT16 *i
int inIdx = 0;
INT32 leftSample, rightSample, leftSoundSample, rightSoundSample;
INT32 v[2], musicVol, soundVol;
// Obtain program volume settings and convert to 24.8 fixed point (0-200 -> 0x00-0x200)
musicVol = m_config["MusicVolume"].ValueAs<int>();
soundVol = m_config["SoundVolume"].ValueAs<int>();
musicVol = (INT32) ((float) 0x100 * (float) musicVol / 100.0f);
soundVol = (INT32) ((float) 0x100 * (float) soundVol / 100.0f);
// Scale volume from 0x00-0xFF -> 0x00-0x100 (24.8 fixed point)
v[0] = (INT16) ((float) 0x100 * (float) volumeL / 255.0f);
v[1] = (INT16) ((float) 0x100 * (float) volumeR / 255.0f);
// Up-sample and mix!
while (outIdx < sizeOut)
{
// nFrac, pFrac will never exceed 1.0 (0x100) (only true if delta does not exceed 1)
leftSample = ((int)inL[inIdx]*pFrac+(int)inL[inIdx+1]*nFrac) >> 8; // left channel
rightSample = ((int)inR[inIdx]*pFrac+(int)inR[inIdx+1]*nFrac) >> 8; // right channel
// Apply DSB volume and then overall music volume setting
leftSample = (leftSample*v[0]*musicVol) >> 16; // multiplied by two 24.8 numbers, shift back by 16
rightSample = (rightSample*v[0]*musicVol) >> 16;
// Apply sound volume setting
leftSoundSample = (outL[outIdx]*soundVol) >> 8;
rightSoundSample = (outR[outIdx]*soundVol) >> 8;
// Mix and output
outL[outIdx] = MixAndClip(leftSoundSample, leftSample);
outR[outIdx] = MixAndClip(rightSoundSample, rightSample);
outIdx++;
// Time step
pFrac -= delta;
nFrac += delta;
// Time to move to next samples?
if (pFrac <= 0) // when pFrac becomes 0, advance samples, reset pFrac to 1
{
pFrac += (1<<8);
nFrac -= (1<<8);
inIdx++; // advance samples (for upsampling only; downsampling may advance by more than one -- add delta every loop iteration)
}
}
@ -210,7 +211,7 @@ UINT8 CDSB1::Read8(UINT32 addr)
// ROM: 0x0000-0x7FFF
if (addr < 0x8000)
return progROM[addr];
// RAM: 0x8000-0xFFFF
return ram[addr&0x7FFF];
}
@ -259,17 +260,17 @@ void CDSB1::IOWrite8(UINT32 addr, UINT8 data)
return;
}
break;
case 0xE2: // MPEG start, high byte
startLatch &= 0x00FFFF;
startLatch |= ((UINT32)data) << 16;
break;
case 0xE3: // MPEG start, middle byte
startLatch &= 0xFF00FF;
startLatch |= ((UINT32)data) << 8;
break;
case 0xE4: // MPEG start, low byte
startLatch &= 0xFFFF00;
startLatch |= data;
@ -297,19 +298,19 @@ void CDSB1::IOWrite8(UINT32 addr, UINT8 data)
MpegDec::UpdateMemory(&mpegROM[usingLoopStart], usingLoopEnd, true);
}
}
break;
case 0xE5: // MPEG end, high byte
endLatch &= 0x00FFFF;
endLatch |= ((UINT32)data) << 16;
break;
break;
case 0xE6: // MPEG end, middle byte
endLatch &= 0xFF00FF;
endLatch |= ((UINT32)data) << 8;
break;
case 0xE7: // MPEG end, low byte
endLatch &= 0xFFFF00;
endLatch |= data;
@ -327,20 +328,20 @@ void CDSB1::IOWrite8(UINT32 addr, UINT8 data)
MpegDec::UpdateMemory(&mpegROM[usingLoopStart], usingLoopEnd, true);
//printf("loopEnd = %08X\n", loopEnd);
}
break;
break;
case 0xE8: // MPEG volume
volume = 0x7F-data;
//printf("Set Volume: %02X\n", volume);
break;
case 0xE9: // MPEG stereo
stereo = data;
break;
case 0xF0: // command echo back
break;
default:
//printf("Z80 Port %02X=%08X\n", addr, data);
break;
@ -350,24 +351,24 @@ void CDSB1::IOWrite8(UINT32 addr, UINT8 data)
UINT8 CDSB1::IORead8(UINT32 addr)
{
int progress;
switch ((addr&0xFF))
{
case 0xE2: // MPEG position, high byte
progress = MpegDec::GetPosition();
progress += mpegStart; // byte address currently playing
return (progress>>16)&0xFF;
case 0xE3: // MPEG position, middle byte
progress = MpegDec::GetPosition();
progress += mpegStart;
return (progress>>8)&0xFF;
case 0xE4: // MPEG position, low byte
progress = MpegDec::GetPosition();
progress += mpegStart;
return progress&0xFF;
case 0xF0: // Latch
UINT8 d;
d = fifo[fifoIdxR]; // retrieve next command byte
@ -376,16 +377,16 @@ UINT8 CDSB1::IORead8(UINT32 addr)
fifoIdxR++;
fifoIdxR &= 127;
}
if (fifoIdxR == fifoIdxW) // FIFO empty?
