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Updated modplay

CQTexperiment
Chris Moeller 2015-11-12 21:35:42 -08:00
parent ce2303ef24
commit b8763c6cad
4 changed files with 721 additions and 633 deletions
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51
Frameworks/modplay/modplay/ft2play.c
View File

@ -2167,7 +2167,8 @@ static void MainPlayer(PLAYER *p) /* periodically called from mixer */
if (oldSongPos != p->Song.SongPos)
{
for (size_t i = 0; i < playedRowsCount; ++i)
size_t i;
for (i = 0; i < playedRowsCount; ++i)
bit_array_set(p->playedRows, oldSongPos * 1024 + p->playedRowsPatLoop[i]);
memset(p->playedRowsPatLoop, 0xFF, playedRowsCount * 2);
playedRowsCount = 0;
@ -3032,7 +3033,7 @@ static inline void mix8b(PLAYER *p, uint32_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler) ||
while (interpolating > 0 && (resampler_get_free_count(resampler) ||
!resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, sampleData[samplePosition], 8);
@ -3071,11 +3072,18 @@ static inline void mix8b(PLAYER *p, uint32_t ch, uint32_t samples)
}
else if ((samplePosition < 0) || (samplePosition >= sampleLength))
{
interpolating = 0;
interpolating = -resampler_get_padding_size();
break;
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler) ||
!resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, 0, 8);
++interpolating;
}
v->samplePosition = samplePosition;
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
@ -3196,7 +3204,7 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler[0]) ||
while (interpolating > 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
@ -3237,11 +3245,20 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
}
else if ((samplePosition < 0) || (samplePosition >= sampleLength))
{
interpolating = 0;
interpolating = -resampler_get_padding_size();
break;
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
resampler_write_sample_fixed(resampler[0], 0, 8);
resampler_write_sample_fixed(resampler[1], 0, 8);
++interpolating;
}
v->samplePosition = samplePosition;
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
@ -3366,7 +3383,7 @@ static inline void mix16b(PLAYER *p, uint32_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler) ||
while (interpolating > 0 && (resampler_get_free_count(resampler) ||
!resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, sampleData[samplePosition], 16);
@ -3405,11 +3422,18 @@ static inline void mix16b(PLAYER *p, uint32_t ch, uint32_t samples)
}
else if ((samplePosition < 0) || (samplePosition >= sampleLength))
{
interpolating = 0;
interpolating = -resampler_get_padding_size();
break;
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler) ||
!resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, 0, 16);
++interpolating;
}
v->samplePosition = samplePosition;
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
@ -3530,7 +3554,7 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler[0]) ||
while (interpolating > 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
@ -3571,11 +3595,20 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
}
else if ((samplePosition < 0) || (samplePosition >= sampleLength))
{
interpolating = 0;
interpolating = -resampler_get_padding_size();
break;
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
resampler_write_sample_fixed(resampler[0], 0, 16);
resampler_write_sample_fixed(resampler[1], 0, 16);
++interpolating;
}
v->samplePosition = samplePosition;
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;

5
Frameworks/modplay/modplay/resampler.c
View File

@ -303,6 +303,11 @@ static int resampler_output_delay(resampler *r)
}
}
int resampler_get_padding_size()
{
return SINC_WIDTH - 1;
}
int resampler_ready(void *_r)
{
resampler * r = ( resampler * ) _r;

3
Frameworks/modplay/modplay/resampler.h
View File

@ -3,7 +3,6 @@
#define RESAMPLER_DECORATE modplay
// Ugglay
#ifdef RESAMPLER_DECORATE
