long GetSampleSize(){
long bytearraylength = 0;
// Lets calculate how big the bytearray must be...
ResetSample(false);
playing_sample=true;
while(playing_sample)
bytearraylength += CalcSampleSize(256);
return bytearraylength;
}
bool SDLPlaySound(const char *filename, bool sfxr)
{
ResetParams();
bool ok = SDLSoundInit() && LoadSound(filename, sfxr);
if (ok)
{
if (cursnd.sfxr)
{
ResetSample(false);
}
else
{
cursndpos = cursnd.buf;
}
SDL_PauseAudioDevice(audioid, 0);
playing_sample=true;
}
return ok;
}
void SynthSample(int length, float* buffer, FILE* file)
{
for(int i=0;i<length;i++)
{
if(!playing_sample)
break;
rep_time++;
if(rep_limit!=0 && rep_time>=rep_limit)
{
rep_time=0;
ResetSample(true);
}
// frequency envelopes/arpeggios
arp_time++;
if(arp_limit!=0 && arp_time>=arp_limit)
{
arp_limit=0;
fperiod*=arp_mod;
}
fslide+=fdslide;
fperiod*=fslide;
if(fperiod>fmaxperiod)
{
fperiod=fmaxperiod;
if(p_freq_limit>0.0f)
playing_sample=false;
}
float rfperiod=fperiod;
if(vib_amp>0.0f)
{
vib_phase+=vib_speed;
rfperiod=fperiod*(1.0+sin(vib_phase)*vib_amp);
}
period=(int)rfperiod;
if(period<8) period=8;
square_duty+=square_slide;
if(square_duty<0.0f) square_duty=0.0f;
if(square_duty>0.5f) square_duty=0.5f;
// volume envelope
env_time++;
if(env_time>env_length[env_stage])
{
env_time=0;
env_stage++;
if(env_stage==3)
playing_sample=false;
}
if(env_stage==0)
env_vol=(float)env_time/env_length[0];
if(env_stage==1)
env_vol=1.0f+pow(1.0f-(float)env_time/env_length[1], 1.0f)*2.0f*p_env_punch;
if(env_stage==2)
env_vol=1.0f-(float)env_time/env_length[2];
// phaser step
fphase+=fdphase;
iphase=abs((int)fphase);
if(iphase>1023) iphase=1023;
if(flthp_d!=0.0f)
{
flthp*=flthp_d;
if(flthp<0.00001f) flthp=0.00001f;
if(flthp>0.1f) flthp=0.1f;
}
float ssample=0.0f;
for(int si=0;si<8;si++) // 8x supersampling
{
float sample=0.0f;
phase++;
if(phase>=period)
{
// phase=0;
phase%=period;
if(wave_type==3)
for(int i=0;i<32;i++)
noise_buffer[i]=frnd(2.0f)-1.0f;
}
// base waveform
float fp=(float)phase/period;
switch(wave_type)
{
case 0: // square
if(fp<square_duty)
sample=0.5f;
else
sample=-0.5f;
break;
case 1: // sawtooth
sample=1.0f-fp*2;
break;
case 2: // sine
sample=(float)sin(fp*2*PI);
break;
case 3: // noise
sample=noise_buffer[phase*32/period];
break;
}
// lp filter
float pp=fltp;
fltw*=fltw_d;
if(fltw<0.0f) fltw=0.0f;
if(fltw>0.1f) fltw=0.1f;
if(p_lpf_freq!=1.0f)
{
fltdp+=(sample-fltp)*fltw;
fltdp-=fltdp*fltdmp;
}
else
{
fltp=sample;
fltdp=0.0f;
}
fltp+=fltdp;
// hp filter
fltphp+=fltp-pp;
fltphp-=fltphp*flthp;
sample=fltphp;
// phaser
phaser_buffer[ipp&1023]=sample;
sample+=phaser_buffer[(ipp-iphase+1024)&1023];
ipp=(ipp+1)&1023;
// final accumulation and envelope application
ssample+=sample*env_vol;
}
ssample=ssample/8*master_vol;
ssample*=2.0f*sound_vol;
if(buffer!=NULL)
{
if(ssample>1.0f) ssample=1.0f;
if(ssample<-1.0f) ssample=-1.0f;
*buffer++=ssample;
}
if(file!=NULL)
{
// quantize depending on format
// accumulate/count to accomodate variable sample rate?
ssample*=4.0f; // arbitrary gain to get reasonable output volume...