status &= ~2; // yes, indicate no commands left
else
status |= 2;
Z80.SetINT(false); // clear IRQ
//printf("Z80: INT cleared, read from FIFO\n");
return d;
case 0xF1: // Status
/*
* Bit 0: Must be 1 for most games.
@ -394,7 +395,7 @@ UINT8 CDSB1::IORead8(UINT32 addr)
*/
return status;
}
//printf("Z80 Port Read %02X\n", addr);
return 0;
}
@ -402,7 +403,7 @@ UINT8 CDSB1::IORead8(UINT32 addr)
static int Z80IRQCallback(CZ80 *Z80)
{
return 0x38;
}
}
void CDSB1::SendCommand(UINT8 data)
{
@ -415,7 +416,7 @@ void CDSB1::SendCommand(UINT8 data)
fifo[fifoIdxW++] = data;
fifoIdxW &= 127;
//printf("Write FIFO: %02X\n", data);
// Have we caught up to the read pointer?
#ifdef DEBUG
if (fifoIdxW == fifoIdxR)
@ -427,7 +428,7 @@ void CDSB1::RunFrame(INT16 *audioL, INT16 *audioR)
{
int cycles;
UINT8 v;
if (!m_config["EmulateDSB"].ValueAs<bool>())
{
// DSB code applies SCSP volume, too, so we must still mix
@ -436,23 +437,23 @@ void CDSB1::RunFrame(INT16 *audioL, INT16 *audioR)
retainedSamples = Resampler.UpSampleAndMix(audioL, audioR, mpegL, mpegR, 0, 0, 44100/60, 32000/60+2, 44100, 32000);
return;
}
// While FIFO not empty, fire interrupts, run for up to one frame
for (cycles = (4000000/60)/4; (cycles > 0) && (fifoIdxR != fifoIdxW); )
for (cycles = (4000000/60); (cycles > 0) && (fifoIdxR != fifoIdxW); )
{
Z80.SetINT(true); // fire an IRQ to indicate pending command
//printf("Z80 INT fired\n");
cycles -= Z80.Run(500);
}
}
// Run remaining cycles
Z80.Run(cycles);
//printf("VOLUME=%02X STEREO=%02X\n", volume, stereo);
// Convert volume from 0x00-0x7F -> 0x00-0xFF
v = (UINT8) ((float) 255.0f * (float) volume /127.0f);
// Decode MPEG for this frame
MpegDec::DecodeAudio(&mpegL[retainedSamples], &mpegR[retainedSamples], 32000 / 60 - retainedSamples + 2);
retainedSamples = Resampler.UpSampleAndMix(audioL, audioR, mpegL, mpegR, v, v, 44100/60, 32000/60+2, 44100, 32000);
@ -463,15 +464,15 @@ void CDSB1::Reset(void)
MpegDec::Stop();
Resampler.Reset();
retainedSamples = 0;
memset(fifo, 0, sizeof(fifo));
fifoIdxW = fifoIdxR = 0;
status = 1;
mpegState = 0; // why doesn't RB ever init this?