#define PASTE(a,b) a ## b
#define EVALUATE(a,b) PASTE(a,b)
@ -14,6 +13,7 @@
#define resampler_dup_inplace EVALUATE(RESAMPLER_DECORATE,_resampler_dup_inplace)
#define resampler_set_quality EVALUATE(RESAMPLER_DECORATE,_resampler_set_quality)
#define resampler_get_free_count EVALUATE(RESAMPLER_DECORATE,_resampler_get_free_count)
#define resampler_get_padding_size EVALUATE(RESAMPLER_DECORATE,_resampler_get_padding_size)
#define resampler_write_sample EVALUATE(RESAMPLER_DECORATE,_resampler_write_sample)
#define resampler_write_sample_fixed EVALUATE(RESAMPLER_DECORATE,_resampler_write_sample_fixed)
#define resampler_set_rate EVALUATE(RESAMPLER_DECORATE,_resampler_set_rate)
@ -47,6 +47,7 @@ enum
void resampler_set_quality(void *, int quality);
int resampler_get_free_count(void *);
int resampler_get_padding_size();
void resampler_write_sample(void *, short sample);
void resampler_write_sample_fixed(void *, int sample, unsigned char depth);
void resampler_set_rate( void *, double new_factor );

331
Frameworks/modplay/modplay/st3play.c
View File

@ -12,7 +12,7 @@
** - Added S9E/S9F (non-ST3, play sample backwards/forwards)
** - Fixed a bug in setspd() in Amiga limit mode
** - Proper tracker handling for non-ST3 effects
** - Panbrello (Yxy) didn't set the panning at all (heh)
** - Panbrello (Yxy) didn't set the panning at all
** - Decodes ADPCM samples at load time instead of play time
** - Mxx (set cannel volume) didn't work correctly
**
@ -199,11 +199,9 @@ typedef struct
#ifdef USE_VOL_RAMP
VOICE voice[32 * 2];
void *resampler[64 * 2];
#else
VOICE voice[32];
void *resampler[64];
#endif
@ -213,7 +211,7 @@ typedef struct
const uint8_t *fmPatchTable[9];
uint8_t fmLastB0[9];
int8_t ** adpcmSamples;
int8_t **adpcmSamples;
float f_outputFreq;
float f_masterVolume;
@ -546,7 +544,7 @@ void * st3play_Alloc(uint32_t outputFreq, int8_t interpolation, int8_t ramp_styl
#ifdef USE_VOL_RAMP
setRampStyle(p, ramp_style);
#endif
setSamplesPerFrame(p, ((outputFreq * 5UL) / 2 / 125));
setSamplesPerFrame(p, ((outputFreq * 5) / 2 / 125));
return (p);
}
@ -645,7 +643,7 @@ static void settempo(PLAYER *p, uint16_t val)
if (val > 32)
{
p->tempo = val;
setSamplesPerFrame(p, ((p->outputFreq * 5UL) / 2) / p->tempo);
setSamplesPerFrame(p, ((p->outputFreq * 5) / 2) / p->tempo);
}
}
@ -653,8 +651,8 @@ static void st3play_AdlibHertzTouch(PLAYER *p, uint8_t ch, int32_t Hertz, uint8_
{
int32_t Oct;
for (Oct = 0; Hertz > 0x1FF; Oct++)
Hertz >>= 1;
for (Oct = 0; Hertz >= 512; Oct++)
Hertz /= 2;
Chip_WriteReg(p->fmChip, 0xA0 + ch, Hertz & 0xFF);
@ -764,7 +762,7 @@ static inline int16_t stnote2herz(PLAYER *p, uint8_t note)
if (note == 254) return (0);
tmpnote = note & 0x0F;
tmpocta = note >> 0x04;
tmpocta = note >> 4;
// ST3 doesn't do this, but do it for safety
if (tmpnote > 11) tmpnote = 11;
@ -778,11 +776,11 @@ static inline int16_t stnote2herz(PLAYER *p, uint8_t note)
static inline int32_t scalec2spd(PLAYER *p, uint8_t ch, int32_t spd)
{
spd *= 8363UL;
spd *= 8363;
if (p->tracker == SCREAM_TRACKER)
{
if ((spd >> 16) > p->chn[ch].ac2spd)
if ((spd / 65536) > p->chn[ch].ac2spd)
return (32767);
}
@ -812,7 +810,7 @@ static inline int32_t roundspd(PLAYER *p, uint8_t ch, int32_t spd)
if (p->tracker == SCREAM_TRACKER)
{
if ((newspd >> 16) >= 8363)
if ((newspd / 65536) >= 8363)
return (spd);
}
@ -820,11 +818,11 @@ static inline int32_t roundspd(PLAYER *p, uint8_t ch, int32_t spd)
// find octave
octa = 0;
lastspd = ((1712 * 8) + notespd[11]) >> 1;
lastspd = ((1712 * 8) + (907 * 16)) / 2;;
while (newspd < lastspd)
{
octa++;
lastspd >>= 1;
lastspd /= 2;
}
// find note
@ -834,7 +832,7 @@ static inline int32_t roundspd(PLAYER *p, uint8_t ch, int32_t spd)
if (p->tracker == SCREAM_TRACKER)
lastspd = 32767;
else
lastspd = 32767 * 2; // Might be wrong? Probably not
lastspd = 32767 * 2;
while (newnote < 11)
{
@ -851,11 +849,11 @@ static inline int32_t roundspd(PLAYER *p, uint8_t ch, int32_t spd)
