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Overhauled volume ramping in modplay, and outright fixed it in ft2play

CQTexperiment
Chris Moeller 2014-09-05 23:05:20 -07:00
parent efcbb5be50
commit 014e03bda5
2 changed files with 491 additions and 440 deletions
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314
Frameworks/modplay/modplay/ft2play.c
View File

@ -394,25 +394,19 @@ typedef struct
TonTyp *NilPatternLine;
TonTyp *Patt[256];
StmTyp Stm[127];
StmTyp Stm[MAX_VOICES];
SongTyp Song;
InstrTyp *Instr[255 + 1];
#ifdef USE_VOL_RAMP
VOICE voice[127*2];
VOICE voice[TOTAL_VOICES];
void *resampler[127*2*2];
#else
VOICE voice[127];
void *resampler[127*2];
#endif
void *resampler[TOTAL_VOICES*2];
float *PanningTab;
float f_outputFreq;
#ifdef USE_VOL_RAMP
float f_samplesPerFrame;
float f_samplesPerFrameSharp;
float f_samplesPerFrame005;
float f_samplesPerFrame010;
#endif
/* pre-initialized variables */
@ -449,7 +443,7 @@ static void voiceSetSource(PLAYER *, uint8_t i, const int8_t *sampleData,
int32_t sampleLoopEnd, int8_t loopEnabled,
int8_t sixteenbit, int8_t stereo);
static void voiceSetSamplePosition(PLAYER *, uint8_t i, uint16_t value);
static void voiceSetVolume(PLAYER *, uint8_t i, float vol, uint8_t pan);
static void voiceSetVolume(PLAYER *, uint8_t i, float vol, uint8_t pan, uint8_t note_on);
static void voiceSetSamplingFrequency(PLAYER *, uint8_t i, uint32_t samplingFrequency);
static void ft2play_FreeSong(PLAYER *);
@ -2064,6 +2058,10 @@ static void MainPlayer(PLAYER *p) /* periodically called from mixer */
uint8_t i;
int8_t tickzero;
#ifdef USE_VOL_RAMP
int32_t rampStyle = p->rampStyle;
#endif
if (p->Playing)
{
@ -2082,9 +2080,9 @@ static void MainPlayer(PLAYER *p) /* periodically called from mixer */
for (i = 0; i < p->Song.AntChn; ++i)
{
if (p->Patt[p->Song.PattNr] != NULL)
GetNewNote(p, &p->Stm[i], &p->Patt[p->Song.PattNr][(p->Song.PattPos * 127) + i]);
GetNewNote(p, &p->Stm[i], &p->Patt[p->Song.PattNr][(p->Song.PattPos * MAX_VOICES) + i]);
else
GetNewNote(p, &p->Stm[i], &p->NilPatternLine[(p->Song.PattPos * 127) + i]);
GetNewNote(p, &p->Stm[i], &p->NilPatternLine[(p->Song.PattPos * MAX_VOICES) + i]);
FixaEnvelopeVibrato(p, &p->Stm[i]);
}
@ -2166,24 +2164,22 @@ static void MainPlayer(PLAYER *p) /* periodically called from mixer */
{
ch = &p->Stm[i];
if (ch->Status & IS_Vol)
voiceSetVolume(p, ch->Nr, ch->FinalVol, ch->FinalPan);
if (ch->Status & IS_Period)
voiceSetSamplingFrequency(p, ch->Nr, GetFrequenceValue(p, ch->FinalPeriod));
if (ch->Status & IS_NyTon)
{
s = ch->InstrOfs;
#ifdef USE_VOL_RAMP
if (voiceIsActive(p, ch->Nr))
if (rampStyle > 0 && voiceIsActive(p, ch->Nr))
{
memcpy(p->voice + SPARE_OFFSET + ch->Nr, p->voice + ch->Nr, sizeof (VOICE));
int32_t ChNr = ch->Nr;
memcpy(p->voice + SPARE_OFFSET + ChNr, p->voice + ChNr, sizeof (VOICE));
p->voice[SPARE_OFFSET + ch->Nr].faderDest = 0.0f;
p->voice[SPARE_OFFSET + ch->Nr].faderDelta =
(p->voice[SPARE_OFFSET + ch->Nr].faderDest - p->voice[SPARE_OFFSET + ch->Nr].fader) / (p->f_outputFreq * 0.010f);
p->voice[SPARE_OFFSET + ChNr].faderDest = 0.0f;
p->voice[SPARE_OFFSET + ChNr].faderDelta =
(p->voice[SPARE_OFFSET + ChNr].faderDest - p->voice[SPARE_OFFSET + ChNr].fader) * p->f_samplesPerFrame010;
resampler_dup_inplace(p->resampler[SPARE_OFFSET + ChNr], p->resampler[ChNr]);
resampler_dup_inplace(p->resampler[TOTAL_VOICES + SPARE_OFFSET + ChNr], p->resampler[TOTAL_VOICES + ChNr]);
