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Updated modplay/ft2play, fixing a crash bug

CQTexperiment
Chris Moeller 2014-03-20 20:07:59 -07:00
parent de8df552c2
commit bdd43b2a4b
1 changed files with 76 additions and 11 deletions
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81
Frameworks/modplay/modplay/ft2play.c
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@ -309,6 +309,7 @@ typedef struct
float targetVolR; float targetVolR;
float volDeltaL; float volDeltaL;
float volDeltaR; float volDeltaR;
int8_t rampTerminates;
#endif #endif
} VOICE; } VOICE;
@ -338,16 +339,22 @@ typedef struct
StmTyp Stm[127]; StmTyp Stm[127];
SongTyp Song; SongTyp Song;
InstrTyp *Instr[255 + 1]; InstrTyp *Instr[255 + 1];
#ifdef USE_VOL_RAMP
VOICE voice[127*2]; VOICE voice[127*2];
void *resampler[127*2*2]; void *resampler[127*2*2];
#else
VOICE voice[127];
void *resampler[127*2];
#endif
float *PanningTab; float *PanningTab;
float f_outputFreq; float f_outputFreq;
#ifdef USE_VOL_RAMP #ifdef USE_VOL_RAMP
float f_samplesPerFrame; float f_samplesPerFrame;
float f_samplesPerFrameInit; float f_samplesPerFrameSharp;
#endif #endif
// pre-initialized variables // pre-initialized variables
@ -386,7 +393,7 @@ static void voiceSetSource(PLAYER *, uint8_t i, const int8_t *sampleData,
int32_t sampleLoopEnd, int8_t loopEnabled, int32_t sampleLoopEnd, int8_t loopEnabled,
int8_t sixteenbit, int8_t stereo); int8_t sixteenbit, int8_t stereo);
static void voiceSetSamplePosition(PLAYER *, uint8_t i, uint16_t value); static void voiceSetSamplePosition(PLAYER *, uint8_t i, uint16_t value);
static void voiceSetVolume(PLAYER *, uint8_t i, float vol, uint8_t pan, uint8_t init); static void voiceSetVolume(PLAYER *, uint8_t i, float vol, uint8_t pan, uint8_t sharp);
static void voiceSetSamplingFrequency(PLAYER *, uint8_t i, float samplingFrequency); static void voiceSetSamplingFrequency(PLAYER *, uint8_t i, float samplingFrequency);
@ -989,8 +996,6 @@ CheckEffects:
pl->Song.Speed = pl->Song.InitSpeed; pl->Song.Speed = pl->Song.InitSpeed;
pl->Song.Tempo = pl->Song.InitTempo; pl->Song.Tempo = pl->Song.InitTempo;
pl->Song.GlobVol = 64; pl->Song.GlobVol = 64;
pl->loopCount++;
} }
else else
{ {
@ -1977,7 +1982,11 @@ static void MainPlayer(PLAYER *p) // periodically called from mixer
{ {
ch = &p->Stm[i]; ch = &p->Stm[i];
#ifdef USE_VOL_RAMP
if ((ch->Status & (IS_Vol | IS_NyTon)) == IS_Vol) if ((ch->Status & (IS_Vol | IS_NyTon)) == IS_Vol)
#else
if (ch->Status & IS_Vol)
#endif
voiceSetVolume(p, ch->Nr, ch->FinalVol, ch->FinalPan, 0); voiceSetVolume(p, ch->Nr, ch->FinalVol, ch->FinalPan, 0);
if (ch->Status & IS_Period) if (ch->Status & IS_Period)
@ -1985,11 +1994,13 @@ static void MainPlayer(PLAYER *p) // periodically called from mixer
if (ch->Status & IS_NyTon) if (ch->Status & IS_NyTon)
{ {
#ifdef USE_VOL_RAMP
p->voice[ch->Nr + 127] = p->voice[ch->Nr]; p->voice[ch->Nr + 127] = p->voice[ch->Nr];
voiceSetVolume(p, ch->Nr, ch->FinalVol, ch->FinalPan, 1); voiceSetVolume(p, ch->Nr, ch->FinalVol, ch->FinalPan, 1);
voiceSetVolume(p, ch->Nr + 127, 0, ch->FinalPan, 1); voiceSetVolume(p, ch->Nr + 127, 0, ch->FinalPan, 1);
lanczos_resampler_dup_inplace(p->resampler[ch->Nr + 127], p->resampler[ch->Nr]); lanczos_resampler_dup_inplace(p->resampler[ch->Nr + 127], p->resampler[ch->Nr]);
lanczos_resampler_dup_inplace(p->resampler[ch->Nr + 127 + 254], p->resampler[ch->Nr + 254]); lanczos_resampler_dup_inplace(p->resampler[ch->Nr + 127 + 254], p->resampler[ch->Nr + 254]);