}
// get new speed from new note
newspd = (uint32_t)(stnote2herz(p, (octa << 4) | (lastnote & 0x0F))) * 8363;
newspd = (stnote2herz(p, (octa << 4) | (lastnote & 0x0F))) * 8363;
if (p->tracker == SCREAM_TRACKER)
{
if ((newspd >> 16) >= p->chn[ch].ac2spd)
if ((newspd / 65536) >= p->chn[ch].ac2spd)
return (spd);
}
@ -867,14 +865,14 @@ static inline int32_t roundspd(PLAYER *p, uint8_t ch, int32_t spd)
static int16_t neworder(PLAYER *p)
{
skip:
newOrderSkip:
p->np_ord++;
if ((p->mseg[0x60+(p->np_ord-1)]==255)||(p->np_ord>get_le16(&p->mseg[0x20]))) // end
if ((p->mseg[0x60 + (p->np_ord - 1)] == 255) || (p->np_ord > get_le16(&p->mseg[0x20]))) // end
p->np_ord = 1;
if (p->mseg[0x60 + (p->np_ord - 1)] == 254) // skip
goto skip; // avoid recursive calling
goto newOrderSkip; // avoid recursive calling
p->np_pat = (int16_t)(p->mseg[0x60 + (p->np_ord - 1)]);
p->np_patoff = -1; // force reseek
@ -896,7 +894,7 @@ static inline void seekpat(PLAYER *p)
if (p->np_patoff == -1) // seek must be done
{
p->np_patseg = (uint32_t)(get_le16(&p->mseg[p->patternadd + (p->np_pat << 1)])) << 4;
p->np_patseg = get_le16(&p->mseg[p->patternadd + (p->np_pat * 2)]) * 16;
if (p->np_patseg)
{
j = 2; // skip packed pat len flag
@ -1031,10 +1029,10 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
if (p->chn[ch].ins <= get_le16(&p->mseg[0x22])) // added for safety reasons
{
insdat = &p->mseg[(uint32_t)(get_le16(&p->mseg[p->instrumentadd + ((p->chn[ch].ins - 1) << 1)])) << 4];
insdat = &p->mseg[get_le16(&p->mseg[p->instrumentadd + ((p->chn[ch].ins - 1) * 2)]) * 16];
if (insdat[0])
{
if (insdat[0] == 1 && adlibChannel >= 9)
if ((insdat[0] == 1) && (adlibChannel >= 9))
{
p->chn[ch].ac2spd = get_le32(&insdat[0x20]);
@ -1045,12 +1043,13 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
}
p->chn[ch].avol = (int8_t)(insdat[0x1C]);
if (p->chn[ch].avol < 0) p->chn[ch].avol = 0;
if (p->chn[ch].avol > 63) p->chn[ch].avol = 63;
else if (p->chn[ch].avol > 63) p->chn[ch].avol = 63;
p->chn[ch].aorgvol = p->chn[ch].avol;
insoffs = (uint32_t)(((uint32_t)(insdat[0x0D])<<16)|((uint16_t)(insdat[0x0F])<<8)|insdat[0x0E])<<4;
insoffs = ((insdat[0x0D] << 16) | (insdat[0x0F] << 8) | insdat[0x0E]) * 16;
if (insoffs > p->mseg_len)
insoffs = p->mseg_len;
@ -1100,7 +1099,7 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
p->chn[ch].surround = 0;
voiceSetSurround(p, ch, 0);
p->chn[ch].apanpos = (p->chn[ch].vol - 128) << 2;
p->chn[ch].apanpos = (p->chn[ch].vol - 128) * 4;
setpan(p, ch);
}
}
@ -1137,7 +1136,7 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
}
else if (insdat[0] == 2 && adlibChannel < 9)
{
p->chn[ch].ac2spd = 8363 * 164 / 249;
p->chn[ch].ac2spd = (8363 * 164) / 249;
p->chn[ch].avol = (int8_t)(insdat[0x1C]);
p->chn[ch].aorgvol = p->chn[ch].avol;
@ -1164,7 +1163,7 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
}
else
{
p->chn[ch].astartoffset = (uint16_t)(p->chn[ch].info) << 8;
p->chn[ch].astartoffset = 256 * p->chn[ch].info;
p->chn[ch].astartoffset00 = p->chn[ch].astartoffset;
}
}
@ -1250,7 +1249,7 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
p->chn[ch].surround = 0;
voiceSetSurround(p, ch, 0);
p->chn[ch].apanpos = (p->chn[ch].vol - 128) << 2;
p->chn[ch].apanpos = (p->chn[ch].vol - 128) * 4;
setpan(p, ch);
}
}
@ -1268,7 +1267,7 @@ static inline void donewnote(PLAYER *p, uint8_t ch, int8_t notedelayflag)
{
p->lastachannelused = ch + 1;
// hackish fix, fixes call_me_an_angel.s3m crash
// sanity fix
if (p->lastachannelused > 31) p->lastachannelused = 31;
}
@ -1305,7 +1304,7 @@ static inline void donotes(PLAYER *p)
ch = getnote(p);
if (ch == 255) break; // end of row/channels
if ((p->mseg[0x40 + ch] & 0x7F) <= 15 + 9)
if ((p->mseg[0x40 + ch] & 0x7F) <= (15 + 9))
donewnote(p, ch, 0);
}
}
@ -1328,9 +1327,6 @@ static inline void docmd1(PLAYER *p)
if (p->chn[i].cmd == ('D' - 64))
{
// THEORY: I think this fix is related to
// AdLib channels...
