Пример #1
0
	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;
	}
Пример #2
0
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;
}
Пример #3
0
Файл: main.cpp Проект: nyov/sfxr
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++;
		}
	}
}
Пример #4
0
Файл: main.cpp Проект: nyov/sfxr
void PlaySample()
{
	ResetSample(false);
	playing_sample=true;
}
Пример #5
0
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++;
		}
	}
}
Пример #6
0
void PlaySample()
{
	ResetSample(false);
	for(int i=0;i<CHANNEL_N;i++)
		channels[i].playing_sample=channels[i].enabled;
}
Пример #7
0
void generator::PlaySample(void)
{
	ResetSample(false);
	playing_sample=true;
}
Пример #8
0
	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;
	}