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Updated modplay, bringing volume ramping to st3play

CQTexperiment
Chris Moeller 2014-03-20 22:02:55 -07:00
parent bdd43b2a4b
commit a567183510
2 changed files with 435 additions and 46 deletions
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8
Frameworks/modplay/modplay/ft2play.c
View File

@ -3563,7 +3563,11 @@ void * ft2play_Alloc(uint32_t _samplingFrequency, int8_t interpolation)
lanczos_init();
#ifdef USE_VOL_RAMP
for ( i = 0; i < 127 * 2 * 2; ++i )
#else
for ( i = 0; i < 127 * 2; ++i )
#endif
{
p->resampler[i] = lanczos_resampler_create();
if ( !p->resampler[i] )
@ -3659,7 +3663,11 @@ void ft2play_Free(void *_p)
if (p->linearPeriods) free(p->linearPeriods); p->linearPeriods = NULL;
if (p->NilPatternLine) free(p->NilPatternLine); p->NilPatternLine = NULL;
#ifdef USE_VOL_RAMP
for ( i = 0; i < 127 * 2 * 2; ++i )
#else
for ( i = 0; i < 127 * 2; ++i )
#endif
{
if ( p->resampler[i] )
lanczos_resampler_delete( p->resampler[i] );

473
Frameworks/modplay/modplay/st3play.c
View File

@ -17,6 +17,7 @@
** * Middle-C speeds beyond 65535
** * Process the last 16 channels as PCM
** * Process 8 octaves instead of 7
** * Compile-time optional volume ramping
**
** - Effects:
** * Command S9x (sound control - only S91/S90 so far)
@ -66,6 +67,8 @@
#include "st3play.h"
#define USE_VOL_RAMP
// TRACKER ID
enum
{
@ -148,6 +151,16 @@ typedef struct
float panningL;
float panningR;
float orgPanR;
#ifdef USE_VOL_RAMP
float targetVol;
float targetPanL;
float targetPanR;
float volDelta;
float panDeltaL;
float panDeltaR;
int8_t rampTerminates;
#endif
} VOICE;
// VARIABLES / STATE
@ -177,13 +190,24 @@ typedef struct
int32_t soundBufferSize;
uint32_t outputFreq;
#ifdef USE_VOL_RAMP
VOICE voice[32*2];
void *resampler[64*2];
#else
VOICE voice[32];
void *resampler[64];
#endif
float f_outputFreq;
float f_masterVolume;
#ifdef USE_VOL_RAMP
float f_samplesPerFrame;
float f_samplesPerFrameSharp;
#endif
// pre-initialized variables
int8_t samplingInterpolation;// = 1;
float *masterBufferL;// = NULL;
@ -301,7 +325,7 @@ static void voiceSetSource(PLAYER *, uint8_t voiceNumber, const int8_t *sampleDa
int32_t sampleLength, int32_t sampleLoopLength, int32_t sampleLoopEnd,
int8_t loopEnabled, int8_t sixteenbit, int8_t stereo, int8_t adpcm);
static void voiceSetSamplePosition(PLAYER *, uint8_t voiceNumber, uint16_t value);
static void voiceSetVolume(PLAYER *, uint8_t voiceNumber, float volume);
static void voiceSetVolume(PLAYER *, uint8_t voiceNumber, float volume, uint8_t sharp);
static void voiceSetSurround(PLAYER *, uint8_t voiceNumber, int8_t surround);
static void voiceSetPanning(PLAYER *, uint8_t voiceNumber, uint16_t pan);
static void voiceSetSamplingFrequency(PLAYER *, uint8_t voiceNumber, float samplingFrequency);
@ -463,7 +487,11 @@ void * st3play_Alloc(uint32_t outputFreq, int8_t interpolation)
lanczos_init();
#ifdef USE_VOL_RAMP
for (i = 0; i < 64 * 2; ++i)
#else
for (i = 0; i < 64; ++i)
#endif
{
p->resampler[i] = lanczos_resampler_create();
if ( !p->resampler[i] )
@ -505,7 +533,11 @@ void st3play_Free(void *_p)
FreeSong(p);
for (i = 0; i < 64; ++i)
#ifdef USE_VOL_RAMP
for (i = 0; i < 64 * 2; ++i)