// fix retrig if Dxy
p->chn[i].atrigcnt = 0;
@ -1420,9 +1416,10 @@ static inline void docmd2(PLAYER *p)
void dorow(PLAYER *p) // periodically called from mixer
{
int32_t offset, bit;
int8_t offset;
int8_t bit;
p->patmusicrand = (uint16_t)(((uint32_t)(p->patmusicrand) * 0xCDEF) >> 16) + 0x1727;
p->patmusicrand = (((p->patmusicrand * 0xCDEF) >> 16) + 0x1727) & 0x0000FFFF;
if (!p->musiccount)
{
@ -1497,22 +1494,23 @@ static inline int8_t get_adpcm_sample(const int8_t *sampleDictionary, const uint
{
uint8_t byte;
byte = sampleData[samplePosition >> 1];
byte = (samplePosition & 1) ? byte >> 4 : byte & 15;
return *lastDelta += sampleDictionary[byte];
byte = sampleData[samplePosition / 2];
byte = (samplePosition & 1) ? (byte >> 4) : (byte & 0x0F);
return (*(lastDelta) += sampleDictionary[byte]);
}
static inline void decode_adpcm(const uint8_t * sampleData, int8_t * decodedSampleData, int32_t sampleLength)
static inline void decode_adpcm(const uint8_t *sampleData, int8_t *decodedSampleData, int32_t sampleLength)
{
int i;
int8_t lastDelta, sample;
const int8_t * sampleDictionary;
int32_t i;
int8_t lastDelta;
int8_t sample;
const int8_t *sampleDictionary;
sampleDictionary = (const int8_t *)(sampleData);
sampleData += 16;
lastDelta = 0;
for (i = 0; i < sampleLength; ++i)
{
sample = get_adpcm_sample(sampleDictionary, sampleData, i, &lastDelta);
@ -1596,8 +1594,8 @@ static void loadheaderparms(PLAYER *p)
insnum = get_le16(&p->mseg[0x22]);
for (i = 0; i < insnum; ++i)
{
insdat = &p->mseg[get_le16(&p->mseg[p->instrumentadd + (i << 1)]) << 4];
insoff = (uint32_t)(((uint32_t)(insdat[0x0D])<<16)|((uint16_t)(insdat[0x0F])<<8)|insdat[0x0E])<<4;
insdat = &p->mseg[get_le16(&p->mseg[p->instrumentadd + (i * 2)]) * 16];
insoff = ((insdat[0x0D] << 16) | (insdat[0x0F] << 8) | insdat[0x0E]) * 16;
if (insoff && (insdat[0] == 1)) // PCM
{
@ -1610,6 +1608,7 @@ static void loadheaderparms(PLAYER *p)
{
if (p->adpcmSamples == NULL)
p->adpcmSamples = (int8_t **)(calloc(sizeof(int8_t *), insnum));
if (p->adpcmSamples)
{
p->adpcmSamples[i] = (int8_t *)(calloc(1, inslen));
@ -1617,28 +1616,32 @@ static void loadheaderparms(PLAYER *p)
{
if ((insoff + (16 + (inslen + 1) / 2)) > p->mseg_len)
inslen = ((p->mseg_len - insoff) - 16) * 2;
if (inslen >= 1)
decode_adpcm(&p->mseg[insoff], p->adpcmSamples[i], inslen);
}
}
continue;
}
if (insdat[0x1F] & 2) inslen <<= 1; // stereo
if (insdat[0x1F] & 2) inslen *= 2; // stereo
if (insdat[0x1F] & 4)
{
// 16-bit
if (insoff + inslen * 2 > p->mseg_len)
if ((insoff + (inslen * 2)) > p->mseg_len)
inslen = (p->mseg_len - insoff) / 2;
for (j = 0; j < inslen; ++j)
set_le16(&p->mseg[insoff + (j << 1)], get_le16(&p->mseg[insoff + (j << 1)]) - 0x8000);
set_le16(&p->mseg[insoff + (j * 2)], get_le16(&p->mseg[insoff + (j * 2)]) - 0x8000);
}
else
{
// 8-bit
if (insoff + inslen > p->mseg_len)
if ((insoff + inslen) > p->mseg_len)
inslen = p->mseg_len - insoff;
for (j = 0; j < inslen; ++j)
p->mseg[insoff + j] = p->mseg[insoff + j] - 0x80;
}
@ -1655,6 +1658,8 @@ static void loadheaderparms(PLAYER *p)
void st3play_PlaySong(void *_p, int16_t startOrder)
{
uint8_t i;
int8_t offset;
int8_t bit;
uint8_t dat;
int16_t pan;
PLAYER *p;
@ -1674,28 +1679,29 @@ void st3play_PlaySong(void *_p, int16_t startOrder)
for (i = 0; i < 32; ++i)
{
pan = (p->mseg[0x33] & 0x80) ? ((p->mseg[0x40 + i] & 0x08) ? 192 : 64) : 128;
if (p->mseg[0x35] == 0xFC) // non-default pannings follow
{
dat = p->mseg[(p->patternadd + (get_le16(&p->mseg[0x24]) << 1)) + i];
dat = p->mseg[(p->patternadd + (get_le16(&p->mseg[0x24]) * 2)) + i];
if (dat & 0x20)
pan = (dat & 0x0F) << 4;
pan = (dat & 0x0F) * 16;
}
if (p->stereomode)
p->chn[i].apanpos = pan;
else
p->chn[i].apanpos = 128;
p->chn[i].apanpos = p->stereomode ? pan : 128;
voiceSetPanning(p, i, pan);
}
p->Playing = 1;
setSamplesPerFrame(p, ((p->outputFreq * 5UL) / 2 / p->tempo));
setSamplesPerFrame(p, ((p->outputFreq * 5) / 2 / p->tempo));