volume = 0x7F; // full volume
usingLoopStart = 0;
// Even if DSB emulation is disabled, must reset to establish valid Z80 state
Z80.Reset();
DebugLog("DSB1 Reset\n");
@ -481,9 +482,9 @@ void CDSB1::SaveState(CBlockFile *StateFile)
{
UINT32 playOffset, endOffset;
UINT8 isPlaying;
StateFile->NewBlock("DSB1", __FILE__);
// MPEG playback state
isPlaying = (UINT8)MpegDec::IsLoaded();
playOffset = (UINT32)MpegDec::GetPosition();
@ -496,7 +497,7 @@ void CDSB1::SaveState(CBlockFile *StateFile)
StateFile->Write(&usingMPEGEnd, sizeof(usingMPEGEnd));
StateFile->Write(&usingLoopStart, sizeof(usingLoopStart));
StateFile->Write(&usingLoopEnd, sizeof(usingLoopEnd));
// MPEG board state
StateFile->Write(ram, 0x8000);
StateFile->Write(fifo, sizeof(fifo));
@ -511,7 +512,7 @@ void CDSB1::SaveState(CBlockFile *StateFile)
StateFile->Write(&cmdLatch, sizeof(cmdLatch));
StateFile->Write(&volume, sizeof(volume));
StateFile->Write(&stereo, sizeof(stereo));
// Z80 CPU state
Z80.SaveState(StateFile, "DSB1 Z80");
}
@ -520,13 +521,13 @@ void CDSB1::LoadState(CBlockFile *StateFile)
{
UINT32 playOffset, endOffset;
UINT8 isPlaying;
if (OKAY != StateFile->FindBlock("DSB1"))
{
ErrorLog("Unable to load Digital Sound Board state. Save state file is corrupt.");
return;
}
StateFile->Read(&isPlaying, sizeof(isPlaying));
StateFile->Read(&playOffset, sizeof(playOffset));
StateFile->Read(&endOffset, sizeof(endOffset));
@ -547,9 +548,9 @@ void CDSB1::LoadState(CBlockFile *StateFile)
StateFile->Read(&cmdLatch, sizeof(cmdLatch));
StateFile->Read(&volume, sizeof(volume));
StateFile->Read(&stereo, sizeof(stereo));
Z80.LoadState(StateFile, "DSB1 Z80");
// Restart MPEG audio at the appropriate position
if (isPlaying)
{
@ -575,27 +576,27 @@ void CDSB1::LoadState(CBlockFile *StateFile)
bool CDSB1::Init(const UINT8 *progROMPtr, const UINT8 *mpegROMPtr)
{
float memSizeMB = (float)DSB1_MEMORY_POOL_SIZE/(float)0x100000;
// Receive ROM
progROM = progROMPtr;
mpegROM = mpegROMPtr;
// Allocate memory pool
memoryPool = new(std::nothrow) UINT8[DSB1_MEMORY_POOL_SIZE];
if (NULL == memoryPool)
return ErrorLog("Insufficient memory for DSB1 board (needs %1.1f MB).", memSizeMB);
memset(memoryPool, 0, DSB1_MEMORY_POOL_SIZE);
// Set up memory pointers
ram = &memoryPool[DSB1_OFFSET_RAM];
mpegL = (INT16 *) &memoryPool[DSB1_OFFSET_MPEG_LEFT];
mpegR = (INT16 *) &memoryPool[DSB1_OFFSET_MPEG_RIGHT];
// Initialize Z80 CPU
Z80.Init(this, Z80IRQCallback);
retainedSamples = 0;
return OKAY;
}
@ -621,24 +622,24 @@ CDSB1::CDSB1(const Util::Config::Node &config)
mpegState = 0;
loopStart = 0;
loopEnd = 0;
DebugLog("Built DSB1 Board\n");
}
CDSB1::~CDSB1(void)
{
{
if (memoryPool != NULL)
{
delete [] memoryPool;
memoryPool = NULL;
}
progROM = NULL;
mpegROM = NULL;
ram = NULL;
mpegL = NULL;
mpegR = NULL;
DebugLog("Destroyed DSB1 Board\n");
}
@ -648,7 +649,7 @@ CDSB1::~CDSB1(void)