#else
for (i = 0; i < 64; ++i)
#endif
{
if ( p->resampler[i] )
lanczos_resampler_delete( p->resampler[i] );
@ -597,10 +629,14 @@ static inline void setspd(PLAYER *p, uint8_t ch)
voiceSetSamplingFrequency(p, ch, 14317056.0f / (float)tmpspd);
}
static inline void setvol(PLAYER *p, uint8_t ch)
static inline void setvol(PLAYER *p, uint8_t ch, uint8_t sharp)
{
p->chn[ch].achannelused |= 0x80;
voiceSetVolume(p, ch, ((float)(p->chn[ch].avol) / 63.0f) * ((float)(p->chn[ch].chanvol) / 64.0f) * ((float)(p->globalvol) / 64.0f));
#ifdef USE_VOL_RAMP
voiceSetVolume(p, ch + ((sharp == 2) ? 32 : 0), (sharp == 2) ? 0.0f : ((float)(p->chn[ch].avol) / 63.0f) * ((float)(p->chn[ch].chanvol) / 64.0f) * ((float)(p->globalvol) / 64.0f), sharp);
#else
voiceSetVolume(p, ch, ((float)(p->chn[ch].avol) / 63.0f) * ((float)(p->chn[ch].chanvol) / 64.0f) * ((float)(p->globalvol) / 64.0f), sharp);
#endif
}
static inline void setpan(PLAYER *p, uint8_t ch)
@ -841,6 +877,9 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
uint32_t insrepbeg;
uint32_t insrepend;
int8_t shift;
#ifdef USE_VOL_RAMP
uint8_t volassigned = 0;
#endif
if (p->chn[ch].ins)
{
@ -861,7 +900,6 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
if (p->chn[ch].avol > 63) p->chn[ch].avol = 63;
p->chn[ch].aorgvol = p->chn[ch].avol;
setvol(p, ch);
insoffs = (uint32_t)(((uint32_t)(insdat[0x0D])<<16)|((uint16_t)(insdat[0x0F])<<8)|insdat[0x0E])<<4;
@ -887,6 +925,30 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
if ((insdat[0x1F] & 1) && inslen && (insrepend > insrepbeg))
loop = 1;
#ifdef USE_VOL_RAMP
p->voice[ch + 32] = p->voice[ch];
setvol(p, ch, 2);
lanczos_resampler_dup_inplace(p->resampler[ch + 32], p->resampler[ch]);
lanczos_resampler_dup_inplace(p->resampler[ch + 32 + 64], p->resampler[ch + 64]);
if (p->chn[ch].vol != 255)
{
if (p->chn[ch].vol <= 64)
{
p->chn[ch].avol = p->chn[ch].vol;
p->chn[ch].aorgvol = p->chn[ch].vol;
}
else
// NON-ST3
if ((p->chn[ch].vol >= 128) && (p->chn[ch].vol <= 192))
{
p->chn[ch].apanpos = (p->chn[ch].vol - 128) << 2;
setpan(p, ch);
}
}
volassigned = 1;
#endif
setvol(p, ch, 1);
voiceSetSource(p, ch, (const int8_t *)(&p->mseg[insoffs]), inslen,
insrepend - insrepbeg, insrepend, loop,
insdat[0x1F] & 4, insdat[0x1F] & 2, insdat[0x1E] == 4);
@ -924,7 +986,7 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
p->chn[ch].asldspd = 65535;
setspd(p, ch);
setvol(p, ch);
setvol(p, ch, 0);
// shutdown channel
voiceSetSource(p, ch, NULL, 0, 0, 0, 0, 0, 0, 0);
@ -951,14 +1013,18 @@ static inline void doamiga(PLAYER *p, uint8_t ch)
}
}
#ifdef USE_VOL_RAMP
if (p->chn[ch].vol != 255 && !volassigned)
#else
if (p->chn[ch].vol != 255)
#endif
{
if (p->chn[ch].vol <= 64)
{
p->chn[ch].avol = p->chn[ch].vol;
p->chn[ch].aorgvol = p->chn[ch].vol;
setvol(p, ch);
setvol(p, ch, 0);
return;
}
@ -1647,7 +1713,7 @@ static void s_volslide(PLAYER *p, chn_t *ch)
if (ch->avol < 0) ch->avol = 0;
if (ch->avol > 63) ch->avol = 63;
setvol(p, ch->channelnum);
setvol(p, ch->channelnum, 0);
if (p->volslidetype == 1)
s_vibrato(p, ch);
@ -1890,7 +1956,7 @@ static void s_tremor(PLAYER *p, chn_t *ch)
ch->atreon = 0;
ch->avol = 0;
setvol(p, ch->channelnum);