#endif
s = ch->InstrOfs; s = ch->InstrOfs;
@ -2666,6 +2677,9 @@ void voiceSetSource(PLAYER *p, uint8_t i, const int8_t *sampleData,
p->voice[i].stereo = stereo; p->voice[i].stereo = stereo;
p->voice[i].interpolating = 1; p->voice[i].interpolating = 1;
p->voice[i].oversampleCount = 0; p->voice[i].oversampleCount = 0;
#ifdef USE_VOL_RAMP
p->voice[i].rampTerminates = 0;
#endif
// for 9xx set offset // for 9xx set offset
if (p->voice[i].samplePosition >= p->voice[i].sampleLength) if (p->voice[i].samplePosition >= p->voice[i].sampleLength)
@ -2675,7 +2689,11 @@ void voiceSetSource(PLAYER *p, uint8_t i, const int8_t *sampleData,
} }
lanczos_resampler_clear(p->resampler[i]); lanczos_resampler_clear(p->resampler[i]);
#ifdef USE_VOL_RAMP
lanczos_resampler_clear(p->resampler[i+254]); lanczos_resampler_clear(p->resampler[i+254]);
#else
lanczos_resampler_clear(p->resampler[i+127]);
#endif
} }
void voiceSetSamplePosition(PLAYER *p, uint8_t i, uint16_t value) void voiceSetSamplePosition(PLAYER *p, uint8_t i, uint16_t value)
@ -2685,18 +2703,27 @@ void voiceSetSamplePosition(PLAYER *p, uint8_t i, uint16_t value)
p->voice[i].oversampleCount = 0; p->voice[i].oversampleCount = 0;
lanczos_resampler_clear(p->resampler[i]); lanczos_resampler_clear(p->resampler[i]);
#ifdef USE_VOL_RAMP
lanczos_resampler_clear(p->resampler[i+254]); lanczos_resampler_clear(p->resampler[i+254]);
#else
lanczos_resampler_clear(p->resampler[i+127]);
#endif
} }
void voiceSetVolume(PLAYER *p, uint8_t i, float vol, uint8_t pan, uint8_t init) void voiceSetVolume(PLAYER *p, uint8_t i, float vol, uint8_t pan, uint8_t sharp)
{ {
#ifdef USE_VOL_RAMP #ifdef USE_VOL_RAMP
const float rampRate = init ? p->f_samplesPerFrameInit : p->f_samplesPerFrame; const float rampRate = sharp ? p->f_samplesPerFrameSharp : p->f_samplesPerFrame;
if (init && vol) if (sharp)
{
if (vol)
{ {
p->voice[i].volumeL = 0.0f; p->voice[i].volumeL = 0.0f;
p->voice[i].volumeR = 0.0f; p->voice[i].volumeR = 0.0f;
} }
else
p->voice[i].rampTerminates = 1;
}
p->voice[i].targetVolL = vol * p->PanningTab[256 - pan]; p->voice[i].targetVolL = vol * p->PanningTab[256 - pan];
p->voice[i].targetVolR = vol * p->PanningTab[ pan]; p->voice[i].targetVolR = vol * p->PanningTab[ pan];
p->voice[i].volDeltaL = (p->voice[i].targetVolL - p->voice[i].volumeL) * rampRate; p->voice[i].volDeltaL = (p->voice[i].targetVolL - p->voice[i].volumeL) * rampRate;
@ -2860,6 +2887,13 @@ static inline void mix8b(PLAYER *p, uint32_t ch, uint32_t samples)
if (p->voice[ch].volumeR < p->voice[ch].targetVolR) if (p->voice[ch].volumeR < p->voice[ch].targetVolR)
p->voice[ch].volumeR = p->voice[ch].targetVolR; p->voice[ch].volumeR = p->voice[ch].targetVolR;
} }
if (p->voice[ch].rampTerminates && !p->voice[ch].volumeL && !p->voice[ch].volumeR)
{
p->voice[ch].sampleData = NULL;
p->voice[ch].samplePosition = 0;
p->voice[ch].busy = 0;
}
#endif #endif
p->masterBufferL[j] += sampleL; p->masterBufferL[j] += sampleL;
@ -2907,7 +2941,11 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
sampleData = p->voice[ch].sampleData; sampleData = p->voice[ch].sampleData;
resampler[0] = p->resampler[ch]; resampler[0] = p->resampler[ch];
#ifdef USE_VOL_RAMP
resampler[1] = p->resampler[ch+254]; resampler[1] = p->resampler[ch+254];
#else
resampler[1] = p->resampler[ch+127];
#endif
lanczos_resampler_set_rate(resampler[0], p->voice[ch].incRate * (float)samplingInterpolation); lanczos_resampler_set_rate(resampler[0], p->voice[ch].incRate * (float)samplingInterpolation);
