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id_sd.cpp
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id_sd.cpp
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//
// ID Engine
// ID_SD.c - Sound Manager for Wolfenstein 3D
// v1.2
// By Jason Blochowiak
//
//
// This module handles dealing with generating sound on the appropriate
// hardware
//
// Depends on: User Mgr (for parm checking)
//
// Globals:
// For User Mgr:
// SoundBlasterPresent - SoundBlaster card present?
// AdLibPresent - AdLib card present?
// SoundMode - What device is used for sound effects
// (Use SM_SetSoundMode() to set)
// MusicMode - What device is used for music
// (Use SM_SetMusicMode() to set)
// DigiMode - What device is used for digitized sound effects
// (Use SM_SetDigiDevice() to set)
//
// For Cache Mgr:
// NeedsDigitized - load digitized sounds?
// NeedsMusic - load music?
//
#include "wl_def.h"
#include <SDL_mixer.h>
#include "dosbox/dbopl.h"
#define ORIGSAMPLERATE 7042
typedef struct
{
char RIFF[4];
longword filelenminus8;
char WAVE[4];
char fmt_[4];
longword formatlen;
word val0x0001;
word channels;
longword samplerate;
longword bytespersec;
word bytespersample;
word bitspersample;
} headchunk;
typedef struct
{
char chunkid[4];
longword chunklength;
} wavechunk;
typedef struct
{
uint32_t startpage;
uint32_t length;
} digiinfo;
static Mix_Chunk *SoundChunks[ STARTMUSIC - STARTDIGISOUNDS];
static byte *SoundBuffers[STARTMUSIC - STARTDIGISOUNDS];
// [DenisBelmondo] backport ecwolf/k1n9_duk3 fixes
SDL_mutex *audioMutex;
globalsoundpos channelSoundPos[MIX_CHANNELS];
// Global variables
boolean AdLibPresent,
SoundBlasterPresent,SBProPresent,
SoundPositioned;
SDMode SoundMode;
SMMode MusicMode;
SDSMode DigiMode;
static byte **SoundTable;
int DigiMap[LASTSOUND];
int DigiChannel[STARTMUSIC - STARTDIGISOUNDS];
// Internal variables
static boolean SD_Started;
static boolean nextsoundpos;
static soundnames SoundNumber;
static soundnames DigiNumber;
static word SoundPriority;
static word DigiPriority;
static int LeftPosition;
static int RightPosition;
word NumDigi;
static digiinfo *DigiList;
static boolean DigiPlaying;
// PC Sound variables
static volatile byte pcLastSample;
static byte * volatile pcSound;
static longword pcLengthLeft;
// AdLib variables
static byte * volatile alSound;
static byte alBlock;
static longword alLengthLeft;
static longword alTimeCount;
static Instrument alZeroInst;
// Sequencer variables
static volatile boolean sqActive;
static word *sqHack;
static word *sqHackPtr;
static int sqHackLen;
static int sqHackSeqLen;
static longword sqHackTime;
DBOPL::Chip oplChip;
static inline bool YM3812Init(int numChips, int clock, int rate)
{
oplChip.Setup(rate);
return false;
}
static inline void YM3812Write(DBOPL::Chip &which, Bit32u reg, Bit8u val)
{
which.WriteReg(reg, val);
}
static inline void YM3812UpdateOne(DBOPL::Chip &which, int16_t *stream, int length)
{
Bit32s buffer[512 * 2];
int i;
// length is at maximum samplesPerMusicTick = param_samplerate / 700
// so 512 is sufficient for a sample rate of 358.4 kHz (default 44.1 kHz)
if(length > 512)
length = 512;
if(which.opl3Active)
{
which.GenerateBlock3(length, buffer);
// GenerateBlock3 generates a number of "length" 32-bit stereo samples
// so we only need to convert them to 16-bit samples
for(i = 0; i < length * 2; i++) // * 2 for left/right channel
{
// Multiply by 4 to match loudness of MAME emulator.