if(ssample>1.0f) ssample=1.0f;
if(ssample<-1.0f) ssample=-1.0f;
filesample+=ssample;
fileacc++;
if(wav_freq==44100 || fileacc==2)
{
filesample/=fileacc;
fileacc=0;
if(wav_bits==16)
{
short isample=(short)(filesample*32000);
fwrite(&isample, 1, 2, file);
}
else
{
unsigned char isample=(unsigned char)(filesample*127+128);
fwrite(&isample, 1, 1, file);
}
filesample=0.0f;
}
file_sampleswritten++;
}
}
}
void PlaySample()
{
ResetSample(false);
playing_sample=true;
}
void SynthSample(int length, float* buffer, FILE* file)
{
for(int i=0;i<length;i++)
{
if(!havePlayingSample())
break;
float ssample=0.0f;
for(int j=0;j<CHANNEL_N;j++)
{
if(!channels[j].playing_sample)
continue;
// volume envelope
channels[j].env_time++;
if(channels[j].env_time>channels[j].env_length[channels[j].env_stage])
{
channels[j].env_time=0;
channels[j].env_stage++;
if(channels[j].env_stage==4)
channels[j].playing_sample=false;
}
if(channels[j].env_stage==0) continue;
switch(channels[j].env_stage)
{
case 1: channels[j].env_vol=(float)channels[j].env_time/channels[j].env_length[1]; break;
case 2: channels[j].env_vol=1.0f+pow(1.0f-(float)channels[j].env_time/channels[j].env_length[2], 1.0f)*2.0f*channels[j].p_env_punch; break;
case 3: channels[j].env_vol=1.0f-(float)channels[j].env_time/channels[j].env_length[3]; break;
}
channels[j].rep_time++;
if(channels[j].rep_limit!=0 && channels[j].rep_time>=channels[j].rep_limit)
{
channels[j].rep_time=0;
ResetSample(true, j);
}
// frequency envelopes/arpeggios
channels[j].arp_time++;
if(channels[j].arp_limit!=0 && channels[j].arp_time>=channels[j].arp_limit)
{
channels[j].arp_limit=0;
channels[j].fperiod*=channels[j].arp_mod;
}
channels[j].fslide+=channels[j].fdslide;
channels[j].fperiod*=channels[j].fslide;
if(channels[j].fperiod>channels[j].fmaxperiod)
{
channels[j].fperiod=channels[j].fmaxperiod;
if(channels[j].p_freq_limit>0.0f)
channels[j].playing_sample=false;
}
float rfperiod=channels[j].fperiod;
if(channels[j].vib_amp>0.0f)
{
channels[j].vib_phase+=channels[j].vib_speed;
rfperiod=channels[j].fperiod*(1.0+sin(channels[j].vib_phase)*channels[j].vib_amp);
}
channels[j].period=(int)rfperiod;
if(channels[j].period<8) channels[j].period=8;
channels[j].square_duty+=channels[j].square_slide;
if(channels[j].square_duty<0.0f) channels[j].square_duty=0.0f;
if(channels[j].square_duty>0.5f) channels[j].square_duty=0.5f;
// phaser step
channels[j].fphase+=channels[j].fdphase;
channels[j].iphase=abs((int)channels[j].fphase);
if(channels[j].iphase>=PHASER_BUFFER_SIZE) channels[j].iphase=PHASER_BUFFER_SIZE;
if(channels[j].flthp_d!=0.0f)
{
channels[j].flthp*=channels[j].flthp_d;
if(channels[j].flthp<0.00001f) channels[j].flthp=0.00001f;
if(channels[j].flthp>0.1f) channels[j].flthp=0.1f;
}
float sub_ssample = 0.0f;
for(int si=0;si<8;si++) // 8x supersampling
{
float sample=0.0f;
channels[j].phase++;
if(channels[j].phase>=channels[j].period)
{
// phase=0;
channels[j].phase%=channels[j].period;
if(channels[j].wave_type==3)
for(int i=0;i<32;i++)
channels[j].noise_buffer[i]=frnd(2.0f)-1.0f;
}
// base waveform
float fp=(float)channels[j].phase/channels[j].period;
switch(channels[j].wave_type)
{
case 0: // square
if(fp<channels[j].square_duty)
sample=0.5f;
else
sample=-0.5f;
break;
case 1: // sawtooth
sample=1.0f-fp*2;
break;
case 2: // sine
sample=(float)sin(fp*2*PI);
break;
case 3: // noise
sample=channels[j].noise_buffer[channels[j].phase*32/channels[j].period];
break;
}
// lp filter
float pp=channels[j].fltp;
channels[j].fltw*=channels[j].fltw_d;
if(channels[j].fltw<0.0f) channels[j].fltw=0.0f;
if(channels[j].fltw>0.1f) channels[j].fltw=0.1f;
if(channels[j].p_lpf_freq!=1.0f)
{
channels[j].fltdp+=(sample-channels[j].fltp)*channels[j].fltw;
channels[j].fltdp-=channels[j].fltdp*channels[j].fltdmp;
}
else
{
channels[j].fltp=sample;
channels[j].fltdp=0.0f;
}
channels[j].fltp+=channels[j].fltdp;
// hp filter
channels[j].fltphp+=channels[j].fltp-pp;
channels[j].fltphp-=channels[j].fltphp*channels[j].flthp;
sample=channels[j].fltphp;
// phaser
channels[j].phaser_buffer[channels[j].ipp&PHASER_BUFFER_MASK]=sample;
sample+=channels[j].phaser_buffer[(channels[j].ipp-channels[j].iphase+PHASER_BUFFER_SIZE)&PHASER_BUFFER_MASK];
channels[j].ipp=(channels[j].ipp+1)&PHASER_BUFFER_MASK;
// final accumulation and envelope application
sub_ssample+=sample*channels[j].env_vol;
}
ssample+=(sub_ssample/8*master_vol*master_vol_multiplier)*(2.0f*channels[j].sound_vol);
}
if(buffer!=NULL)
{
if(ssample>1.0f) ssample=1.0f;
if(ssample<-1.0f) ssample=-1.0f;
*buffer++=ssample;
}
if(file!=NULL)
{
// quantize depending on format
// accumulate/count to accomodate variable sample rate?