******************************************************************************/
// MPEG state machine
enum
enum
{
ST_IDLE = 0,
ST_GOT14, // start/loop addr
@ -672,7 +673,7 @@ enum
ST_GOTB6
};
static const char *stateName[] =
static const char *stateName[] =
{
"idle",
"st_got_14",
@ -717,7 +718,7 @@ void CDSB2::WriteMPEGFIFO(UINT8 byte)
MpegDec::SetMemory(&mpegROM[mpegStart], mpegEnd - mpegStart, false);
mpegState = ST_IDLE;
mpegState = ST_IDLE;
}
else if (byte == 0x84 || byte == 0x85)
@ -766,7 +767,7 @@ void CDSB2::WriteMPEGFIFO(UINT8 byte)
break;
case ST_GOT24:
mpegEnd &= 0x00FFFF;
mpegEnd &= 0x00FFFF;
mpegEnd |= (byte<<16);
mpegState++;
break;
@ -782,12 +783,12 @@ void CDSB2::WriteMPEGFIFO(UINT8 byte)
stereo = StereoMode::Stereo;
mpegState = ST_IDLE;
break;
case ST_GOTA0:
stereo = (byte != 0x00) ? StereoMode::MonoLeft : StereoMode::Stereo;
mpegState = ST_IDLE;
break;
case ST_GOTA4: // dayto2pe plays advertise tune from this state by writing 0x75
mpegState = ST_IDLE;
if (byte == 0x75)
@ -818,7 +819,7 @@ void CDSB2::WriteMPEGFIFO(UINT8 byte)
case ST_GOTB5:
mpegState = ST_IDLE;
break;
/*
* Speaker Volume:
*
@ -868,7 +869,7 @@ UINT8 CDSB2::Read8(UINT32 addr)
if (addr < (128*1024))
return progROM[addr^1];
if (addr == 0xc00001)
if (addr == 0xc00001)
{
return cmdLatch;
}
@ -911,7 +912,7 @@ UINT16 CDSB2::Read16(UINT32 addr)
UINT32 CDSB2::Read32(UINT32 addr)
{
UINT32 hi, lo;
if (addr < (128*1024))
{
hi = *(UINT16 *) &progROM[addr];
@ -937,10 +938,10 @@ void CDSB2::Write8(UINT32 addr, UINT8 data)
ram[addr^1] = data;
return;
}
if (addr == 0xd00001) return;
if (addr == 0xe00003)
if (addr == 0xe00003)
{
WriteMPEGFIFO(data);
return;
@ -983,14 +984,14 @@ void CDSB2::SendCommand(UINT8 data)
fifo[fifoIdxW++] = data;
fifoIdxW &= 127;
//printf("Write FIFO: %02X\n", data);
// Have we caught up to the read pointer?
#ifdef DEBUG
if (fifoIdxW == fifoIdxR)
printf("DSB2 FIFO overflow!\n");
#endif
}
void CDSB2::RunFrame(INT16 *audioL, INT16 *audioR)
{
@ -1005,28 +1006,43 @@ void CDSB2::RunFrame(INT16 *audioL, INT16 *audioR)
M68KSetContext(&M68K);
//printf("DSB2 run frame PC=%06X\n", M68KGetPC());
// While FIFO not empty...
while (fifoIdxR != fifoIdxW)
{
cmdLatch = fifo[fifoIdxR]; // retrieve next command byte
fifoIdxR++;
fifoIdxR &= 127;
fifoIdxR &= 127;
M68KSetIRQ(1); // indicate pending command
//printf("68K INT fired\n");
M68KRun(500);
}
m_totalCyclesElapsed += M68KRun(500);
}
// gm_matthew made the interesting discovery that IRQ2 may in fact be a timer interrupt
// rather than a per-frame interrupt.For Daytona 2 and Sega Rally 2, assuming a value
// of 1KHz fixes music fade outs and some timing issues. It is very likely this is a
// configurable timer and we should be on the look-out for games which appear to use
// different values. It is equally likely that all games share a similar code base and
// use 1KHz as the timer rate.