setvol(p, ch->channelnum, 0);
ch->atremor = ch->info & 0x0F;
}
@ -1899,7 +1965,7 @@ static void s_tremor(PLAYER *p, chn_t *ch)
ch->atreon = 1;
ch->avol = ch->aorgvol;
setvol(p, ch->channelnum);
setvol(p, ch->channelnum, 0);
ch->atremor = ch->info >> 4;
}
@ -1988,7 +2054,7 @@ 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;
setvol(p, ch->channelnum);
setvol(p, ch->channelnum, 0);
}
}
@ -2077,7 +2143,7 @@ static void s_retrig(PLAYER *p, chn_t *ch)
if (ch->avol > 63) ch->avol = 63;
if (ch->avol < 0) ch->avol = 0;
setvol(p, ch->channelnum);
setvol(p, ch->channelnum, 0);
ch->atrigcnt++; // probably a mistake? Keep it anyways.
}
@ -2172,7 +2238,7 @@ static void s_tremolo(PLAYER *p, chn_t *ch)
if (dat < 0) dat = 0;
ch->avol = (int8_t)(dat);
setvol(p, ch->channelnum);
setvol(p, ch->channelnum, 0);
ch->avibcnt = (cnt + ((ch->info >> 4) << 1)) & 0x7E;
}
@ -2376,7 +2442,7 @@ static void s_globvolslide(PLAYER *p, chn_t *ch) // NON-ST3
if (p->globalvol > 64) p->globalvol = 64;
// update all channels now
for (i = 0; i < (p->lastachannelused + 1); ++i) setvol(p, i);
for (i = 0; i < (p->lastachannelused + 1); ++i) setvol(p, i, 0);
}
}
@ -2507,6 +2573,11 @@ static void s_panbrello(PLAYER *p, chn_t *ch) // NON-ST3
void setSamplesPerFrame(PLAYER *p, uint32_t val)
{
p->samplesPerFrame = val;
#ifdef USE_VOL_RAMP
p->f_samplesPerFrame = 1.0f / ((float)(val) / 4.0f);
p->f_samplesPerFrameSharp = 1.0f / (p->f_outputFreq / 1000.0f); // 1ms
#endif
}
void setSamplingInterpolation(PLAYER *p, int8_t value)
@ -2550,12 +2621,19 @@ void voiceSetSource(PLAYER *p, uint8_t voiceNumber, const int8_t *sampleData,
p->voice[voiceNumber].mixing = 1;
p->voice[voiceNumber].interpolating = 1;
p->voice[voiceNumber].oversampleCount = 0;
#ifdef USE_VOL_RAMP
p->voice[voiceNumber].rampTerminates = 0;
#endif
if (p->voice[voiceNumber].samplePosition >= p->voice[voiceNumber].sampleLength)
p->voice[voiceNumber].samplePosition = 0;
lanczos_resampler_clear( p->resampler[voiceNumber] );
#ifdef USE_VOL_RAMP
lanczos_resampler_clear( p->resampler[voiceNumber+64] );
#else
lanczos_resampler_clear( p->resampler[voiceNumber+32] );
#endif
}
void voiceSetSamplePosition(PLAYER *p, uint8_t voiceNumber, uint16_t value)
@ -2578,30 +2656,67 @@ void voiceSetSamplePosition(PLAYER *p, uint8_t voiceNumber, uint16_t value)
}
lanczos_resampler_clear( p->resampler[voiceNumber] );
#ifdef USE_VOL_RAMP
lanczos_resampler_clear( p->resampler[voiceNumber+64] );
#else
lanczos_resampler_clear( p->resampler[voiceNumber+32] );
#endif
}
void voiceSetVolume(PLAYER *p, uint8_t voiceNumber, float volume)
void voiceSetVolume(PLAYER *p, uint8_t voiceNumber, float volume, uint8_t sharp)
{
#ifdef USE_VOL_RAMP
const float rampRate = sharp ? p->f_samplesPerFrameSharp : p->f_samplesPerFrame;
if (sharp)
{
if (volume)
p->voice[voiceNumber].volume = 0.0f;
else
p->voice[voiceNumber].rampTerminates = 1;
}
p->voice[voiceNumber].targetVol = volume;
p->voice[voiceNumber].volDelta = (p->voice[voiceNumber].targetVol - p->voice[voiceNumber].volume) * rampRate;
#else
p->voice[voiceNumber].volume = volume;
#endif
}
void voiceSetSurround(PLAYER *p, uint8_t voiceNumber, int8_t surround)
{
#ifdef USE_VOL_RAMP