Bit32s sample = buffer[i] << 2;
if(sample > 32767) sample = 32767;
else if(sample < -32768) sample = -32768;
stream[i] = sample;
}
}
else
{
which.GenerateBlock2(length, buffer);
// GenerateBlock3 generates a number of "length" 32-bit mono samples
// so we need to convert them to 32-bit stereo samples
for(i = 0; i < length; i++)
{
// Multiply by 4 to match loudness of MAME emulator.
// Then upconvert to stereo.
Bit32s sample = buffer[i] << 2;
if(sample > 32767) sample = 32767;
else if(sample < -32768) sample = -32768;
stream[i * 2] = stream[i * 2 + 1] = (int16_t) sample;
}
}
}
static void SDL_SoundFinished(void)
{
SoundNumber = (soundnames)0;
SoundPriority = 0;
}
#ifdef NOTYET
void SDL_turnOnPCSpeaker(word timerval);
#pragma aux SDL_turnOnPCSpeaker = \
"mov al,0b6h" \
"out 43h,al" \
"mov al,bl" \
"out 42h,al" \
"mov al,bh" \
"out 42h,al" \
"in al,61h" \
"or al,3" \
"out 61h,al" \
parm [bx] \
modify exact [al]
void SDL_turnOffPCSpeaker();
#pragma aux SDL_turnOffPCSpeaker = \
"in al,61h" \
"and al,0fch" \
"out 61h,al" \
modify exact [al]
void SDL_setPCSpeaker(byte val);
#pragma aux SDL_setPCSpeaker = \
"in al,61h" \
"and al,0fch" \
"or al,ah" \
"out 61h,al" \
parm [ah] \
modify exact [al]
void inline SDL_DoFX()
{
if(pcSound)
{
if(*pcSound!=pcLastSample)
{
pcLastSample=*pcSound;
if(pcLastSample)
SDL_turnOnPCSpeaker(pcLastSample*60);
else
SDL_turnOffPCSpeaker();
}
pcSound++;
pcLengthLeft--;
if(!pcLengthLeft)
{
pcSound=0;
SoundNumber=(soundnames)0;
SoundPriority=0;
SDL_turnOffPCSpeaker();
}
}
// [adlib sound stuff removed...]
}
static void SDL_DigitizedDoneInIRQ(void);
void inline SDL_DoFast()
{
count_fx++;
if(count_fx>=5)
{
count_fx=0;
SDL_DoFX();
count_time++;
if(count_time>=2)
{
TimeCount++;
count_time=0;
}
}
// [adlib music and soundsource stuff removed...]
TimerCount+=TimerDivisor;
if(*((word *)&TimerCount+1))
{
*((word *)&TimerCount+1)=0;
t0OldService();
}
else
{
outp(0x20,0x20);
}
}
// Timer 0 ISR for 7000Hz interrupts
void __interrupt SDL_t0ExtremeAsmService(void)
{
if(pcindicate)
{
if(pcSound)
{
SDL_setPCSpeaker(((*pcSound++)&0x80)>>6);
pcLengthLeft--;
if(!pcLengthLeft)
{
pcSound=0;
SDL_turnOffPCSpeaker();
SDL_DigitizedDoneInIRQ();
}
}
}
extreme++;
if(extreme>=10)
{
extreme=0;
SDL_DoFast();
}
else
outp(0x20,0x20);
}
// Timer 0 ISR for 700Hz interrupts
void __interrupt SDL_t0FastAsmService(void)
{
SDL_DoFast();
}
// Timer 0 ISR for 140Hz interrupts
void __interrupt SDL_t0SlowAsmService(void)
{
count_time++;
if(count_time>=2)
{
TimeCount++;
count_time=0;
}
SDL_DoFX();
TimerCount+=TimerDivisor;
if(*((word *)&TimerCount+1))
{
*((word *)&TimerCount+1)=0;
t0OldService();
}
else
outp(0x20,0x20);
}
void SDL_IndicatePC(boolean ind)
{
pcindicate=ind;
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_SetTimer0() - Sets system timer 0 to the specified speed
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_SetTimer0(word speed)
{
#ifndef TPROF // If using Borland's profiling, don't screw with the timer
// _asm pushfd
_asm cli
outp(0x43,0x36); // Change timer 0
outp(0x40,(byte)speed);
outp(0x40,speed >> 8);
// Kludge to handle special case for digitized PC sounds
if (TimerDivisor == (1192030 / (TickBase * 100)))
TimerDivisor = (1192030 / (TickBase * 10));
else
TimerDivisor = speed;
// _asm popfd
_asm sti
#else
TimerDivisor = 0x10000;
#endif
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_SetIntsPerSec() - Uses SDL_SetTimer0() to set the number of
// interrupts generated by system timer 0 per second
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_SetIntsPerSec(word ints)
{
TimerRate = ints;
SDL_SetTimer0(1192030 / ints);
}
static void
SDL_SetTimerSpeed(void)
{
word rate;
void (_interrupt *isr)(void);
