apps/plugins/midi/synth.c - rockbox - Gitiles

 /***************************************************************************
  *             __________               __   ___.
  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
  *                     \/            \/     \/    \/            \/
  *
  * Copyright (C) 2005 Stepan Moskovchenko
  *
  * All files in this archive are subject to the GNU General Public License.
  * See the file COPYING in the source tree root for full license agreement.
  *
  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  * KIND, either express or implied.
  *
  ****************************************************************************/

 extern struct plugin_api * rb;

 struct Event * getEvent(struct Track * tr, int evNum)
 {
 	return tr->dataBlock + (evNum*sizeof(struct Event));
 }

 void readTextBlock(int file, char * buf)
 {
 	char c = 0;
 	do
 	{
 		c = readChar(file);
 	} while(c == '\n' || c == ' ' || c=='\t');

 	rb->lseek(file, -1, SEEK_CUR);
 	int cp = 0;
 	do
 	{
 		c = readChar(file);
 		buf[cp] = c;
 		cp++;
 	} while (c != '\n' && c != ' ' && c != '\t' && !eof(file));
 	buf[cp-1]=0;
 	rb->lseek(file, -1, SEEK_CUR);
 }


 //Filename is the name of the config file
 //The MIDI file should have been loaded at this point
 int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
 {
 	char patchUsed[128];
 	char drumUsed[128];
 	int a=0;
 	for(a=0; a<MAX_VOICES; a++)
 	{
 		voices[a].cp=0;
 		voices[a].vol=0;
 		voices[a].ch=0;
 		voices[a].isUsed=0;
 		voices[a].note=0;
 	}

 	for(a=0; a<16; a++)
 	{
 		chVol[a]=100; //Default, not quite full blast..
         	chPanLeft[a]=64;   //Center
 		chPanRight[a]=64;   //Center
 		chPat[a]=0;   //Ac Gr Piano
 		chPW[a]=64; // .. not .. bent ?
 	}
 	for(a=0; a<128; a++)
 	{
 		patchSet[a]=NULL;
 		drumSet[a]=NULL;
 		patchUsed[a]=0;
 		drumUsed[a]=0;
 	}

 	//Always load the piano.
 	//Some files will assume its loaded without specifically
 	//issuing a Patch command... then we wonder why we can't hear anything
 	patchUsed[0]=1;

 	//Scan the file to see what needs to be loaded
 	for(a=0; a<mf->numTracks; a++)
 	{
 		int ts=0;

 		if(mf->tracks[a] == NULL)
 		{
 			printf("\nNULL TRACK !!!");
 			rb->splash(HZ*2, true, "Null Track in loader.");
 			return -1;
 		}

 		for(ts=0; ts<mf->tracks[a]->numEvents; ts++)
 		{

 			if((getEvent(mf->tracks[a], ts)->status) == (MIDI_NOTE_ON+9))
 				drumUsed[getEvent(mf->tracks[a], ts)->d1]=1;

 			if( (getEvent(mf->tracks[a], ts)->status & 0xF0) == MIDI_PRGM)
 			{
 				if(patchUsed[getEvent(mf->tracks[a], ts)->d1]==0)
 					printf("\nI need to load patch %d.", getEvent(mf->tracks[a], ts)->d1);
 				patchUsed[getEvent(mf->tracks[a], ts)->d1]=1;
 			}
 		}
 	}

 	int file = rb->open(filename, O_RDONLY);
 	if(file == -1)
 	{
 		rb->splash(HZ*2, true, "Bad patch config.\nDid you install the patchset?");
 		return -1;
 	}

 	char name[40];
 	char fn[40];

 	//Scan our config file and load the right patches as needed
 	int c = 0;
 	rb->snprintf(name, 40, "");
 	for(a=0; a<128; a++)
 	{
 		while(readChar(file)!=' ' && !eof(file));
 		readTextBlock(file, name);

 		rb->snprintf(fn, 40, "/.rockbox/patchset/%s.pat", name);
 		printf("\nLOADING: <%s> ", fn);

 		if(patchUsed[a]==1)
 			patchSet[a]=gusload(fn);