const float rampRate = p->f_samplesPerFrameSharp;
if (surround)
p->voice[voiceNumber].targetPanR = -p->voice[voiceNumber].orgPanR;
else
p->voice[voiceNumber].targetPanR = p->voice[voiceNumber].orgPanR;
p->voice[voiceNumber].panDeltaR = (p->voice[voiceNumber].targetPanR - p->voice[voiceNumber].panningR) * rampRate;
#else
if (surround)
p->voice[voiceNumber].panningR = -p->voice[voiceNumber].orgPanR;
else
p->voice[voiceNumber].panningR = p->voice[voiceNumber].orgPanR;
#endif
}
void voiceSetPanning(PLAYER *p, uint8_t voiceNumber, uint16_t pan)
{
#ifdef USE_VOL_RAMP
const float rampRate = p->f_samplesPerFrameSharp;
#endif
float pf;
pf = (float)(pan) / 256.0f;
#ifdef USE_VOL_RAMP
p->voice[voiceNumber].targetPanL = 1.0f - pf;
p->voice[voiceNumber].targetPanR = pf;
p->voice[voiceNumber].panDeltaL = (p->voice[voiceNumber].targetPanL - p->voice[voiceNumber].panningL) * rampRate;
p->voice[voiceNumber].panDeltaR = (p->voice[voiceNumber].targetPanR - p->voice[voiceNumber].panningR) * rampRate;
#else
p->voice[voiceNumber].panningL = 1.0f - pf;
p->voice[voiceNumber].panningR = pf;
#endif
p->voice[voiceNumber].orgPanR = pf;
}
@ -2692,7 +2807,57 @@ static inline void mix8b(PLAYER *p, uint8_t ch, uint32_t samples)
p->masterBufferL[j] += (sample * panningL);
p->masterBufferR[j] += (sample * panningR);
#ifdef USE_VOL_RAMP
volume += p->voice[ch].volDelta;
panningL += p->voice[ch].panDeltaL;
panningR += p->voice[ch].panDeltaR;
if (p->voice[ch].volDelta >= 0.0f)
{
if (volume > p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
else
{
if (volume < p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
if (p->voice[ch].panDeltaL >= 0.0f)
{
if (panningL > p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
else
{
if (panningL < p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
if (p->voice[ch].panDeltaR >= 0.0f)
{
if (panningR > p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
else
{
if (panningR < p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
if (p->voice[ch].rampTerminates && !volume)
{
p->voice[ch].mixing = 0;
break;
}
#endif
}
#ifdef USE_VOL_RAMP
p->voice[ch].volume = volume;
p->voice[ch].panningL = panningL;
p->voice[ch].panningR = panningR;
#endif
}
static inline void mix8bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
@ -2728,7 +2893,11 @@ static inline void mix8bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
samplingInterpolation = p->samplingInterpolation ? 1 : 32;
resampler[0] = p->resampler[ch];
#ifdef USE_VOL_RAMP
resampler[1] = p->resampler[ch+64];
#else
resampler[1] = p->resampler[ch+32];
#endif
lanczos_resampler_set_rate(resampler[0], p->voice[ch].incRate * (float)samplingInterpolation);
lanczos_resampler_set_rate(resampler[1], p->voice[ch].incRate * (float)samplingInterpolation);
@ -2786,7 +2955,57 @@ static inline void mix8bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
p->masterBufferL[j] += (sampleL * panningL);
p->masterBufferR[j] += (sampleR * panningR);
#ifdef USE_VOL_RAMP
volume += p->voice[ch].volDelta;
panningL += p->voice[ch].panDeltaL;
panningR += p->voice[ch].panDeltaR;
if (p->voice[ch].volDelta >= 0.0f)
{
if (volume > p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
else
{
