/**
 *******************************************************************************
 * @file appl/generator.h
 * @brief Fx Creator : audio generator (header)
 * @author Edouard DUPIN
 * @date 24/07/2012
 * @par Project
 * fxCreator
 *
 * @par Copyright
 * Copyright 2010 Edouard DUPIN, all right reserved
 *
 * This software is distributed in the hope that it will be useful, but WITHOUT
 * ANY WARRANTY.
 *
 * Licence summary : 
 *    You can modify and redistribute the sources code and binaries.
 *    You can send me the bug-fix
 *    You can not earn money with this Software (if the source extract from Edn
 *        represent less than 50% of original Sources)
 * Term of the licence in in the file licence.txt.
 *
 *******************************************************************************
 */

#include <appl/Debug.h>
#include <appl/generator.h>

#include <stdio.h>
#include <stdarg.h>
#include <stdlib.h>
#include <time.h>
#include <math.h>


#undef __class__
#define __class__	"generator"


#define rnd(n) (rand()%(n+1))

#define PI 3.14159265f

float frnd(float range)
{
	return (float)rnd(10000)/10000*range;
}

int32_t wave_type;

float   p_base_freq;
float   p_freq_limit;
float   p_freq_ramp;
float   p_freq_dramp;
float   p_duty;
float   p_duty_ramp;

float   p_vib_strength;
float   p_vib_speed;
float   p_vib_delay;

float   p_env_attack;
float   p_env_sustain;
float   p_env_decay;
float   p_env_punch;

bool    filter_on;
float   p_lpf_resonance;
float   p_lpf_freq;
float   p_lpf_ramp;
float   p_hpf_freq;
float   p_hpf_ramp;

float   p_pha_offset;
float   p_pha_ramp;

float   p_repeat_speed;

float   p_arp_speed;
float   p_arp_mod;

float   master_vol=0.05f;

float   sound_vol=0.5f;


bool    playing_sample=false;
int32_t phase;
double  fperiod;
double  fmaxperiod;
double  fslide;
double  fdslide;
int32_t period;
float   square_duty;
float   square_slide;
int32_t env_stage;
int32_t env_time;
int32_t env_length[3];
float   env_vol;
float   fphase;
float   fdphase;
int32_t iphase;
float   phaser_buffer[1024];
int32_t ipp;
float   noise_buffer[32];
float   fltp;
float   fltdp;
float   fltw;
float   fltw_d;
float   fltdmp;
float   fltphp;
float   flthp;
float   flthp_d;
float   vib_phase;
float   vib_speed;
float   vib_amp;
int32_t rep_time;
int32_t rep_limit;
int32_t arp_time;
int32_t arp_limit;
double  arp_mod;

float*  vselected=NULL;
int32_t vcurbutton=-1;

int32_t wav_bits=16;
int32_t wav_freq=44100;

int32_t file_sampleswritten;
float   filesample=0.0f;
int32_t fileacc=0;

void ResetParams()
{
	wave_type=0;

	p_base_freq=0.3f;
	p_freq_limit=0.0f;
	p_freq_ramp=0.0f;
	p_freq_dramp=0.0f;
	p_duty=0.0f;
	p_duty_ramp=0.0f;

	p_vib_strength=0.0f;
	p_vib_speed=0.0f;
	p_vib_delay=0.0f;

	p_env_attack=0.0f;
	p_env_sustain=0.3f;
	p_env_decay=0.4f;
	p_env_punch=0.0f;

	filter_on=false;
	p_lpf_resonance=0.0f;
	p_lpf_freq=1.0f;
	p_lpf_ramp=0.0f;
	p_hpf_freq=0.0f;
	p_hpf_ramp=0.0f;
	
	p_pha_offset=0.0f;
	p_pha_ramp=0.0f;

	p_repeat_speed=0.0f;

	p_arp_speed=0.0f;
	p_arp_mod=0.0f;
}

bool LoadSettings(char* filename)
{
	FILE* file=fopen(filename, "rb");
	if(!file)
		return false;

	int version=0;
	fread(&version, 1, sizeof(int), file);
	if(version!=100 && version!=101 && version!=102)
		return false;

	fread(&wave_type, 1, sizeof(int), file);

	sound_vol=0.5f;
	if(version==102)
		fread(&sound_vol, 1, sizeof(float), file);

	fread(&p_base_freq, 1, sizeof(float), file);
	fread(&p_freq_limit, 1, sizeof(float), file);
	fread(&p_freq_ramp, 1, sizeof(float), file);
	if(version>=101)
		fread(&p_freq_dramp, 1, sizeof(float), file);
	fread(&p_duty, 1, sizeof(float), file);
	fread(&p_duty_ramp, 1, sizeof(float), file);