if ((DigiMode == sds_PC) && DigiPlaying)
{
rate = TickBase * 100;
isr = SDL_t0ExtremeAsmService;
}
else if ((MusicMode == smm_AdLib) || ((DigiMode == sds_SoundSource) && DigiPlaying) )
{
rate = TickBase * 10;
isr = SDL_t0FastAsmService;
}
else
{
rate = TickBase * 2;
isr = SDL_t0SlowAsmService;
}
if (rate != TimerRate)
{
_dos_setvect(8,isr);
SDL_SetIntsPerSec(rate);
TimerRate = rate;
}
}
//
// PC Sound code
//
///////////////////////////////////////////////////////////////////////////
//
// SDL_PCPlaySample() - Plays the specified sample on the PC speaker
//
///////////////////////////////////////////////////////////////////////////
#ifdef _MUSE_
void
#else
static void
#endif
SDL_PCPlaySample(byte *data,longword len,boolean inIRQ)
{
if(!inIRQ)
{
// _asm pushfd
_asm cli
}
SDL_IndicatePC(true);
pcLengthLeft = len;
pcSound = (volatile byte *)data;
if(!inIRQ)
{
// _asm popfd
_asm sti
}
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_PCStopSample() - Stops a sample playing on the PC speaker
//
///////////////////////////////////////////////////////////////////////////
#ifdef _MUSE_
void
#else
static void
#endif
SDL_PCStopSampleInIRQ(void)
{
pcSound = 0;
SDL_IndicatePC(false);
_asm in al,0x61 // Turn the speaker off
_asm and al,0xfd // ~2
_asm out 0x61,al
}
#endif
///////////////////////////////////////////////////////////////////////////
//
// SDL_PCPlaySound() - Plays the specified sound on the PC speaker
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_PCPlaySound(PCSound *sound)
{
/*
// _asm pushfd
_asm cli
*/
pcLastSample = -1;
pcLengthLeft = sound->common.length;
pcSound = sound->data;
/*
// _asm popfd
_asm sti
*/
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_PCStopSound() - Stops the current sound playing on the PC Speaker
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_PCStopSound(void)
{
/*
// _asm pushfd
_asm cli
*/
pcSound = 0;
/*
_asm in al,0x61 // Turn the speaker off
_asm and al,0xfd // ~2
_asm out 0x61,al
// _asm popfd
_asm sti
*/
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_ShutPC() - Turns off the pc speaker
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_ShutPC(void)
{
/*
// _asm pushfd
_asm cli
*/
pcSound = 0;
/*
_asm in al,0x61 // Turn the speaker & gate off
_asm and al,0xfc // ~3
_asm out 0x61,al
// _asm popfd
_asm sti
*/
}
// Adapted from Chocolate Doom (chocolate-doom/pcsound/pcsound_sdl.c)
#define SQUARE_WAVE_AMP 0x2000
static void SDL_PCMixCallback(void *udata, Uint8 *stream, int len)
{
static int current_remaining = 0;
static int current_freq = 0;
static int phase_offset = 0;
void *streamp = stream;
Sint16 *leftptr;
Sint16 *rightptr;
Sint16 this_value;
int oldfreq;
int i;
int nsamples;
// Number of samples is quadrupled, because of 16-bit and stereo
nsamples = len / 4;
leftptr = (Sint16 *) streamp;
rightptr = ((Sint16 *) streamp) + 1;
// Fill the output buffer
for (i=0; i<nsamples; ++i)
{
// Has this sound expired? If so, retrieve the next frequency
while (current_remaining == 0)
{
oldfreq = current_freq;
// Get the next frequency to play
if(pcSound)
{
if(*pcSound!=pcLastSample)
{
pcLastSample=*pcSound;
if(pcLastSample)
// The PC PIC counts down at 1.193180MHz
// So pwm_freq = counter_freq / reload_value
// reload_value = pcLastSample * 60 (see SDL_DoFX)
current_freq = 1193180 / (pcLastSample * 60);
else
current_freq = 0;
// The PC speaker sample rate is 140Hz (see SDL_t0SlowAsmService)
current_remaining = param_samplerate / 140;
}
pcSound++;
pcLengthLeft--;
if(!pcLengthLeft)
{
pcSound=0;
SoundNumber=(soundnames)0;
SoundPriority=0;
}
}
else
{
current_freq = 0;
current_remaining = 1;
}
if (current_freq != 0)
{
// Adjust phase to match to the new frequency.