if (volume < p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
if (p->voice[ch].panDeltaL >= 0.0f)
{
if (panningL > p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
else
{
if (panningL < p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
if (p->voice[ch].panDeltaR >= 0.0f)
{
if (panningR > p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
else
{
if (panningR < p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
if (p->voice[ch].rampTerminates && !volume)
{
p->voice[ch].mixing = 0;
break;
}
#endif
}
#ifdef USE_VOL_RAMP
p->voice[ch].volume = volume;
p->voice[ch].panningL = panningL;
p->voice[ch].panningR = panningR;
#endif
}
static inline void mix16b(PLAYER *p, uint8_t ch, uint32_t samples)
@ -2871,7 +3090,57 @@ static inline void mix16b(PLAYER *p, uint8_t ch, uint32_t samples)
p->masterBufferL[j] += (sample * panningL);
p->masterBufferR[j] += (sample * panningR);
#ifdef USE_VOL_RAMP
volume += p->voice[ch].volDelta;
panningL += p->voice[ch].panDeltaL;
panningR += p->voice[ch].panDeltaR;
if (p->voice[ch].volDelta >= 0.0f)
{
if (volume > p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
else
{
if (volume < p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
if (p->voice[ch].panDeltaL >= 0.0f)
{
if (panningL > p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
else
{
if (panningL < p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
if (p->voice[ch].panDeltaR >= 0.0f)
{
if (panningR > p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
else
{
if (panningR < p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
if (p->voice[ch].rampTerminates && !volume)
{
p->voice[ch].mixing = 0;
break;
}
#endif
}
#ifdef USE_VOL_RAMP
p->voice[ch].volume = volume;
p->voice[ch].panningL = panningL;
p->voice[ch].panningR = panningR;
#endif
}
static inline void mix16bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
@ -2907,7 +3176,11 @@ static inline void mix16bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
samplingInterpolation = p->samplingInterpolation ? 1 : 32;
resampler[0] = p->resampler[ch];
#ifdef USE_VOL_RAMP
resampler[1] = p->resampler[ch+64];
#else
resampler[1] = p->resampler[ch+32];
#endif
lanczos_resampler_set_rate(resampler[0], p->voice[ch].incRate * (float)samplingInterpolation);
lanczos_resampler_set_rate(resampler[1], p->voice[ch].incRate * (float)samplingInterpolation);
@ -2965,7 +3238,57 @@ static inline void mix16bstereo(PLAYER *p, uint8_t ch, uint32_t samples)
p->masterBufferL[j] += (sampleL * panningL);
p->masterBufferR[j] += (sampleR * panningR);
#ifdef USE_VOL_RAMP
volume += p->voice[ch].volDelta;
panningL += p->voice[ch].panDeltaL;
panningR += p->voice[ch].panDeltaR;
if (p->voice[ch].volDelta >= 0.0f)
{
if (volume > p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
else
{
if (volume < p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
if (p->voice[ch].panDeltaL >= 0.0f)
{
if (panningL > p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
else
{
if (panningL < p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
if (p->voice[ch].panDeltaR >= 0.0f)
{
if (panningR > p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
else
{