	fread(&p_vib_strength, 1, sizeof(float), file);
	fread(&p_vib_speed, 1, sizeof(float), file);
	fread(&p_vib_delay, 1, sizeof(float), file);

	fread(&p_env_attack, 1, sizeof(float), file);
	fread(&p_env_sustain, 1, sizeof(float), file);
	fread(&p_env_decay, 1, sizeof(float), file);
	fread(&p_env_punch, 1, sizeof(float), file);

	fread(&filter_on, 1, sizeof(bool), file);
	fread(&p_lpf_resonance, 1, sizeof(float), file);
	fread(&p_lpf_freq, 1, sizeof(float), file);
	fread(&p_lpf_ramp, 1, sizeof(float), file);
	fread(&p_hpf_freq, 1, sizeof(float), file);
	fread(&p_hpf_ramp, 1, sizeof(float), file);
	
	fread(&p_pha_offset, 1, sizeof(float), file);
	fread(&p_pha_ramp, 1, sizeof(float), file);

	fread(&p_repeat_speed, 1, sizeof(float), file);

	if(version>=101)
	{
		fread(&p_arp_speed, 1, sizeof(float), file);
		fread(&p_arp_mod, 1, sizeof(float), file);
	}

	fclose(file);
	return true;
}

bool SaveSettings(char* filename)
{
	FILE* file=fopen(filename, "wb");
	if(!file)
		return false;

	int version=102;
	fwrite(&version, 1, sizeof(int), file);

	fwrite(&wave_type, 1, sizeof(int), file);

	fwrite(&sound_vol, 1, sizeof(float), file);

	fwrite(&p_base_freq, 1, sizeof(float), file);
	fwrite(&p_freq_limit, 1, sizeof(float), file);
	fwrite(&p_freq_ramp, 1, sizeof(float), file);
	fwrite(&p_freq_dramp, 1, sizeof(float), file);
	fwrite(&p_duty, 1, sizeof(float), file);
	fwrite(&p_duty_ramp, 1, sizeof(float), file);

	fwrite(&p_vib_strength, 1, sizeof(float), file);
	fwrite(&p_vib_speed, 1, sizeof(float), file);
	fwrite(&p_vib_delay, 1, sizeof(float), file);

	fwrite(&p_env_attack, 1, sizeof(float), file);
	fwrite(&p_env_sustain, 1, sizeof(float), file);
	fwrite(&p_env_decay, 1, sizeof(float), file);
	fwrite(&p_env_punch, 1, sizeof(float), file);

	fwrite(&filter_on, 1, sizeof(bool), file);
	fwrite(&p_lpf_resonance, 1, sizeof(float), file);
	fwrite(&p_lpf_freq, 1, sizeof(float), file);
	fwrite(&p_lpf_ramp, 1, sizeof(float), file);
	fwrite(&p_hpf_freq, 1, sizeof(float), file);
	fwrite(&p_hpf_ramp, 1, sizeof(float), file);
	
	fwrite(&p_pha_offset, 1, sizeof(float), file);
	fwrite(&p_pha_ramp, 1, sizeof(float), file);

	fwrite(&p_repeat_speed, 1, sizeof(float), file);

	fwrite(&p_arp_speed, 1, sizeof(float), file);
	fwrite(&p_arp_mod, 1, sizeof(float), file);

	fclose(file);
	return true;
}

void ResetSample(bool restart)
{
	if(!restart) {
		phase=0;
	}
	fperiod=100.0/(p_base_freq*p_base_freq+0.001);
	period=(int)fperiod;
	fmaxperiod=100.0/(p_freq_limit*p_freq_limit+0.001);
	fslide=1.0-pow((double)p_freq_ramp, 3.0)*0.01;
	fdslide=-pow((double)p_freq_dramp, 3.0)*0.000001;
	square_duty=0.5f-p_duty*0.5f;
	square_slide=-p_duty_ramp*0.00005f;
	if(p_arp_mod>=0.0f) {
		arp_mod=1.0-pow((double)p_arp_mod, 2.0)*0.9;
	} else {
		arp_mod=1.0+pow((double)p_arp_mod, 2.0)*10.0;
	}
	arp_time=0;
	arp_limit=(int)(pow(1.0f-p_arp_speed, 2.0f)*20000+32);
	if(p_arp_speed==1.0f) {
		arp_limit=0;
	}
	if(!restart) {
		// reset filter
		fltp=0.0f;
		fltdp=0.0f;
		fltw=pow(p_lpf_freq, 3.0f)*0.1f;
		fltw_d=1.0f+p_lpf_ramp*0.0001f;
		fltdmp=5.0f/(1.0f+pow(p_lpf_resonance, 2.0f)*20.0f)*(0.01f+fltw);
		if(fltdmp>0.8f) {
			fltdmp=0.8f;
		}
		fltphp=0.0f;
		flthp=pow(p_hpf_freq, 2.0f)*0.1f;
		flthp_d=1.0+p_hpf_ramp*0.0003f;
		// reset vibrato
		vib_phase=0.0f;
		vib_speed=pow(p_vib_speed, 2.0f)*0.01f;
		vib_amp=p_vib_strength*0.5f;
		// reset envelope
		env_vol=0.0f;
		env_stage=0;
		env_time=0;
		env_length[0]=(int)(p_env_attack*p_env_attack*100000.0f);
		env_length[1]=(int)(p_env_sustain*p_env_sustain*100000.0f);
		env_length[2]=(int)(p_env_decay*p_env_decay*100000.0f);