}
#endif
@ -2191,12 +2187,21 @@ static void MainPlayer(PLAYER *p) /* periodically called from mixer */
voiceSetSamplePosition(p, ch->Nr, ch->SmpStartPos);
#ifdef USE_VOL_RAMP
p->voice[ch->Nr].fader = 0.0f;
p->voice[ch->Nr].faderDest = 1.0f;
p->voice[ch->Nr].faderDelta = (p->voice[ch->Nr].faderDest - p->voice[ch->Nr].fader) / (p->f_outputFreq * 0.005f);
if (rampStyle > 0)
{
p->voice[ch->Nr].fader = 0.0f;
p->voice[ch->Nr].faderDest = 1.0f;
p->voice[ch->Nr].faderDelta = (p->voice[ch->Nr].faderDest - p->voice[ch->Nr].fader) * p->f_samplesPerFrame005;
}
#endif
}
if (ch->Status & IS_Vol)
voiceSetVolume(p, ch->Nr, ch->FinalVol, ch->FinalPan, ch->Status & IS_NyTon);
if (ch->Status & IS_Period)
voiceSetSamplingFrequency(p, ch->Nr, GetFrequenceValue(p, ch->FinalPeriod));
ch->Status = 0;
}
}
@ -2226,7 +2231,7 @@ static void StopVoices(PLAYER *p)
ch->FinalPan = 128;
ch->VibDepth = 0;
voiceSetVolume(p, a, ch->FinalVol, ch->FinalPan);
voiceSetVolume(p, a, ch->FinalVol, ch->FinalPan, 1);
}
}
@ -2856,11 +2861,6 @@ int8_t ft2play_LoadModule(void *_p, const uint8_t *buffer, size_t size)
static 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
}
static void setSamplingInterpolation(PLAYER *p, int8_t value)
@ -2901,9 +2901,6 @@ static inline void voiceSetSource(PLAYER *p, uint8_t i, const int8_t *sampleData
v->loopingForward = 1;
v->stereo = stereo;
v->interpolating = 1;
resampler_clear(p->resampler[i]);
resampler_clear(p->resampler[i+TOTAL_VOICES]);
}
static inline void voiceSetSamplePosition(PLAYER *p, uint8_t i, uint16_t value)
@ -2919,21 +2916,28 @@ static inline void voiceSetSamplePosition(PLAYER *p, uint8_t i, uint16_t value)
}
v->interpolating = 1;
resampler_clear(p->resampler[i]);
resampler_clear(p->resampler[i+TOTAL_VOICES]);
}
static inline void voiceSetVolume(PLAYER *p, uint8_t i, float vol, uint8_t pan)
static inline void voiceSetVolume(PLAYER *p, uint8_t i, float vol, uint8_t pan, uint8_t note_on)
{
VOICE *v;
v = &p->voice[i];
#ifdef USE_VOL_RAMP
v->targetVolL = vol * p->PanningTab[256 - pan];
v->targetVolR = vol * p->PanningTab[pan];
v->volDeltaL = (v->targetVolL - v->volumeL) / (p->f_outputFreq * 0.005f);
v->volDeltaR = (v->targetVolR - v->volumeR) / (p->f_outputFreq * 0.005f);
if (!note_on && p->rampStyle > 1)
{
v->targetVolL = vol * p->PanningTab[256 - pan];
v->targetVolR = vol * p->PanningTab[pan];
v->volDeltaL = (v->targetVolL - v->volumeL) * p->f_samplesPerFrame005;
v->volDeltaR = (v->targetVolR - v->volumeR) * p->f_samplesPerFrame005;
}
else
{
v->targetVolL = v->volumeL = vol * p->PanningTab[256 - pan];
v->targetVolR = v->volumeR = vol * p->PanningTab[pan];
v->volDeltaL = 0.0f;
v->volDeltaR = 0.0f;
}
#else
v->volumeL = vol * p->PanningTab[256 - pan];
v->volumeR = vol * p->PanningTab[pan];
@ -3043,8 +3047,9 @@ static inline void mix8b(PLAYER *p, uint32_t ch, uint32_t samples)
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
if ( !resampler_ready(resampler) )
if ( !resampler_get_sample_count(resampler) )
{
resampler_clear(resampler);
v->sampleData = NULL;
break;
}
@ -3053,50 +3058,50 @@ static inline void mix8b(PLAYER *p, uint32_t ch, uint32_t samples)
resampler_remove_sample(resampler);
#ifdef USE_VOL_RAMP
v->fader += v->faderDelta;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
if (rampStyle > 0)
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
v->sampleData = NULL;
}