if (panningR < p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
if (p->voice[ch].rampTerminates && !volume)
{
p->voice[ch].mixing = 0;
break;
}
#endif
}
#ifdef USE_VOL_RAMP
p->voice[ch].volume = volume;
p->voice[ch].panningL = panningL;
p->voice[ch].panningR = panningR;
#endif
}
static inline int8_t get_adpcm_sample(const int8_t *sampleDictionary, const uint8_t *sampleData, int32_t samplePosition, int8_t *lastDelta)
@ -3082,6 +3405,79 @@ static inline void mixadpcm(PLAYER *p, uint8_t ch, uint32_t samples)
p->masterBufferL[j] += (sample * panningL);
p->masterBufferR[j] += (sample * panningR);
#ifdef USE_VOL_RAMP
volume += p->voice[ch].volDelta;
panningL += p->voice[ch].panDeltaL;
panningR += p->voice[ch].panDeltaR;
if (p->voice[ch].volDelta >= 0.0f)
{
if (volume > p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
else
{
if (volume < p->voice[ch].targetVol)
volume = p->voice[ch].targetVol;
}
if (p->voice[ch].panDeltaL >= 0.0f)
{
if (panningL > p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
else
{
if (panningL < p->voice[ch].targetPanL)
panningL = p->voice[ch].targetPanL;
}
if (p->voice[ch].panDeltaR >= 0.0f)
{
if (panningR > p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
else
{
if (panningR < p->voice[ch].targetPanR)
panningR = p->voice[ch].targetPanR;
}
if (p->voice[ch].rampTerminates && !volume)
{
p->voice[ch].mixing = 0;
break;
}
#endif
}
#ifdef USE_VOL_RAMP
p->voice[ch].volume = volume;
p->voice[ch].panningL = panningL;
p->voice[ch].panningR = panningR;
#endif
}
void mixChannel(PLAYER *p, uint8_t i, uint32_t sampleBlockLength)
{
if (p->voice[i].incRate && p->voice[i].mixing)
{
if (p->voice[i].stereo)
{
if (p->voice[i].sixteenBit)
mix16bstereo(p, i, sampleBlockLength);
else
mix8bstereo(p, i, sampleBlockLength);
}
else
{
if (p->voice[i].sixteenBit)
mix16b(p, i, sampleBlockLength);
else if (p->voice[i].adpcm)
mixadpcm(p, i, sampleBlockLength);
else
mix8b(p, i, sampleBlockLength);
}
}
}
@ -3103,25 +3499,10 @@ void mixSampleBlock(PLAYER *p, float *outputStream, uint32_t sampleBlockLength)
{
if (p->muted[i / 8] & (1 << (i % 8)))
continue;
if (p->voice[i].incRate && p->voice[i].mixing)
{
if (p->voice[i].stereo)
{
if (p->voice[i].sixteenBit)
mix16bstereo(p, i, sampleBlockLength);
else
mix8bstereo(p, i, sampleBlockLength);
}
else
{
if (p->voice[i].sixteenBit)
mix16b(p, i, sampleBlockLength);
else if (p->voice[i].adpcm)
mixadpcm(p, i, sampleBlockLength);
else
mix8b(p, i, sampleBlockLength);
}
}
mixChannel(p, i, sampleBlockLength);
#ifdef USE_VOL_RAMP
mixChannel(p, i + 32, sampleBlockLength);
#endif
}
for (j = 0; j < sampleBlockLength; ++j)
@ -3226,18 +3607,18 @@ void FreeSong(PLAYER *p)
p->ModuleLoaded = 0;
}
void st3play_Mute(void *_p, int8_t channel, int8_t mute)
{
PLAYER * p = (PLAYER *)_p;
int8_t mask = 1 << (channel % 8);
if (channel > 31)
return;
if (mute)
p->muted[channel / 8] |= mask;
else
p->muted[channel / 8] &= ~mask;
}
void st3play_Mute(void *_p, int8_t channel, int8_t mute)
{
PLAYER * p = (PLAYER *)_p;
int8_t mask = 1 << (channel % 8);
if (channel > 31)
return;
if (mute)
p->muted[channel / 8] |= mask;
else
p->muted[channel / 8] &= ~mask;
}
int32_t st3play_GetLoopCount(void *_p)
{
PLAYER * p = (PLAYER *)_p;