while (m_totalCyclesElapsed < m_nextFrameEndCycles)
{
if (m_totalCyclesElapsed >= m_nextTimerInterruptCycles)
{
// Fire timer interrupt and schedule next one
M68KSetIRQ(2);
m_nextTimerInterruptCycles = (m_totalCyclesElapsed + k_timerPeriod) - (m_totalCyclesElapsed + k_timerPeriod) % k_timerPeriod;
}
int cyclesToRun = (std::min)(m_nextTimerInterruptCycles, m_nextFrameEndCycles) - m_totalCyclesElapsed;
m_totalCyclesElapsed += M68KRun(cyclesToRun);
}
m_nextFrameEndCycles = (m_totalCyclesElapsed + k_framePeriod) - (m_totalCyclesElapsed + k_framePeriod) % k_framePeriod;
// Per-frame interrupt
M68KSetIRQ(2);
M68KRun(4000000/60);
M68KGetContext(&M68K);
// Decode MPEG for this frame
MpegDec::DecodeAudio(&mpegL[retainedSamples], &mpegR[retainedSamples], 32000 / 60 - retainedSamples + 2);
INT16 *leftChannelSource = nullptr;
INT16 *rightChannelSource = nullptr;
UINT8 volL=0, volR=0;
@ -1062,10 +1078,10 @@ void CDSB2::Reset(void)
MpegDec::Stop();
Resampler.Reset();
retainedSamples = 0;
memset(fifo, 0, sizeof(fifo));
fifoIdxW = fifoIdxR = 0;
mpegState = ST_IDLE;
mpegStart = 0;
mpegEnd = 0;
@ -1073,13 +1089,17 @@ void CDSB2::Reset(void)
volume[0] = 0xFF; // set to max volume in case we miss the volume commands
volume[1] = 0xFF;
stereo = StereoMode::Stereo;
// Even if DSB emulation is disabled, must reset to establish valid Z80 state
M68KSetContext(&M68K);
M68KReset();
//printf("DSB2 PC=%06X\n", M68KGetPC());
M68KGetContext(&M68K);
m_totalCyclesElapsed = 0;
m_nextFrameEndCycles = k_framePeriod;
m_nextTimerInterruptCycles = k_timerPeriod;
DebugLog("DSB2 Reset\n");
}
@ -1087,9 +1107,9 @@ void CDSB2::SaveState(CBlockFile *StateFile)
{
UINT32 playOffset, endOffset;
UINT8 isPlaying;
StateFile->NewBlock("DSB2", __FILE__);
// MPEG playback state
isPlaying = (UINT8)MpegDec::IsLoaded();
playOffset = (UINT32)MpegDec::GetPosition();
@ -1102,7 +1122,7 @@ void CDSB2::SaveState(CBlockFile *StateFile)
StateFile->Write(&usingMPEGEnd, sizeof(usingMPEGEnd));
StateFile->Write(&usingLoopStart, sizeof(usingLoopStart));
StateFile->Write(&usingLoopEnd, sizeof(usingLoopEnd));
// MPEG board state
StateFile->Write(ram, 0x20000);
StateFile->Write(fifo, sizeof(fifo));
@ -1115,11 +1135,11 @@ void CDSB2::SaveState(CBlockFile *StateFile)
StateFile->Write(&playing, sizeof(playing));
StateFile->Write(volume, sizeof(volume));
StateFile->Write(&stereo, sizeof(stereo));
// 68K CPU state
M68KSetContext(&M68K);
M68KSaveState(StateFile, "DSB2 68K");
//DEBUG
//printf("DSB2 PC=%06X\n", M68KGetPC());
//printf("mpegStart=%X, mpegEnd=%X\n", mpegStart, mpegEnd);
@ -1131,13 +1151,13 @@ void CDSB2::LoadState(CBlockFile *StateFile)
{
UINT32 playOffset, endOffset;
UINT8 isPlaying;
if (OKAY != StateFile->FindBlock("DSB2"))
{
ErrorLog("Unable to load Digital Sound Board state. Save state file is corrupt.");