 //		if(patchSet[a] == NULL)
 //			return -1;

 		while((c != '\n'))
 			c = readChar(file);
 	}
 	rb->close(file);

 	file = rb->open(drumConfig, O_RDONLY);
 	if(file == -1)
 	{
 		rb->splash(HZ*2, true, "Bad drum config.\nDid you install the patchset?");
 		return -1;
 	}

 	//Scan our config file and load the drum data
 	int idx=0;
 	char number[30];
 	while(!eof(file))
 	{
 		readTextBlock(file, number);
 		readTextBlock(file, name);
 		rb->snprintf(fn, 40, "/.rockbox/patchset/%s.pat", name);

 		idx = rb->atoi(number);
 		if(idx == 0)
 			break;

 		if(drumUsed[idx]==1)
 			drumSet[idx]=gusload(fn);

 //		if(drumSet[idx] == NULL)
 //			return -1;

 		while((c != '\n') && (c != 255) && (!eof(file)))
 			c = readChar(file);
 	}
 	rb->close(file);
 	return 0;
 }


 int currentVoice IDATA_ATTR;
 struct SynthObject * so IDATA_ATTR;
 struct GWaveform * wf IDATA_ATTR;
 int s IDATA_ATTR;
 short s1 IDATA_ATTR;
 short s2 IDATA_ATTR;
 short sample IDATA_ATTR;	//For synthSample
 unsigned int cpShifted IDATA_ATTR;

 unsigned char b1 IDATA_ATTR;
 unsigned char b2 IDATA_ATTR;


 inline int getSample(int s)
 {

 	//16 bit samples
 	if(wf->mode&1)
 	{

 		if(s<<1 >= wf->wavSize)
 		{
 			printf("\n!!!!! %d \t %d", s<<1, wf->wavSize);
 			return 0;
 		}
 //		signed short a =  ((short *)wf->data)[s];

 		//Sign conversion moved into guspat.c
 		b1=wf->data[s<<1]+((wf->mode & 2) << 6);
 		b2=wf->data[(s<<1)|1]+((wf->mode & 2) << 6);
 		return (b1 | (b2<<8)) ;

 		//Get rid of this sometime
 	}
 	else
 	{       //8-bit samples
 		//Do we even have anything 8-bit in our set?
 		int b1=wf->data[s]+((wf->mode & 2) << 6);
 		return b1<<8;
 	}
 }


 inline void setPoint(struct SynthObject * so, int pt)
 {
 	if(so->ch==9) //Drums, no ADSR
 	{
 		so->curOffset = 1<<27;
 		so->curRate = 1;
 		return;
 	}

 	if(so->wf==NULL)
 	{
 		printf("\nCrap... null waveform...");
 		exit(1);
 	}
 	if(so->wf->envRate==NULL)
 	{
 		printf("\nWaveform has no envelope set");
 		exit(1);
 	}

 	so->curPoint = pt;

 	int r=0;
 	int rate = so->wf->envRate[pt];

 	r=3-((rate>>6) & 0x3);		// Some blatant Timidity code for rate conversion...
 	r*=3;
 	r = (rate & 0x3f) << r;

 	/*
 	Okay. This is the rate shift. Timidity defaults to 9, and sets
 	it to 10 if you use the fast decay option. Slow decay sounds better
 	on some files, except on some other files... you get chords that aren't
 	done decaying yet.. and they dont harmonize with the next chord and it
 	sounds like utter crap. Yes, even Timitidy does that. So I'm going to
 	default this to 10, and maybe later have an option to set it to 9
 	for longer decays.
 	*/
 	so->curRate = r<<10;