p->isMixing = 1;
p->loopCount = 0;
memset(p->playedOrder, 0, sizeof (p->playedOrder));
p->playedOrder[startOrder / 8] = 1 << (startOrder % 8);
offset = startOrder / 8;
bit = 1 << (startOrder % 8);
p->playedOrder[offset] = bit;
}
int8_t st3play_LoadModule(void *_p, const uint8_t *module, size_t size)
@ -1738,7 +1744,7 @@ int8_t st3play_LoadModule(void *_p, const uint8_t *module, size_t size)
p->mseg_len = (uint32_t)(size);
p->instrumentadd = 0x60 + p->mseg[0x20];
p->patternadd = p->instrumentadd + (p->mseg[0x22] << 1);
p->patternadd = p->instrumentadd + (p->mseg[0x22] * 2);
p->tickdelay = 0;
p->musiccount = 0;
p->patterndelay = 0;
@ -1832,7 +1838,8 @@ static void s_tickdelay(PLAYER *p, chn_t *ch) // NON-ST3
if ( (p->tracker == OPENMPT)
|| (p->tracker == BEROTRACKER)
|| (p->tracker == IMPULSE_TRACKER)
|| (p->tracker == SCHISM_TRACKER))
|| (p->tracker == SCHISM_TRACKER)
)
{
p->tickdelay += (ch->info & 0x0F);
}
@ -1843,7 +1850,7 @@ static void s_setpanpos(PLAYER *p, chn_t *ch)
ch->surround = 0;
voiceSetSurround(p, ch->channelnum, 0);
ch->apanpos = (ch->info & 0x0F) << 4;
ch->apanpos = (ch->info & 0x0F) * 16;
setpan(p, ch->channelnum);
}
@ -1879,6 +1886,9 @@ static void s_sndcntrl(PLAYER *p, chn_t *ch) // NON-ST3
static void s_patloop(PLAYER *p, chn_t *ch)
{
int8_t offset;
int8_t bit;
if (!(ch->info & 0x0F))
{
p->patloopstart = p->np_row;
@ -1900,7 +1910,12 @@ static void s_patloop(PLAYER *p, chn_t *ch)
p->np_patoff = -1; // force reseek
if (p->patloopstart == 0)
p->playedOrder[(p->np_ord - 1) / 8] &= ~(1 << ((p->np_ord - 1) % 8));
{
offset = (p->np_ord - 1) / 8;
bit = 1 << ((p->np_ord - 1) % 8);
p->playedOrder[offset] &= ~bit;
}
}
else
{
@ -1963,7 +1978,7 @@ static void s_setspeed(PLAYER *p, chn_t *ch)
static void s_jmpto(PLAYER *p, chn_t *ch)
{
if (ch->info != 255)
if (ch->info != 0xFF)
{
p->breakpat = 1;
p->np_ord = ch->info;
@ -1987,7 +2002,7 @@ static void s_volslide(PLAYER *p, chn_t *ch)
getlastnfo(p, ch);
infohi = ch->info >> 0x04;
infohi = ch->info >> 4;
infolo = ch->info & 0x0F;
if (infolo == 0x0F)
@ -2017,14 +2032,12 @@ static void s_volslide(PLAYER *p, chn_t *ch)
}
if (ch->avol < 0) ch->avol = 0;
if (ch->avol > 63) ch->avol = 63;
else if (ch->avol > 63) ch->avol = 63;
setvol(p, ch->channelnum, 0, 0);
if (p->volslidetype == 1)
s_vibrato(p, ch);
else if (p->volslidetype == 2)
s_toneslide(p, ch);
if (p->volslidetype == 1) s_vibrato(p, ch);
else if (p->volslidetype == 2) s_toneslide(p, ch);
}
static void s_slidedown(PLAYER *p, chn_t *ch)
@ -2037,7 +2050,7 @@ static void s_slidedown(PLAYER *p, chn_t *ch)
{
if (ch->info >= 0xE0) return; // no fine slides here
ch->aspd += ((uint16_t)(ch->info) << 2);
ch->aspd += (ch->info * 4);
if (ch->aspd > 32767) ch->aspd = 32767;
}
else
@ -2051,7 +2064,7 @@ static void s_slidedown(PLAYER *p, chn_t *ch)
}
else
{
ch->aspd += ((ch->info & 0x0F) << 2);
ch->aspd += ((ch->info & 0x0F) * 4);
if (ch->aspd > 32767) ch->aspd = 32767;
}
}
@ -2071,7 +2084,7 @@ static void s_slideup(PLAYER *p, chn_t *ch)
{
if (ch->info >= 0xE0) return; // no fine slides here
ch->aspd -= ((uint16_t)(ch->info) << 2);
ch->aspd -= (ch->info * 4);
if (ch->aspd < 0) ch->aspd = 0;
}
else
@ -2085,7 +2098,7 @@ static void s_slideup(PLAYER *p, chn_t *ch)
}
else
{
ch->aspd -= ((ch->info & 0x0F) << 2);
ch->aspd -= ((ch->info & 0x0F) * 4);
if (ch->aspd < 0) ch->aspd = 0;
}
}
@ -2122,13 +2135,13 @@ static void s_toneslide(PLAYER *p, chn_t *ch)
{
if (ch->aorgspd < ch->asldspd)
{
ch->aorgspd += ((uint16_t)(ch->info) << 2);
ch->aorgspd += (ch->info * 4);
if (ch->aorgspd > ch->asldspd)
ch->aorgspd = ch->asldspd;
}
else
{
ch->aorgspd -= ((uint16_t)(ch->info) << 2);
ch->aorgspd -= (ch->info * 4);
if (ch->aorgspd < ch->asldspd)
ch->aorgspd = ch->asldspd;
}
@ -2180,7 +2193,7 @@ static void s_vibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
}
// ramp
@ -2195,7 +2208,7 @@ static void s_vibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibramp[cnt >> 1];