		fphase=pow(p_pha_offset, 2.0f)*1020.0f;
		if(p_pha_offset<0.0f) {
			fphase=-fphase;
		}
		fdphase=pow(p_pha_ramp, 2.0f)*1.0f;
		if(p_pha_ramp<0.0f) {
			fdphase=-fdphase;
		}
		iphase=abs((int)fphase);
		ipp=0;
		for(int i=0;i<1024;i++) {
			phaser_buffer[i]=0.0f;
		}

		for(int i=0;i<32;i++) {
			noise_buffer[i]=frnd(2.0f)-1.0f;
		}

		rep_time=0;
		rep_limit=(int)(pow(1.0f-p_repeat_speed, 2.0f)*20000+32);
		if(p_repeat_speed==0.0f) {
			rep_limit=0;
		}
	}
}

void PlaySample()
{
	ResetSample(false);
	playing_sample=true;
}

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%=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++;
		}
	}
}


//lets use SDL in stead
static void SDLAudioCallback(int16_t *stream, int32_t len)
{
	if (playing_sample) {
		float fbuf[len];
		memset(fbuf, 0, sizeof(int32_t));
		SynthSample(len, fbuf, NULL);
		while (len--) {
			float f = fbuf[len];
			if (f < -1.0) {
				f = -1.0;
			}
			if (f > 1.0) {
				f = 1.0;
			}
			stream[len] = (int16_t)(f * 32767);
		}
	} else {
		memset(stream, 0, len*sizeof(int16_t));
	}
}

bool ExportWAV(char* filename)
{
	FILE* foutput=fopen(filename, "wb");
	if(!foutput) {
		return false;
	}
	// write wav header
	uint32_t dword=0;
	uint16_t word=0;
	fwrite("RIFF", 4, 1, foutput); // "RIFF"
	dword=0;
	fwrite(&dword, 1, 4, foutput); // remaining file size
	fwrite("WAVE", 4, 1, foutput); // "WAVE"

	fwrite("fmt ", 4, 1, foutput); // "fmt "
	dword=16;
	fwrite(&dword, 1, 4, foutput); // chunk size
	word=1;
	fwrite(&word, 1, 2, foutput); // compression code
	word=1;
	fwrite(&word, 1, 2, foutput); // channels
	dword=wav_freq;
	fwrite(&dword, 1, 4, foutput); // sample rate
	dword=wav_freq*wav_bits/8;
	fwrite(&dword, 1, 4, foutput); // bytes/sec
	word=wav_bits/8;
	fwrite(&word, 1, 2, foutput); // block align
	word=wav_bits;
	fwrite(&word, 1, 2, foutput); // bits per sample

	fwrite("data", 4, 1, foutput); // "data"
	dword=0;
	int foutstream_datasize=ftell(foutput);
	fwrite(&dword, 1, 4, foutput); // chunk size

	// write sample data
	file_sampleswritten=0;
	filesample=0.0f;
	fileacc=0;
	PlaySample();
	while(playing_sample) {
		SynthSample(256, NULL, foutput);
	}

	// seek back to header and write size info
	fseek(foutput, 4, SEEK_SET);
	dword=0;
	dword=foutstream_datasize-4+file_sampleswritten*wav_bits/8;
	fwrite(&dword, 1, 4, foutput); // remaining file size
	fseek(foutput, foutstream_datasize, SEEK_SET);
	dword=file_sampleswritten*wav_bits/8;
	fwrite(&dword, 1, 4, foutput); // chunk size (data)
	fclose(foutput);
	
	return true;
}