 	//Do this here because the patches assume a 44100 sampling rate
 	//We've halved our sampling rate, ergo the ADSR code will be
 	//called half the time. Ergo, double the rate to keep stuff
 	//sounding right.
 	so->curRate = so->curRate << 1;


 	so->targetOffset = so->wf->envOffset[pt]<<(20);
 	if(pt==0)
 		so->curOffset = 0;
 }


 inline void stopVoice(struct SynthObject * so)
 {
 	if(so->state == STATE_RAMPDOWN)
 		return;
 	so->state = STATE_RAMPDOWN;
 	so->decay = 255;

 }


 inline signed short int synthVoice()
 {
 	so = &voices[currentVoice];
 	wf = so->wf;


 	if(so->state != STATE_RAMPDOWN)
 	{
 		so->cp += so->delta;
 	}

 	cpShifted = so->cp >> 10;

 	if( (cpShifted >= (wf->wavSize>>1)) && (so->state != STATE_RAMPDOWN))
 		stopVoice(so);

         s2 = getSample((cpShifted)+1);

 	if((wf->mode & (LOOP_REVERSE|LOOP_PINGPONG)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop>>1)))
 	{
 		if(wf->mode & LOOP_REVERSE)
 		{
 			so->cp = (wf->endLoop)<<9;
 			cpShifted = so->cp >> 10;
 			s2=getSample((cpShifted));
 	        } else
 	        {
 	        	so->delta = -so->delta;
 			so->loopDir = LOOPDIR_FORWARD;
 		}
 	}

 	if((wf->mode & 28) && (so->cp>>10 >= wf->endLoop>>1))
 	{
 		so->loopState = STATE_LOOPING;
 		if((wf->mode & (24)) == 0)
 		{
 			so->cp = (wf->startLoop)<<9;
 			cpShifted = so->cp >> 10;
 			s2=getSample((cpShifted));
 		} else
 		{
 			so->delta = -so->delta;
 			so->loopDir = LOOPDIR_REVERSE;
 		}
 	}

 	//Better, working, linear interpolation
 	s1=getSample((cpShifted));
 	s = s1 + ((signed)((s2 - s1) * (so->cp & 1023))>>10);


 //ADSR COMMENT WOULD GO FROM HERE.........

 	if(so->curRate == 0)
 		stopVoice(so);


 	if(so->ch != 9) //Stupid ADSR code... and don't do ADSR for drums
 	{
 		if(so->curOffset < so->targetOffset)
 		{
 			so->curOffset += (so->curRate);
 			if(so -> curOffset > so->targetOffset && so->curPoint != 2)
 			{
 				if(so->curPoint != 5)
 					setPoint(so, so->curPoint+1);
 				else
 					stopVoice(so);
 			}
 		} else
 		{
 			so->curOffset -= (so->curRate);
 			if(so -> curOffset < so->targetOffset && so->curPoint != 2)
 			{

 				if(so->curPoint != 5)
 					setPoint(so, so->curPoint+1);
 				else
 					stopVoice(so);

 			}
 		}
 	}

 	if(so->curOffset < 0)
 		so->isUsed=0;  //This is OK


 	s = (s * (so->curOffset >> 22) >> 6);

 // ............. TO HERE


 	if(so->state == STATE_RAMPDOWN)
 	{
 		so->decay--;
 		if(so->decay == 0)
 			so->isUsed=0;
 	}

 	s = s * so->decay; s = s >> 10;

 	return s*((signed short int)so->vol*(signed short int)chVol[so->ch])>>14;
 }


 inline void synthSample(int * mixL, int * mixR)
 {
 //	signed int leftMix=0, rightMix=0,
 	*mixL = 0;
 	*mixR = 0;
 	for(currentVoice=0; currentVoice<MAX_VOICES; currentVoice++)
 	{
 		if(voices[currentVoice].isUsed==1)
 		{
 			sample = synthVoice(currentVoice);
 			*mixL += (sample*chPanLeft[voices[currentVoice].ch])>>7;
 			*mixR += (sample*chPanRight[voices[currentVoice].ch])>>7;
 		}
 	}