dat = vibramp[cnt / 2];
}
// square
@ -2210,7 +2223,7 @@ static void s_vibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsqu[cnt >> 1];
dat = vibsqu[cnt / 2];
}
// random
@ -2225,7 +2238,7 @@ static void s_vibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = (int32_t)(vibsin[cnt >> 1]);
dat = vibsin[cnt / 2];
cnt += (p->patmusicrand & 0x1E);
}
@ -2243,7 +2256,7 @@ static void s_vibrato(PLAYER *p, chn_t *ch)
ch->aspd = dat;
setspd(p, ch->channelnum);
ch->avibcnt = (cnt + ((ch->info >> 4) << 1)) & 0x7E;
ch->avibcnt = (cnt + ((ch->info >> 4) * 2)) & 0x7E;
}
}
@ -2287,16 +2300,15 @@ static void s_arp(PLAYER *p, chn_t *ch)
getlastnfo(p, ch);
tick = p->musiccount % 3;
noteadd = 0;
if (tick == 1)
noteadd = ch->info >> 4;
else if (tick == 2)
noteadd = ch->info & 0x0F;
if (tick == 1) noteadd = ch->info >> 4;
else if (tick == 2) noteadd = ch->info & 0x0F;
else noteadd = 0;
// check for octave overflow
octa = ch->lastnote & 0xF0;
note = (ch->lastnote & 0x0F) + noteadd;
while (note >= 12)
{
note -= 12;
@ -2330,7 +2342,7 @@ static void s_chanvolslide(PLAYER *p, chn_t *ch) // NON-ST3
else
ch->info = ch->nxymem;
infohi = ch->nxymem >> 0x04;
infohi = ch->nxymem >> 4;
infolo = ch->nxymem & 0x0F;
if (infolo == 0x0F)
@ -2360,7 +2372,7 @@ static void s_chanvolslide(PLAYER *p, chn_t *ch) // NON-ST3
}
if (ch->chanvol < 0) ch->chanvol = 0;
if (ch->chanvol > 64) ch->chanvol = 64;
else if (ch->chanvol > 64) ch->chanvol = 64;
setvol(p, ch->channelnum, 0, 0);
}
@ -2390,29 +2402,29 @@ static void s_panslide(PLAYER *p, chn_t *ch) // NON-ST3
else
ch->info = ch->pxymem;
infohi = ch->pxymem >> 0x04;
infohi = ch->pxymem >> 4;
infolo = ch->pxymem & 0x0F;
if (infolo == 0x0F)
{
if (!infohi)
ch->apanpos += (infolo << 2);
ch->apanpos += (infolo * 4);
else if (!p->musiccount)
ch->apanpos -= (infohi << 2);
ch->apanpos -= (infohi * 4);
}
else if (infohi == 0x0F)
{
if (!infolo)
ch->apanpos -= (infohi << 2);
ch->apanpos -= (infohi * 4);
else if (!p->musiccount)
ch->apanpos += (infolo << 2);
ch->apanpos += (infolo * 4);
}
else if (p->musiccount) // don't rely on fastvolslide flag here
{
if (!infolo)
ch->apanpos -= (infohi << 2);
ch->apanpos -= (infohi * 4);
else
ch->apanpos += (infolo << 2);
ch->apanpos += (infolo * 4);
}
else
{
@ -2420,7 +2432,7 @@ static void s_panslide(PLAYER *p, chn_t *ch) // NON-ST3
}
if (ch->apanpos < 0) ch->apanpos = 0;
if (ch->apanpos > 256) ch->apanpos = 256;
else if (ch->apanpos > 256) ch->apanpos = 256;
setpan(p, ch->channelnum);
}
@ -2431,7 +2443,7 @@ static void s_retrig(PLAYER *p, chn_t *ch)
uint8_t infohi;
getlastnfo(p, ch);
infohi = ch->info >> 0x04;
infohi = ch->info >> 4;
if (!(ch->info & 0x0F) || (ch->atrigcnt < (ch->info & 0x0F)))
{
@ -2447,10 +2459,10 @@ static void s_retrig(PLAYER *p, chn_t *ch)
if (!retrigvoladd[16 + infohi])
ch->avol += retrigvoladd[infohi];
else
ch->avol = (int8_t)(((int16_t)(ch->avol) * retrigvoladd[16 + infohi]) >> 4);
ch->avol = (int8_t)((ch->avol * retrigvoladd[16 + infohi]) / 16);
if (ch->avol > 63) ch->avol = 63;
if (ch->avol < 0) ch->avol = 0;
else if (ch->avol < 0) ch->avol = 0;
setvol(p, ch->channelnum, 0, 0);
@ -2487,7 +2499,7 @@ static void s_tremolo(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
}
// ramp
@ -2502,7 +2514,7 @@ static void s_tremolo(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibramp[cnt >> 1];
dat = vibramp[cnt / 2];
}
// square
@ -2517,7 +2529,7 @@ static void s_tremolo(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsqu[cnt >> 1];
dat = vibsqu[cnt / 2];
}
// random
@ -2532,7 +2544,7 @@ static void s_tremolo(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
cnt += (p->patmusicrand & 0x1E);
}
@ -2543,13 +2555,14 @@ static void s_tremolo(PLAYER *p, chn_t *ch)
}
dat = ((dat * (ch->info & 0x0F)) >> 7) + ch->aorgvol;
if (dat > 63) dat = 63;
if (dat < 0) dat = 0;
else if (dat < 0) dat = 0;
ch->avol = (int8_t)(dat);
