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Updated st3play and fixed two horrible bugs with ft2play

CQTexperiment
Chris Moeller 2015-02-20 00:16:52 -08:00
parent 1503ece011
commit 8808e1a55a
2 changed files with 42 additions and 23 deletions
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14
Frameworks/modplay/modplay/ft2play.c
View File

@ -3248,7 +3248,8 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
if ( !resampler_get_sample_count(resampler[0]) )
{
resampler_clear(resampler);
resampler_clear(resampler[0]);
resampler_clear(resampler[1]);
v->sampleData = NULL;
break;
}
@ -3270,7 +3271,8 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
resampler_clear(resampler);
resampler_clear(resampler[0]);
resampler_clear(resampler[1]);
v->sampleData = NULL;
}
@ -3580,7 +3582,8 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
if ( !resampler_get_sample_count(resampler[0]) )
{
resampler_clear(resampler);
resampler_clear(resampler[0]);
resampler_clear(resampler[1]);
v->sampleData = NULL;
break;
}
@ -3602,7 +3605,8 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
else if ((v->faderDelta < 0.0f) && (v->fader < v->faderDest))
{
v->fader = v->faderDest;
resampler_clear(resampler);
resampler_clear(resampler[0]);
resampler_clear(resampler[1]);
v->sampleData = NULL;
}
@ -3647,6 +3651,8 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
static inline void mixChannel(PLAYER *p, uint32_t i, uint32_t sampleBlockLength)
{
if (!p->voice[i].incRate)
return;
if (p->voice[i].stereo)
{
if (p->voice[i].sixteenBit)

51
Frameworks/modplay/modplay/st3play.c
View File

@ -653,12 +653,21 @@ static void st3play_AdlibHertzTouch(PLAYER *p, uint8_t ch, int Hertz, uint8_t ke
static inline void setspd(PLAYER *p, uint8_t ch)
{
int32_t tmpspd;
uint8_t adlibChannel = (p->mseg[0x40 + ch] & 0x7F) - 16;
p->chn[ch].achannelused |= 0x80;
tmpspd = p->chn[ch].aspd;
if (tmpspd && (tmpspd < p->aspdmin))
tmpspd = p->aspdmin;
if (p->amigalimits)
{
if (p->chn[ch].aorgspd > p->aspdmax) p->chn[ch].aorgspd = p->aspdmax;
if (p->chn[ch].aorgspd < p->aspdmin) p->chn[ch].aorgspd = p->aspdmin;
if (p->chn[ch].aspd > p->aspdmax)
p->chn[ch].aspd = p->aspdmax;
}
if (p->tracker == SCREAM_TRACKER)
{
@ -667,26 +676,30 @@ static inline void setspd(PLAYER *p, uint8_t ch)
}
else
{
// *ABSOLUTE* max!
/* *ABSOLUTE* max! */
if (tmpspd > 14317056)
tmpspd = 14317056;
}
if (p->amigalimits)
{
if (p->chn[ch].aorgspd > p->aspdmax) p->chn[ch].aorgspd = p->aspdmax;
if (p->chn[ch].aorgspd < p->aspdmin) p->chn[ch].aorgspd = p->aspdmin;
p->chn[ch].aspd = tmpspd;
}
p->chn[ch].achannelused |= 0x80;
// ST3 actually uses 14317056/per instead of 14317456 (8363*1712).
// 14317056 is used in both the ST3 replayer and the S3M format docs
if (tmpspd)
voiceSetSamplingFrequency(p, ch, 14317056.0f / (float)tmpspd);
if (tmpspd == 0)
{
// cut channel
voiceSetSamplingFrequency(p, ch, 0);
return;
}
if (tmpspd < p->aspdmin)
{
tmpspd = p->aspdmin;
if (p->amigalimits && (p->chn[ch].aspd < p->aspdmin))
p->chn[ch].aspd = p->aspdmin;
}
// ST3 actually uses 14317056 (3.579264MHz * 4) instead of 14317456 (8363*1712)
if (tmpspd > 0)
voiceSetSamplingFrequency(p, ch, 14317056 / tmpspd);
if (adlibChannel < 9)
st3play_AdlibHertzTouch(p, adlibChannel, 14317056.0f / (float)tmpspd, 0);
}