return;
}
StateFile->Read(&isPlaying, sizeof(isPlaying));
StateFile->Read(&playOffset, sizeof(playOffset));
StateFile->Read(&endOffset, sizeof(endOffset));
@ -1145,7 +1165,7 @@ void CDSB2::LoadState(CBlockFile *StateFile)
StateFile->Read(&usingMPEGEnd, sizeof(usingMPEGEnd));
StateFile->Read(&usingLoopStart, sizeof(usingLoopStart));
StateFile->Read(&usingLoopEnd, sizeof(usingLoopEnd));
StateFile->Read(ram, 0x20000);
StateFile->Read(fifo, sizeof(fifo));
StateFile->Read(&fifoIdxR, sizeof(fifoIdxR));
@ -1157,11 +1177,17 @@ void CDSB2::LoadState(CBlockFile *StateFile)
StateFile->Read(&playing, sizeof(playing));
StateFile->Read(volume, sizeof(volume));
StateFile->Read(&stereo, sizeof(stereo));
M68KSetContext(&M68K);
M68KLoadState(StateFile, "DSB2 68K");
M68KGetContext(&M68K);
// Technically these should be saved/restored rather than being reset but that would mean
// the save state format has to be modified and the difference would be imperceptible anyway
m_totalCyclesElapsed = 0;
m_nextFrameEndCycles = k_framePeriod;
m_nextTimerInterruptCycles = k_timerPeriod;
// Restart MPEG audio at the appropriate position
if (isPlaying)
{
@ -1176,13 +1202,13 @@ void CDSB2::LoadState(CBlockFile *StateFile)
else {
MpegDec::Stop();
}
//DEBUG
//printf("DSB2 PC=%06X\n", M68KGetPC());
//printf("mpegStart=%X, mpegEnd=%X\n", mpegStart, mpegEnd);
//printf("usingMPEGStart=%X, usingMPEGEnd=%X\n", usingMPEGStart, usingMPEGEnd);
//printf("usingLoopStart=%X, usingLoopEnd=%X\n", usingLoopStart, usingLoopEnd);
}
// Offsets of memory regions within DSB2's pool
@ -1194,22 +1220,22 @@ void CDSB2::LoadState(CBlockFile *StateFile)
bool CDSB2::Init(const UINT8 *progROMPtr, const UINT8 *mpegROMPtr)
{
float memSizeMB = (float)DSB2_MEMORY_POOL_SIZE/(float)0x100000;
// Receive ROM
progROM = progROMPtr;
mpegROM = mpegROMPtr;
// Allocate memory pool
memoryPool = new(std::nothrow) UINT8[DSB2_MEMORY_POOL_SIZE];
if (NULL == memoryPool)
return ErrorLog("Insufficient memory for DSB2 board (needs %1.1f MB).", memSizeMB);
memset(memoryPool, 0, DSB2_MEMORY_POOL_SIZE);
// Set up memory pointers
ram = &memoryPool[DSB2_OFFSET_RAM];
mpegL = (INT16 *) &memoryPool[DSB2_OFFSET_MPEG_LEFT];
mpegR = (INT16 *) &memoryPool[DSB2_OFFSET_MPEG_RIGHT];
// Initialize 68K CPU
M68KSetContext(&M68K);
M68KInit();
@ -1218,7 +1244,7 @@ bool CDSB2::Init(const UINT8 *progROMPtr, const UINT8 *mpegROMPtr)
M68KGetContext(&M68K);
retainedSamples = 0;
return OKAY;
}
@ -1248,18 +1274,18 @@ CDSB2::CDSB2(const Util::Config::Node &config)
}
CDSB2::~CDSB2(void)
{
{
if (memoryPool != NULL)
{
delete [] memoryPool;
memoryPool = NULL;
}
progROM = NULL;
mpegROM = NULL;
ram = NULL;
mpegL = NULL;
mpegR = NULL;
DebugLog("Destroyed DSB2 Board\n");
}

View file