v->fader += v->faderDelta;
sample *= v->fader;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
resampler_clear(resampler);
v->sampleData = NULL;
}
sample *= v->fader;
}
#endif
sampleL = sample * v->volumeL;
sampleR = sample * v->volumeR;
#ifdef USE_VOL_RAMP
if (rampStyle > 0)
if (rampStyle > 1)
{
v->volumeL += v->volDeltaL;
v->volumeR += v->volDeltaR;
if (v->volDeltaL >= 0.0f)
if ((v->volDeltaL > 0.0f) && (v->volumeL > v->targetVolL))
{
if (v->volumeL > v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
else
else if ((v->volDeltaL < 0.0f) && (v->volumeL < v->targetVolL))
{
if (v->volumeL < v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
if (v->volDeltaR >= 0.0f)
if ((v->volDeltaR > 0.0f) && (v->volumeR > v->targetVolR))
{
if (v->volumeR > v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
else
else if ((v->volDeltaR < 0.0f) && (v->volumeR < v->targetVolR))
{
if (v->volumeR < v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
}
#endif
@ -3206,8 +3211,9 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
if ( !resampler_ready(resampler[0]) )
if ( !resampler_get_sample_count(resampler[0]) )
{
resampler_clear(resampler);
v->sampleData = NULL;
break;
}
@ -3218,51 +3224,51 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
resampler_remove_sample(resampler[1]);
#ifdef USE_VOL_RAMP
v->fader += v->faderDelta;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
if (rampStyle > 0)
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
v->sampleData = NULL;
}
v->fader += v->faderDelta;
sampleL *= v->fader;
sampleR *= v->fader;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
resampler_clear(resampler);
v->sampleData = NULL;
}
sampleL *= v->fader;
sampleR *= v->fader;
}
#endif
sampleL *= v->volumeL;
sampleR *= v->volumeR;
#ifdef USE_VOL_RAMP
if (rampStyle > 0)
if (rampStyle > 1)
{
v->volumeL += v->volDeltaL;
v->volumeR += v->volDeltaR;
if (v->volDeltaL >= 0.0f)
if ((v->volDeltaL > 0.0f) && (v->volumeL > v->targetVolL))
{
if (v->volumeL > v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
else
else if ((v->volDeltaL < 0.0f) && (v->volumeL < v->targetVolL))
{
if (v->volumeL < v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
if (v->volDeltaR >= 0.0f)
if ((v->volDeltaR > 0.0f) && (v->volumeR > v->targetVolR))
{
if (v->volumeR > v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
else
else if ((v->volDeltaR < 0.0f) && (v->volumeR < v->targetVolR))
{
if (v->volumeR < v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
}
#endif
@ -3370,8 +3376,9 @@ static inline void mix16b(PLAYER *p, uint32_t ch, uint32_t samples)
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
if ( !resampler_ready(resampler) )
if ( !resampler_get_sample_count(resampler) )
{
resampler_clear(resampler);
v->sampleData = NULL;
break;
}
@ -3380,50 +3387,50 @@ static inline void mix16b(PLAYER *p, uint32_t ch, uint32_t samples)
resampler_remove_sample(resampler);
#ifdef USE_VOL_RAMP
v->fader += v->faderDelta;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
if (rampStyle > 0)
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
v->sampleData = NULL;
}
v->fader += v->faderDelta;
sample *= v->fader;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
resampler_clear(resampler);
v->sampleData = NULL;
}
sample *= v->fader;
}
#endif
sampleL = sample * v->volumeL;
sampleR = sample * v->volumeR;
#ifdef USE_VOL_RAMP
if (rampStyle > 0)