// This gives us a smooth transition between different tones,
// with no impulse changes.
phase_offset = (phase_offset * oldfreq) / current_freq;
}
}
// Set the value for this sample.
if (current_freq == 0)
{
// Silence
this_value = 0;
}
else
{
int frac;
// Determine whether we are at a peak or trough in the current
// sound. Multiply by 2 so that frac % 2 will give 0 or 1
// depending on whether we are at a peak or trough.
frac = (phase_offset * current_freq * 2) / param_samplerate;
if ((frac % 2) == 0)
{
this_value = SQUARE_WAVE_AMP;
}
else
{
this_value = -SQUARE_WAVE_AMP;
}
++phase_offset;
}
--current_remaining;
// Use the same value for the left and right channels.
*leftptr += this_value;
*rightptr += this_value;
leftptr += 2;
rightptr += 2;
}
}
void
SD_StopDigitized(void)
{
DigiPlaying = false;
DigiNumber = (soundnames) 0;
DigiPriority = 0;
SoundPositioned = false;
if ((DigiMode == sds_PC) && (SoundMode == sdm_PC))
SDL_SoundFinished();
switch (DigiMode)
{
case sds_PC:
// SDL_PCStopSampleInIRQ();
SDL_PCStopSound();
break;
case sds_SoundBlaster:
// SDL_SBStopSampleInIRQ();
Mix_HaltChannel(-1);
break;
default:
break;
}
}
int SD_GetChannelForDigi(int which)
{
if(DigiChannel[which] != -1) return DigiChannel[which];
int channel = Mix_GroupAvailable(1);
if(channel == -1) channel = Mix_GroupOldest(1);
if(channel == -1) // All sounds stopped in the meantime?
return Mix_GroupAvailable(1);
return channel;
}
void SD_SetPosition(int channel, int leftpos, int rightpos)
{
if((leftpos < 0) || (leftpos > 15) || (rightpos < 0) || (rightpos > 15)
|| ((leftpos == 15) && (rightpos == 15)))
Quit("SD_SetPosition: Illegal position");
switch (DigiMode)
{
case sds_SoundBlaster:
// SDL_PositionSBP(leftpos,rightpos);
Mix_SetPanning(channel, ((15 - leftpos) << 4) + 15,
((15 - rightpos) << 4) + 15);
break;
default:
break;
}
}
Sint16 GetSample(float csample, byte *samples, int size)
{
float s0=0, s1=0, s2=0;
int cursample = (int) csample;
float sf = csample - (float) cursample;
if(cursample-1 >= 0) s0 = (float) (samples[cursample-1] - 128);
s1 = (float) (samples[cursample] - 128);
if(cursample+1 < size) s2 = (float) (samples[cursample+1] - 128);
float val = s0*sf*(sf-1)/2 - s1*(sf*sf-1) + s2*(sf+1)*sf/2;
int32_t intval = (int32_t) (val * 256);
if(intval < -32768) intval = -32768;
else if(intval > 32767) intval = 32767;
return (Sint16) intval;
}
void SD_PrepareSound(int which)
{
if(DigiList == NULL)
Quit("SD_PrepareSound(%i): DigiList not initialized!\n", which);
int page = DigiList[which].startpage;
int size = DigiList[which].length;
byte *origsamples = PM_GetSound(page);
if(origsamples + size >= PM_GetEnd())
Quit("SD_PrepareSound(%i): Sound reaches out of page file!\n", which);
longword destsamples = (int) ((float) size * (float) param_samplerate
/ (float) ORIGSAMPLERATE);
byte *wavebuffer = (byte *) malloc(sizeof(headchunk) + sizeof(wavechunk)
+ destsamples * 2); // dest are 16-bit samples
if(wavebuffer == NULL)
Quit("Unable to allocate wave buffer for sound %i!\n", which);