lanczos_resampler_set_rate(resampler[1], p->voice[ch].incRate * (float)samplingInterpolation); lanczos_resampler_set_rate(resampler[1], p->voice[ch].incRate * (float)samplingInterpolation);
@ -3021,6 +3059,13 @@ static inline void mix8bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
if (p->voice[ch].volumeR < p->voice[ch].targetVolR) if (p->voice[ch].volumeR < p->voice[ch].targetVolR)
p->voice[ch].volumeR = p->voice[ch].targetVolR; p->voice[ch].volumeR = p->voice[ch].targetVolR;
} }
if (p->voice[ch].rampTerminates && !p->voice[ch].volumeL && !p->voice[ch].volumeR)
{
p->voice[ch].sampleData = NULL;
p->voice[ch].samplePosition = 0;
p->voice[ch].busy = 0;
}
#endif #endif
p->masterBufferL[j] += sampleL; p->masterBufferL[j] += sampleL;
@ -3176,6 +3221,13 @@ static inline void mix16b(PLAYER *p, uint32_t ch, uint32_t samples)
if (p->voice[ch].volumeR < p->voice[ch].targetVolR) if (p->voice[ch].volumeR < p->voice[ch].targetVolR)
p->voice[ch].volumeR = p->voice[ch].targetVolR; p->voice[ch].volumeR = p->voice[ch].targetVolR;
} }
if (p->voice[ch].rampTerminates && !p->voice[ch].volumeL && !p->voice[ch].volumeR)
{
p->voice[ch].sampleData = NULL;
p->voice[ch].samplePosition = 0;
p->voice[ch].busy = 0;
}
#endif #endif
p->masterBufferL[j] += sampleL; p->masterBufferL[j] += sampleL;
@ -3223,7 +3275,11 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
sampleData = (const int16_t *)(p->voice[ch].sampleData); sampleData = (const int16_t *)(p->voice[ch].sampleData);
resampler[0] = p->resampler[ch]; resampler[0] = p->resampler[ch];
#ifdef USE_VOL_RAMP
resampler[1] = p->resampler[ch+254]; resampler[1] = p->resampler[ch+254];
#else
resampler[1] = p->resampler[ch+127];
#endif
lanczos_resampler_set_rate(resampler[0], p->voice[ch].incRate * (float)samplingInterpolation); lanczos_resampler_set_rate(resampler[0], p->voice[ch].incRate * (float)samplingInterpolation);
lanczos_resampler_set_rate(resampler[1], p->voice[ch].incRate * (float)samplingInterpolation); lanczos_resampler_set_rate(resampler[1], p->voice[ch].incRate * (float)samplingInterpolation);
@ -3337,6 +3393,13 @@ static inline void mix16bstereo(PLAYER *p, uint32_t ch, uint32_t samples)
if (p->voice[ch].volumeR < p->voice[ch].targetVolR) if (p->voice[ch].volumeR < p->voice[ch].targetVolR)
p->voice[ch].volumeR = p->voice[ch].targetVolR; p->voice[ch].volumeR = p->voice[ch].targetVolR;
} }
if (p->voice[ch].rampTerminates && !p->voice[ch].volumeL && !p->voice[ch].volumeR)
{
p->voice[ch].sampleData = NULL;
p->voice[ch].samplePosition = 0;
p->voice[ch].busy = 0;
}
#endif #endif
p->masterBufferL[j] += sampleL; p->masterBufferL[j] += sampleL;
@ -3380,7 +3443,9 @@ static void mixSampleBlock(PLAYER *p, float *outputStream, uint32_t sampleBlockL
if (p->muted[i / 8] & (1 << (i % 8))) if (p->muted[i / 8] & (1 << (i % 8)))
continue; continue;
mixChannel(p, i, sampleBlockLength); mixChannel(p, i, sampleBlockLength);
#ifdef USE_VOL_RAMP
mixChannel(p, i + 127, sampleBlockLength); mixChannel(p, i + 127, sampleBlockLength);
#endif
} }
for (i = 0; i < sampleBlockLength; ++i) for (i = 0; i < sampleBlockLength; ++i)
@ -3756,7 +3821,7 @@ static void setSamplesPerFrame(PLAYER *p, uint32_t val)
#ifdef USE_VOL_RAMP #ifdef USE_VOL_RAMP
p->f_samplesPerFrame = 1.0f / ((float)(val) / 4.0f); p->f_samplesPerFrame = 1.0f / ((float)(val) / 4.0f);
p->f_samplesPerFrameInit = 1.0f / (p->f_outputFreq / 1000.0f); // 1ms p->f_samplesPerFrameSharp = 1.0f / (p->f_outputFreq / 1000.0f); // 1ms
#endif #endif
} }