if (rampStyle > 1)
{
v->volumeL += v->volDeltaL;
v->volumeR += v->volDeltaR;
if (v->volDeltaL >= 0.0f)
if ((v->volDeltaL > 0.0f) && (v->volumeL > v->targetVolL))
{
if (v->volumeL > v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
else
else if ((v->volDeltaL < 0.0f) && (v->volumeL < v->targetVolL))
{
if (v->volumeL < v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
if (v->volDeltaR >= 0.0f)
if ((v->volDeltaR > 0.0f) && (v->volumeR > v->targetVolR))
{
if (v->volumeR > v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
else
else if ((v->volDeltaR < 0.0f) && (v->volumeR < v->targetVolR))
{
if (v->volumeR < v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
}
#endif
@ -3533,8 +3540,9 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
v->loopingForward = loopingForward;
v->interpolating = (int8_t)interpolating;
if ( !resampler_ready(resampler[0]) )
if ( !resampler_get_sample_count(resampler[0]) )
{
resampler_clear(resampler);
v->sampleData = NULL;
break;
}
@ -3545,51 +3553,51 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
resampler_remove_sample(resampler[1]);
#ifdef USE_VOL_RAMP
v->fader += v->faderDelta;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
if (rampStyle > 0)
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
v->sampleData = NULL;
}
v->fader += v->faderDelta;
sampleL *= v->fader;
sampleR *= v->fader;
if ((v->faderDelta > 0.0f) && (v->fader > v->faderDest))
{
v->fader = v->faderDest;
}
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
resampler_clear(resampler);
v->sampleData = NULL;
}
sampleL *= v->fader;
sampleR *= v->fader;
}
#endif
sampleL *= v->volumeL;
sampleR *= v->volumeR;
#ifdef USE_VOL_RAMP
if (rampStyle > 0)
if (rampStyle > 1)
{
v->volumeL += v->volDeltaL;
v->volumeR += v->volDeltaR;
if (v->volDeltaL >= 0.0f)
if ((v->volDeltaL > 0.0f) && (v->volumeL > v->targetVolL))
{
if (v->volumeL > v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
else
else if ((v->volDeltaL < 0.0f) && (v->volumeL < v->targetVolL))
{
if (v->volumeL < v->targetVolL)
v->volumeL = v->targetVolL;
v->volumeL = v->targetVolL;
}
if (v->volDeltaR >= 0.0f)
if ((v->volDeltaR > 0.0f) && (v->volumeR > v->targetVolR))
{
if (v->volumeR > v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
else
else if ((v->volDeltaR < 0.0f) && (v->volumeR < v->targetVolR))
{
if (v->volumeR < v->targetVolR)
v->volumeR = v->targetVolR;
v->volumeR = v->targetVolR;
}
}
#endif
@ -3748,6 +3756,10 @@ void * ft2play_Alloc(uint32_t _samplingFrequency, int8_t interpolation, int8_t r
p->outputFreq = _samplingFrequency;
p->f_outputFreq = (float)(p->outputFreq);
#ifdef USE_VOL_RAMP
p->f_samplesPerFrame010= 1.0f / (p->f_outputFreq * 0.010f);
p->f_samplesPerFrame005= 1.0f / (p->f_outputFreq * 0.005f);
#endif
p->soundBufferSize = _soundBufferSize;
p->masterBufferL = (float *)(malloc(p->soundBufferSize * sizeof (float)));
@ -3874,11 +3886,7 @@ 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
for ( i = 0; i < TOTAL_VOICES * 2; ++i )
{
if ( p->resampler[i] )
resampler_delete( p->resampler[i] );
@ -4049,7 +4057,7 @@ void ft2play_Mute(void *_p, int8_t channel, int8_t mute)
{
PLAYER * p = (PLAYER *)_p;
int8_t mask = 1 << (channel % 8);
if (channel > 127)
if (channel > MAX_VOICES)
return;
if (mute)
p->muted[channel / 8] |= mask;

617
Frameworks/modplay/modplay/st3play.c
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