headchunk head = {{'R','I','F','F'}, 0, {'W','A','V','E'},
{'f','m','t',' '}, 0x10, 0x0001, 1, param_samplerate, param_samplerate*2, 2, 16};
wavechunk dhead = {{'d', 'a', 't', 'a'}, destsamples*2};
head.filelenminus8 = sizeof(head) + destsamples*2; // (sizeof(dhead)-8 = 0)
memcpy(wavebuffer, &head, sizeof(head));
memcpy(wavebuffer+sizeof(head), &dhead, sizeof(dhead));
// alignment is correct, as wavebuffer comes from malloc
// and sizeof(headchunk) % 4 == 0 and sizeof(wavechunk) % 4 == 0
Sint16 *newsamples = (Sint16 *)(void *) (wavebuffer + sizeof(headchunk)
+ sizeof(wavechunk));
float cursample = 0.F;
float samplestep = (float) ORIGSAMPLERATE / (float) param_samplerate;
for(longword i=0; i<destsamples; i++, cursample+=samplestep)
{
newsamples[i] = GetSample((float)size * (float)i / (float)destsamples,
origsamples, size);
}
SoundBuffers[which] = wavebuffer;
SoundChunks[which] = Mix_LoadWAV_RW(SDL_RWFromMem(wavebuffer,
sizeof(headchunk) + sizeof(wavechunk) + destsamples * 2), 1);
}
int SD_PlayDigitized(word which,int leftpos,int rightpos)
{
if (!DigiMode)
return 0;
if (which >= NumDigi)
Quit("SD_PlayDigitized: bad sound number %i", which);
int channel = SD_GetChannelForDigi(which);
SD_SetPosition(channel, leftpos,rightpos);
DigiPlaying = true;
Mix_Chunk *sample = SoundChunks[which];
if(sample == NULL)
{
printf("SoundChunks[%i] is NULL!\n", which);
return 0;
}
if(Mix_PlayChannel(channel, sample, 0) == -1)
{
printf("Unable to play sound: %s\n", Mix_GetError());
return 0;
}
return channel;
}
void SD_ChannelFinished(int channel)
{
channelSoundPos[channel].valid = 0;
}
void
SD_SetDigiDevice(SDSMode mode)
{
boolean devicenotpresent;
if (mode == DigiMode)
return;
SD_StopDigitized();
devicenotpresent = false;
switch (mode)
{
case sds_SoundBlaster:
if (!SoundBlasterPresent)
devicenotpresent = true;
break;
default:
break;
}
if (!devicenotpresent)
{
DigiMode = mode;
#ifdef NOTYET
SDL_SetTimerSpeed();
#endif
}
}
void
SDL_SetupDigi(void)
{
// Correct padding enforced by PM_Startup()
word *soundInfoPage = (word *) (void *) PM_GetPage(ChunksInFile-1);
NumDigi = (word) PM_GetPageSize(ChunksInFile - 1) / 4;
DigiList = (digiinfo *) malloc(NumDigi * sizeof(digiinfo));
int i;
for(i = 0; i < NumDigi; i++)
{
// Calculate the size of the digi from the sizes of the pages between
// the start page and the start page of the next sound
DigiList[i].startpage = soundInfoPage[i * 2];
if((int) DigiList[i].startpage >= ChunksInFile - 1)
{
NumDigi = i;
break;
}
int lastPage;
if(i < NumDigi - 1)
{
lastPage = soundInfoPage[i * 2 + 2];
if(lastPage == 0 || lastPage + PMSoundStart > ChunksInFile - 1) lastPage = ChunksInFile - 1;
else lastPage += PMSoundStart;
}
else lastPage = ChunksInFile - 1;
int size = 0;
for(int page = PMSoundStart + DigiList[i].startpage; page < lastPage; page++)
size += PM_GetPageSize(page);
// Don't include padding of sound info page, if padding was added
if(lastPage == ChunksInFile - 1 && PMSoundInfoPagePadded) size--;
// Patch lower 16-bit of size with size from sound info page.