setvol(p, ch->channelnum, 0, 0);
ch->avibcnt = (cnt + ((ch->info >> 4) << 1)) & 0x7E;
ch->avibcnt = (cnt + ((ch->info >> 4) * 2)) & 0x7E;
}
}
@ -2632,7 +2645,7 @@ static void s_finevibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
}
// ramp
@ -2647,7 +2660,7 @@ static void s_finevibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibramp[cnt >> 1];
dat = vibramp[cnt / 2];
}
// square
@ -2662,7 +2675,7 @@ static void s_finevibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsqu[cnt >> 1];
dat = vibsqu[cnt / 2];
}
// random
@ -2677,7 +2690,7 @@ static void s_finevibrato(PLAYER *p, chn_t *ch)
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
cnt += (p->patmusicrand & 0x1E);
}
@ -2695,7 +2708,7 @@ static void s_finevibrato(PLAYER *p, chn_t *ch)
ch->aspd = dat;
setspd(p, ch->channelnum);
ch->avibcnt = (cnt + ((ch->info >> 4) << 1)) & 0x7E;
ch->avibcnt = (cnt + ((ch->info >> 4) * 2)) & 0x7E;
}
}
@ -2718,7 +2731,7 @@ static void s_globvolslide(PLAYER *p, chn_t *ch) // NON-ST3
else
ch->info = ch->wxymem;
infohi = ch->wxymem >> 0x04;
infohi = ch->wxymem >> 4;
infolo = ch->wxymem & 0x0F;
if (infolo == 0x0F)
@ -2748,10 +2761,11 @@ static void s_globvolslide(PLAYER *p, chn_t *ch) // NON-ST3
}
if (p->globalvol < 0) p->globalvol = 0;
if (p->globalvol > 64) p->globalvol = 64;
else if (p->globalvol > 64) p->globalvol = 64;
// update all channels now
for (i = 0; i < (p->lastachannelused + 1); ++i) setvol(p, i, 0, 0);
for (i = 0; i < (p->lastachannelused + 1); ++i)
setvol(p, i, 0, 0);
}
}
@ -2765,7 +2779,7 @@ static void s_setpan(PLAYER *p, chn_t *ch) // NON-ST3
ch->surround = 0;
voiceSetSurround(p, ch->channelnum, 0);
ch->apanpos = (int16_t)(ch->info) << 1;
ch->apanpos = ch->info * 2;
setpan(p, ch->channelnum);
}
else if (ch->info == 0xA4) // surround
@ -2815,7 +2829,7 @@ static void s_panbrello(PLAYER *p, chn_t *ch) // NON-ST3
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
}
// ramp
@ -2830,7 +2844,7 @@ static void s_panbrello(PLAYER *p, chn_t *ch) // NON-ST3
if (cnt & 0x80) cnt = 0;
}
dat = vibramp[cnt >> 1];
dat = vibramp[cnt / 2];
}
// square
@ -2845,7 +2859,7 @@ static void s_panbrello(PLAYER *p, chn_t *ch) // NON-ST3
if (cnt & 0x80) cnt = 0;
}
dat = vibsqu[cnt >> 1];
dat = vibsqu[cnt / 2];
}
// random
@ -2860,7 +2874,7 @@ static void s_panbrello(PLAYER *p, chn_t *ch) // NON-ST3
if (cnt & 0x80) cnt = 0;
}
dat = vibsin[cnt >> 1];
dat = vibsin[cnt / 2];
cnt += (p->patmusicrand & 0x1E);
}
@ -2873,11 +2887,11 @@ static void s_panbrello(PLAYER *p, chn_t *ch) // NON-ST3
dat = ((dat * (ch->info & 0x0F)) >> 4) + ch->apanpos;
if (dat < 0) dat = 0;
if (dat > 256) dat = 256;
else if (dat > 256) dat = 256;
voiceSetPanning(p, ch->channelnum, dat);
ch->apancnt = (cnt + ((ch->info >> 6) << 1)) & 0x7E;
ch->apancnt = (cnt + ((ch->info >> 6) * 2)) & 0x7E;
}
}
@ -3106,7 +3120,7 @@ static inline void mix8b(PLAYER *p, uint8_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler) || !resampler_get_sample_count(resampler)))
while (interpolating > 0 && (resampler_get_free_count(resampler) || !resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, sampleData[samplePosition], 8);
@ -3124,7 +3138,7 @@ static inline void mix8b(PLAYER *p, uint8_t ch, uint32_t samples)
if (samplePosition < 0)
{
samplePosition = 0;
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
@ -3140,11 +3154,17 @@ static inline void mix8b(PLAYER *p, uint8_t ch, uint32_t samples)
else
{
if (samplePosition >= sampleLength)
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler) || !resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, 0, 8);
++interpolating;
}
v->samplePosition = samplePosition;
v->interpolating = (int8_t)(interpolating);
@ -3269,7 +3289,7 @@ static inline void mix8bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler[0]) ||
while (interpolating > 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