@ -1,12 +1,13 @@
/**
** Supermodel
** A Sega Model 3 Arcade Emulator.
** Copyright 2011 Bart Trzynadlowski, Nik Henson
** Copyright 2011-2021 Bart Trzynadlowski, Nik Henson, Ian Curtis,
** Harry Tuttle, and Spindizzi
**
** This file is part of Supermodel.
**
** Supermodel 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
** the terms of the GNU General Public License as published by the Free
** Software Foundation, either version 3 of the License, or (at your option)
** any later version.
**
@ -18,10 +19,10 @@
** You should have received a copy of the GNU General Public License along
** with Supermodel. If not, see <http://www.gnu.org/licenses/>.
**/
/*
* DSB.h
*
*
* Header file for the Sega Digital Sound Board (Type 1 and 2) devices. CDSB1
* is an implementation of the Z80-based DSB Type 1, and CDSB2 is the 68K-based
* Type 2 board. Only one may be active at a time because they rely on non-
@ -38,7 +39,7 @@
/******************************************************************************
Resampling
Used internally by the DSB's MPEG code. If this becomes sufficiently generic,
it can be moved to Sound/. Not intended for general use for now.
******************************************************************************/
@ -47,8 +48,8 @@
* CDSBResampler:
*
* Frame-by-frame resampler. Resamples one single frame of audio and maintains
* continuity between frames by copying unprocessed input samples to the
* beginning of the buffer and retaining the internal interpolation state.
* continuity between frames by copying unprocessed input samples to the
* beginning of the buffer and retaining the internal interpolation state.
*
* See DSB.cpp for a detailed description of how this works.
*
@ -56,9 +57,9 @@
* Reset(). Whether the resampler will otherwise behave correctly and stay
* within array bounds has not been verified.
*
* Designed for use at 60 Hz, for input frequencies of 11.025, 22.05, 16, and
* Designed for use at 60 Hz, for input frequencies of 11.025, 22.05, 16, and
* 32 KHz and 44.1 KHz output frequencies. Theoretically, it should be able to
* operate on most output frequencies and input frequencies that are simply
* operate on most output frequencies and input frequencies that are simply
* lower, but it has not been extensively verified.
*/
class CDSBResampler
@ -86,17 +87,17 @@ private:
*
* Abstract base class defining the common interface for both DSB board types.
*/
class CDSB: public IBus
{
public:
/*
* SendCommand(data):
*
* Send a MIDI command to the DSB board.
* Send a MIDI command to the DSB board.
*/
virtual void SendCommand(UINT8 data) = 0;
/*
* RunFrame(audioL, audioR):
*
@ -108,14 +109,14 @@ public:
* audioR Right audio channel.
*/
virtual void RunFrame(INT16 *audioL, INT16 *audioR) = 0;
/*
* Reset(void):
*
* Resets the DSB. Must be called prior to RunFrame().
*/
virtual void Reset(void) = 0;
/*
* SaveState(SaveState):
*
@ -135,7 +136,7 @@ public:
* SaveState Block file to load state information from.
*/
virtual void LoadState(CBlockFile *SaveState) = 0;
/*
* Init(progROMPtr, mpegROMPtr):
*
@ -146,10 +147,10 @@ public:
* mpegROMPtr MPEG data ROM.
*
* Returns:
* OKAY if successful, otherwise FAIL.
* OKAY if successful, otherwise FAIL.
*/
virtual bool Init(const UINT8 *progROMPtr, const UINT8 *mpegROMPtr) = 0;
virtual ~CDSB()
{
}
@ -157,8 +158,8 @@ public:
/******************************************************************************
DSB Classes
DSB Classes
DSB1 and DSB2 hardware. The base class, CDSB, should ideally be dynamically
allocated using one of these. See CDSB for descriptions of member functions.
******************************************************************************/
@ -166,7 +167,7 @@ public:
/*
* CDSB1:
*
* Sega Digital Sound Board Type 1: Z80 plus custom gate array for MPEG
* Sega Digital Sound Board Type 1: Z80 plus custom gate array for MPEG
* decoding.