 	//TODO: Automatic Gain Control, anyone?
 	//Or, should this be implemented on the DSP's output volume instead?
 	return;  //No more ghetto lowpass filter.. linear intrpolation works well.
 }
	/***************************************************************************
	* __________ __ ___.
	* Open \______ \ ____ ____ \| \| _\_ \|__ _______ ___
	* Source \| _// _ \_/ ___\\| \|/ /\| __ \ / _ \ \/ /
	* Jukebox \| \| ( <_> ) \___\| < \| \_\ ( <_> > < <
	* Firmware \|____\|_ /\____/ \___ >__\|_ \\|___ /\____/__/\_ \
	* \/ \/ \/ \/ \/
	*
	* Copyright (C) 2005 Stepan Moskovchenko
	*
	* All files in this archive are subject to the GNU General Public License.
	* See the file COPYING in the source tree root for full license agreement.
	*
	* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
	* KIND, either express or implied.
	*
	****************************************************************************/

	extern struct plugin_api * rb;

	struct Event * getEvent(struct Track * tr, int evNum)
	{
	return tr->dataBlock + (evNum*sizeof(struct Event));
	}

	void readTextBlock(int file, char * buf)
	{
	char c = 0;
	do
	{
	c = readChar(file);
	} while(c == '\n' \|\| c == ' ' \|\| c=='\t');

	rb->lseek(file, -1, SEEK_CUR);
	int cp = 0;
	do
	{
	c = readChar(file);
	buf[cp] = c;
	cp++;
	} while (c != '\n' && c != ' ' && c != '\t' && !eof(file));
	buf[cp-1]=0;
	rb->lseek(file, -1, SEEK_CUR);
	}



	//Filename is the name of the config file
	//The MIDI file should have been loaded at this point
	int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
	{
	char patchUsed[128];
	char drumUsed[128];
	int a=0;
	for(a=0; a<MAX_VOICES; a++)
	{
	voices[a].cp=0;
	voices[a].vol=0;
	voices[a].ch=0;
	voices[a].isUsed=0;
	voices[a].note=0;
	}

	for(a=0; a<16; a++)
	{
	chVol[a]=100; //Default, not quite full blast..
	chPanLeft[a]=64; //Center
	chPanRight[a]=64; //Center
	chPat[a]=0; //Ac Gr Piano
	chPW[a]=64; // .. not .. bent ?
	}
	for(a=0; a<128; a++)
	{
	patchSet[a]=NULL;
	drumSet[a]=NULL;
	patchUsed[a]=0;
	drumUsed[a]=0;
	}

	//Always load the piano.
	//Some files will assume its loaded without specifically
	//issuing a Patch command... then we wonder why we can't hear anything
	patchUsed[0]=1;

	//Scan the file to see what needs to be loaded
	for(a=0; a<mf->numTracks; a++)
	{
	int ts=0;

	if(mf->tracks[a] == NULL)
	{
	printf("\nNULL TRACK !!!");
	rb->splash(HZ*2, true, "Null Track in loader.");
	return -1;
	}

	for(ts=0; ts<mf->tracks[a]->numEvents; ts++)
	{

	if((getEvent(mf->tracks[a], ts)->status) == (MIDI_NOTE_ON+9))
	drumUsed[getEvent(mf->tracks[a], ts)->d1]=1;

	if( (getEvent(mf->tracks[a], ts)->status & 0xF0) == MIDI_PRGM)
	{
	if(patchUsed[getEvent(mf->tracks[a], ts)->d1]==0)
	printf("\nI need to load patch %d.", getEvent(mf->tracks[a], ts)->d1);
	patchUsed[getEvent(mf->tracks[a], ts)->d1]=1;
	}
	}
	}

	int file = rb->open(filename, O_RDONLY);
	if(file == -1)
	{
	rb->splash(HZ*2, true, "Bad patch config.\nDid you install the patchset?");
	return -1;
	}

	char name[40];
	char fn[40];