@ -3290,7 +3310,7 @@ static inline void mix8bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
if (samplePosition < 0)
{
samplePosition = 0;
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
@ -3306,11 +3326,20 @@ static inline void mix8bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
else
{
if (samplePosition >= sampleLength)
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
resampler_write_sample_fixed(resampler[0], 0, 8);
resampler_write_sample_fixed(resampler[1], 0, 8);
++interpolating;
}
v->samplePosition = samplePosition;
v->interpolating = (int8_t)(interpolating);
@ -3432,7 +3461,7 @@ static inline void mix16b(PLAYER *p, uint8_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler) || !resampler_get_sample_count(resampler)))
while (interpolating > 0 && (resampler_get_free_count(resampler) || !resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, (int16_t)(get_le16(&sampleData[samplePosition])), 16);
@ -3450,7 +3479,7 @@ static inline void mix16b(PLAYER *p, uint8_t ch, uint32_t samples)
if (samplePosition < 0)
{
samplePosition = 0;
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
@ -3466,11 +3495,17 @@ static inline void mix16b(PLAYER *p, uint8_t ch, uint32_t samples)
else
{
if (samplePosition >= sampleLength)
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler) || !resampler_get_sample_count(resampler)))
{
resampler_write_sample_fixed(resampler, 0, 16);
++interpolating;
}
v->samplePosition = samplePosition;
v->interpolating = (int8_t)(interpolating);
@ -3595,7 +3630,7 @@ static inline void mix16bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
{
samplePosition = v->samplePosition;
while (interpolating && (resampler_get_free_count(resampler[0]) ||
while (interpolating > 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
@ -3616,7 +3651,7 @@ static inline void mix16bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
if (samplePosition < 0)
{
samplePosition = 0;
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
@ -3632,11 +3667,20 @@ static inline void mix16bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
else
{
if (samplePosition >= sampleLength)
interpolating = 0;
interpolating = -resampler_get_padding_size();
}
}
}
while (interpolating < 0 && (resampler_get_free_count(resampler[0]) ||
(!resampler_get_sample_count(resampler[0]) &&
!resampler_get_sample_count(resampler[1]))))
{
resampler_write_sample_fixed(resampler[0], 0, 16);
resampler_write_sample_fixed(resampler[1], 0, 16);
++interpolating;
}
v->samplePosition = samplePosition;
v->interpolating = (int8_t)(interpolating);
@ -3840,7 +3884,7 @@ void st3play_RenderFloat(void *_p, float *buffer, int32_t count)
if (p->fmChip && p->fmResampler)
st3play_AdlibMix(p, outputStream, samplesTodo);
outputStream += (samplesTodo << 1);
outputStream += (samplesTodo * 2);
}
p->samplesLeft -= samplesTodo;
@ -3923,16 +3967,18 @@ void st3play_RenderFixed16(void *_p, int16_t *buffer, int32_t count, int8_t dept
void FreeSong(PLAYER *p)
{
uint16_t i;
p->Playing = 0;
if (p->adpcmSamples)
if (p->adpcmSamples != NULL)
{
int i, j;
for (i = 0, j = get_le16(&p->mseg[0x22]); i < j; i++)
for (i = 0; i < get_le16(&p->mseg[0x22]); ++i)
{
if (p->adpcmSamples[i])
if (p->adpcmSamples[i] != NULL)
free(p->adpcmSamples[i]);
}
free(p->adpcmSamples);
p->adpcmSamples = NULL;
}
@ -3951,7 +3997,8 @@ void FreeSong(PLAYER *p)
void st3play_Mute(void *_p, int8_t channel, int8_t mute)
{
PLAYER *p;
int8_t mask;
int8_t offset;
int8_t bit;
uint8_t adlibChannel;
if (channel > 31)
@ -3960,12 +4007,14 @@ void st3play_Mute(void *_p, int8_t channel, int8_t mute)
p = (PLAYER *)(_p);
adlibChannel = (p->mseg[0x40 + channel] & 0x7F) - 16;
mask = 1 << (channel % 8);
offset = channel / 8;
bit = 1 << (channel % 8);
if (mute)
p->muted[channel / 8] |= mask;
p->muted[offset] |= bit;
else
p->muted[channel / 8] &= ~mask;
p->muted[offset] &= ~bit;
if (adlibChannel < 9)
Chip_Mute(p->fmChip, adlibChannel, mute);