*/
class CDSB1: public CDSB
@ -177,7 +178,7 @@ public:
void IOWrite8(UINT32 addr, UINT8 data);
UINT8 Read8(UINT32 addr);
void Write8(UINT32 addr, UINT8 data);
// DSB interface (see CDSB definition)
void SendCommand(UINT8 data);
void RunFrame(INT16 *audioL, INT16 *audioR);
@ -185,48 +186,48 @@ public:
void SaveState(CBlockFile *StateFile);
void LoadState(CBlockFile *StateFile);
bool Init(const UINT8 *progROMPtr, const UINT8 *mpegROMPtr);
// Returns a reference to the Z80 CPU
CZ80 *GetZ80(void);
// Constructor and destructor
CDSB1(const Util::Config::Node &config);
~CDSB1(void);
private:
const Util::Config::Node &m_config;
// Resampler
CDSBResampler Resampler;
int retainedSamples; // how many MPEG samples carried over from previous frame
// MPEG decode buffers (48KHz, 1/60th second + 2 extra padding samples)
INT16 *mpegL, *mpegR;
// DSB memory
const UINT8 *progROM; // Z80 program ROM (passed in from parent object)
const UINT8 *mpegROM; // MPEG music ROM
UINT8 *memoryPool; // all memory allocated here
UINT8 *ram; // Z80 RAM
// Command FIFO
UINT8 fifo[128];
int fifoIdxR; // read position
int fifoIdxW; // write position
// MPEG playback variables
int mpegStart;
int mpegEnd;
int mpegState;
int loopStart;
int loopEnd;
// Settings of currently playing stream (may not match the playback register variables above)
UINT32 usingLoopStart; // what was last set by MPEG_SetLoop() or MPEG_PlayMemory()
UINT32 usingLoopEnd;
UINT32 usingMPEGStart; // what was last set by MPEG_PlayMemory()
UINT32 usingMPEGEnd;
// Registers
UINT32 startLatch; // MPEG start address latch
UINT32 endLatch; // MPEG end address latch
@ -234,7 +235,7 @@ private:
UINT8 cmdLatch;
UINT8 volume; // 0x00-0x7F
UINT8 stereo;
// Z80 CPU
CZ80 Z80;
};
@ -254,7 +255,7 @@ public:
void Write8(UINT32 addr, UINT8 data);
void Write16(UINT32 addr, UINT16 data);
void Write32(UINT32 addr, UINT32 data);
// DSB interface (see definition of CDSB)
void SendCommand(UINT8 data);
void RunFrame(INT16 *audioL, INT16 *audioR);
@ -269,20 +270,20 @@ public:
// Constructor and destructor
CDSB2(const Util::Config::Node &config);
~CDSB2(void);
private:
const Util::Config::Node &m_config;
// Private helper functions
void WriteMPEGFIFO(UINT8 byte);
// Resampler
CDSBResampler Resampler;
int retainedSamples; // how many MPEG samples carried over from previous frame
// MPEG decode buffers (48KHz, 1/60th second + 2 extra padding samples)
INT16 *mpegL, *mpegR;
// Stereo mode (do not change values because they are used in save states!)
enum class StereoMode: uint8_t
{
@ -290,7 +291,7 @@ private:
MonoLeft = 1, // mono, using left stream as source data
MonoRight = 2 // mono, using right stream as source data
};
// DSB memory
const UINT8 *progROM; // 68K program ROM (passed in from parent object)
const UINT8 *mpegROM; // MPEG music ROM
@ -301,22 +302,27 @@ private:
UINT8 fifo[128];
int fifoIdxR; // read position
int fifoIdxW; // write position
// Registers
int cmdLatch;
int mpegState;
int mpegStart, mpegEnd, playing;
UINT8 volume[2]; // left, right volume (0x00-0xFF)
StereoMode stereo;
// Settings of currently playing stream (may not match the playback register variables above)
UINT32 usingLoopStart; // what was last set by MPEG_SetLoop() or MPEG_PlayMemory()
UINT32 usingLoopEnd;
UINT32 usingMPEGStart; // what was last set by MPEG_PlayMemory()
UINT32 usingMPEGEnd;
// M68K CPU
M68KCtx M68K;
static const int k_framePeriod = 4000000/60;
static const int k_timerPeriod = 4000000/1000; // 1KHz timer
int m_totalCyclesElapsed;
int m_nextFrameEndCycles;
int m_nextTimerInterruptCycles;
};