// The original VSWAP contains padding which is included in the page size,
// but not included in the 16-bit size. So we use the more precise value.
if((size & 0xffff0000) != 0 && (size & 0xffff) < soundInfoPage[i * 2 + 1])
size -= 0x10000;
size = (size & 0xffff0000) | soundInfoPage[i * 2 + 1];
DigiList[i].length = size;
}
for(i = 0; i < LASTSOUND; i++)
{
DigiMap[i] = -1;
DigiChannel[i] = -1;
}
}
// AdLib Code
///////////////////////////////////////////////////////////////////////////
//
// SDL_ALStopSound() - Turns off any sound effects playing through the
// AdLib card
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_ALStopSound(void)
{
// [DenisBelmondo] backport ecwolf/k1n9_duk3 fixes
SDL_LockMutex(audioMutex);
alSound = 0;
alOut(alFreqH + 0, 0);
SDL_UnlockMutex(audioMutex);
}
static void
SDL_AlSetFXInst(Instrument *inst)
{
byte c,m;
m = 0; // modulator cell for channel 0
c = 3; // carrier cell for channel 0
alOut(m + alChar,inst->mChar);
alOut(m + alScale,inst->mScale);
alOut(m + alAttack,inst->mAttack);
alOut(m + alSus,inst->mSus);
alOut(m + alWave,inst->mWave);
alOut(c + alChar,inst->cChar);
alOut(c + alScale,inst->cScale);
alOut(c + alAttack,inst->cAttack);
alOut(c + alSus,inst->cSus);
alOut(c + alWave,inst->cWave);
// Note: Switch commenting on these lines for old MUSE compatibility
// alOutInIRQ(alFeedCon,inst->nConn);
alOut(alFeedCon,0);
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_ALPlaySound() - Plays the specified sound on the AdLib card
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_ALPlaySound(AdLibSound *sound)
{
Instrument *inst;
byte *data;
SDL_ALStopSound();
// [DenisBelmondo] backport ecwolf/k1n9_duk3 fixes
SDL_LockMutex(audioMutex);
alLengthLeft = sound->common.length;
data = sound->data;
alBlock = ((sound->block & 7) << 2) | 0x20;
inst = &sound->inst;
if (!(inst->mSus | inst->cSus))
{
Quit("SDL_ALPlaySound() - Bad instrument");
}
SDL_AlSetFXInst(inst);
alSound = (byte *)data;
SDL_UnlockMutex(audioMutex);
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_ShutAL() - Shuts down the AdLib card for sound effects
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_ShutAL(void)
{
// [DenisBelmondo] backport ecwolf/k1n9_duk3 fixes
SDL_LockMutex(audioMutex);
alSound = 0;
alOut(alEffects,0);
alOut(alFreqH + 0,0);
SDL_AlSetFXInst(&alZeroInst);
SDL_UnlockMutex(audioMutex);
}
///////////////////////////////////////////////////////////////////////////
//
// SDL_StartAL() - Starts up the AdLib card for sound effects
//
///////////////////////////////////////////////////////////////////////////
static void
SDL_StartAL(void)
{
alOut(alEffects, 0);
SDL_AlSetFXInst(&alZeroInst);
}
////////////////////////////////////////////////////////////////////////////
//
// SDL_ShutDevice() - turns off whatever device was being used for sound fx
//
////////////////////////////////////////////////////////////////////////////
static void
SDL_ShutDevice(void)
{
switch (SoundMode)
{