	//Scan our config file and load the right patches as needed
	int c = 0;
	rb->snprintf(name, 40, "");
	for(a=0; a<128; a++)
	{
	while(readChar(file)!=' ' && !eof(file));
	readTextBlock(file, name);

	rb->snprintf(fn, 40, "/.rockbox/patchset/%s.pat", name);
	printf("\nLOADING: <%s> ", fn);

	if(patchUsed[a]==1)
	patchSet[a]=gusload(fn);

	// if(patchSet[a] == NULL)
	// return -1;

	while((c != '\n'))
	c = readChar(file);
	}
	rb->close(file);

	file = rb->open(drumConfig, O_RDONLY);
	if(file == -1)
	{
	rb->splash(HZ*2, true, "Bad drum config.\nDid you install the patchset?");
	return -1;
	}

	//Scan our config file and load the drum data
	int idx=0;
	char number[30];
	while(!eof(file))
	{
	readTextBlock(file, number);
	readTextBlock(file, name);
	rb->snprintf(fn, 40, "/.rockbox/patchset/%s.pat", name);

	idx = rb->atoi(number);
	if(idx == 0)
	break;

	if(drumUsed[idx]==1)
	drumSet[idx]=gusload(fn);

	// if(drumSet[idx] == NULL)
	// return -1;

	while((c != '\n') && (c != 255) && (!eof(file)))
	c = readChar(file);
	}
	rb->close(file);
	return 0;
	}



	int currentVoice IDATA_ATTR;
	struct SynthObject * so IDATA_ATTR;
	struct GWaveform * wf IDATA_ATTR;
	int s IDATA_ATTR;
	short s1 IDATA_ATTR;
	short s2 IDATA_ATTR;
	short sample IDATA_ATTR; //For synthSample
	unsigned int cpShifted IDATA_ATTR;

	unsigned char b1 IDATA_ATTR;
	unsigned char b2 IDATA_ATTR;


	inline int getSample(int s)
	{

	//16 bit samples
	if(wf->mode&1)
	{

	if(s<<1 >= wf->wavSize)
	{
	printf("\n!!!!! %d \t %d", s<<1, wf->wavSize);
	return 0;
	}
	// signed short a = ((short *)wf->data)[s];

	//Sign conversion moved into guspat.c
	b1=wf->data[s<<1]+((wf->mode & 2) << 6);
	b2=wf->data[(s<<1)\|1]+((wf->mode & 2) << 6);
	return (b1 \| (b2<<8)) ;

	//Get rid of this sometime
	}
	else
	{ //8-bit samples
	//Do we even have anything 8-bit in our set?
	int b1=wf->data[s]+((wf->mode & 2) << 6);
	return b1<<8;
	}
	}




	inline void setPoint(struct SynthObject * so, int pt)
	{
	if(so->ch==9) //Drums, no ADSR
	{
	so->curOffset = 1<<27;
	so->curRate = 1;
	return;
	}

	if(so->wf==NULL)
	{
	printf("\nCrap... null waveform...");
	exit(1);
	}
	if(so->wf->envRate==NULL)
	{
	printf("\nWaveform has no envelope set");
	exit(1);
	}

	so->curPoint = pt;

	int r=0;
	int rate = so->wf->envRate[pt];

	r=3-((rate>>6) & 0x3); // Some blatant Timidity code for rate conversion...
	r*=3;
	r = (rate & 0x3f) << r;

	/*
	Okay. This is the rate shift. Timidity defaults to 9, and sets
	it to 10 if you use the fast decay option. Slow decay sounds better
	on some files, except on some other files... you get chords that aren't
	done decaying yet.. and they dont harmonize with the next chord and it
	sounds like utter crap. Yes, even Timitidy does that. So I'm going to
	default this to 10, and maybe later have an option to set it to 9
	for longer decays.
	*/
	so->curRate = r<<10;