ssample*=4.0f; // arbitrary gain to get reasonable output volume...
if(ssample>1.0f) ssample=1.0f;
if(ssample<-1.0f) ssample=-1.0f;
filesample+=ssample;
fileacc++;
if(wav_freq==44100 || fileacc==2)
{
filesample/=fileacc;
fileacc=0;
if(wav_bits==16)
{
short isample=(short)(filesample*32000);
fwrite(&isample, 1, 2, file);
}
else
{
unsigned char isample=(unsigned char)(filesample*127+128);
fwrite(&isample, 1, 1, file);
}
filesample=0.0f;
}
file_sampleswritten++;
}
}
}
void PlaySample()
{
ResetSample(false);
for(int i=0;i<CHANNEL_N;i++)
channels[i].playing_sample=channels[i].enabled;
}
void generator::PlaySample(void)
{
ResetSample(false);
playing_sample=true;
}
int SynthSample(int length, float* buffer) {
for(int i=0;i<length;i++)
{
if(!playing_sample)
return i;
rep_time++;
if(rep_limit!=0 && rep_time>=rep_limit)
{
rep_time=0;
ResetSample(true);
}
// frequency envelopes/arpeggios
arp_time++;
if(arp_limit!=0 && arp_time>=arp_limit)
{
arp_limit=0;
fperiod*=arp_mod;
}
fslide+=fdslide;
fperiod*=fslide;
if(fperiod>fmaxperiod)
{
fperiod=fmaxperiod;
if(p_freq_limit>0.0f) {
playing_sample=false;
}
}
float rfperiod=fperiod;
if(vib_amp>0.0f)
{
vib_phase+=vib_speed;
rfperiod=fperiod*(1.0+sin(vib_phase)*vib_amp);
}
period=(int)rfperiod;
if(period<8) period=8;
square_duty+=square_slide;
if(square_duty<0.0f) square_duty=0.0f;
if(square_duty>0.5f) square_duty=0.5f;
// volume envelope
env_time++;
if(env_time>env_length[env_stage])
{
env_time=0;
env_stage++;
if(env_stage==3) {
playing_sample=false;
}
}
if(env_stage==0)
env_vol=(float)env_time/env_length[0];
if(env_stage==1)
env_vol=1.0f+pow(1.0f-(float)env_time/env_length[1], 1.0f)*2.0f*p_env_punch;
if(env_stage==2)
env_vol=1.0f-(float)env_time/env_length[2];
// phaser step
fphase+=fdphase;
iphase=abs((int)fphase);
if(iphase>1023) iphase=1023;
if(flthp_d!=0.0f)
{
flthp*=flthp_d;
if(flthp<0.00001f) flthp=0.00001f;
if(flthp>0.1f) flthp=0.1f;
}
float ssample=0.0f;
for(int si=0;si<8;si++) // 8x supersampling
{
float sample=0.0f;
phase++;
if(phase>=period)
{
phase%=period;
if(wave_type==3)
for(int i=0;i<32;i++)
noise_buffer[i]=frnd(2.0f)-1.0f;
}
// base waveform
float fp=(float)phase/period;
switch(wave_type)
{
case 0: // square
if(fp<square_duty)
sample=0.5f;
else
sample=-0.5f;
break;
case 1: // sawtooth
sample=1.0f-fp*2;
break;
case 2: // sine
sample=(float)sin(fp*2*PI);
break;
case 3: // noise
sample=noise_buffer[phase*32/period];
break;
}
// lp filter
float pp=fltp;
fltw*=fltw_d;
if(fltw<0.0f) fltw=0.0f;
if(fltw>0.1f) fltw=0.1f;
if(p_lpf_freq!=1.0f)
{
fltdp+=(sample-fltp)*fltw;
fltdp-=fltdp*fltdmp;
}
else
{
fltp=sample;
fltdp=0.0f;
}
fltp+=fltdp;
// hp filter
fltphp+=fltp-pp;
fltphp-=fltphp*flthp;
sample=fltphp;
// phaser
phaser_buffer[ipp&1023]=sample;
sample+=phaser_buffer[(ipp-iphase+1024)&1023];
ipp=(ipp+1)&1023;
// final accumulation and envelope application
ssample+=sample*env_vol;
}
ssample=ssample/8*master_vol;
ssample*=2.0f*sound_vol;
if(buffer!=NULL)
{
if(ssample>1.0f) ssample=1.0f;
if(ssample<-1.0f) ssample=-1.0f;
*buffer++=ssample;
}
}
return length;
}