	//Do this here because the patches assume a 44100 sampling rate
	//We've halved our sampling rate, ergo the ADSR code will be
	//called half the time. Ergo, double the rate to keep stuff
	//sounding right.
	so->curRate = so->curRate << 1;


	so->targetOffset = so->wf->envOffset[pt]<<(20);
	if(pt==0)
	so->curOffset = 0;
	}


	inline void stopVoice(struct SynthObject * so)
	{
	if(so->state == STATE_RAMPDOWN)
	return;
	so->state = STATE_RAMPDOWN;
	so->decay = 255;

	}


	inline signed short int synthVoice()
	{
	so = &voices[currentVoice];
	wf = so->wf;


	if(so->state != STATE_RAMPDOWN)
	{
	so->cp += so->delta;
	}

	cpShifted = so->cp >> 10;

	if( (cpShifted >= (wf->wavSize>>1)) && (so->state != STATE_RAMPDOWN))
	stopVoice(so);

	s2 = getSample((cpShifted)+1);

	if((wf->mode & (LOOP_REVERSE\|LOOP_PINGPONG)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop>>1)))
	{
	if(wf->mode & LOOP_REVERSE)
	{
	so->cp = (wf->endLoop)<<9;
	cpShifted = so->cp >> 10;
	s2=getSample((cpShifted));
	} else
	{
	so->delta = -so->delta;
	so->loopDir = LOOPDIR_FORWARD;
	}
	}

	if((wf->mode & 28) && (so->cp>>10 >= wf->endLoop>>1))
	{
	so->loopState = STATE_LOOPING;
	if((wf->mode & (24)) == 0)
	{
	so->cp = (wf->startLoop)<<9;
	cpShifted = so->cp >> 10;
	s2=getSample((cpShifted));
	} else
	{
	so->delta = -so->delta;
	so->loopDir = LOOPDIR_REVERSE;
	}
	}

	//Better, working, linear interpolation
	s1=getSample((cpShifted));
	s = s1 + ((signed)((s2 - s1) * (so->cp & 1023))>>10);


	//ADSR COMMENT WOULD GO FROM HERE.........

	if(so->curRate == 0)
	stopVoice(so);


	if(so->ch != 9) //Stupid ADSR code... and don't do ADSR for drums
	{
	if(so->curOffset < so->targetOffset)
	{
	so->curOffset += (so->curRate);
	if(so -> curOffset > so->targetOffset && so->curPoint != 2)
	{
	if(so->curPoint != 5)
	setPoint(so, so->curPoint+1);
	else
	stopVoice(so);
	}
	} else
	{
	so->curOffset -= (so->curRate);
	if(so -> curOffset < so->targetOffset && so->curPoint != 2)
	{

	if(so->curPoint != 5)
	setPoint(so, so->curPoint+1);
	else
	stopVoice(so);

	}
	}
	}

	if(so->curOffset < 0)
	so->isUsed=0; //This is OK


	s = (s * (so->curOffset >> 22) >> 6);

	// ............. TO HERE


	if(so->state == STATE_RAMPDOWN)
	{
	so->decay--;
	if(so->decay == 0)
	so->isUsed=0;
	}

	s = s * so->decay; s = s >> 10;

	return s((signed short int)so->vol(signed short int)chVol[so->ch])>>14;
	}


	inline void synthSample(int * mixL, int * mixR)
	{
	// signed int leftMix=0, rightMix=0,
	*mixL = 0;
	*mixR = 0;
	for(currentVoice=0; currentVoice<MAX_VOICES; currentVoice++)
	{
	if(voices[currentVoice].isUsed==1)
	{
	sample = synthVoice(currentVoice);
	mixL += (samplechPanLeft[voices[currentVoice].ch])>>7;
	mixR += (samplechPanRight[voices[currentVoice].ch])>>7;
	}
	}

	//TODO: Automatic Gain Control, anyone?
	//Or, should this be implemented on the DSP's output volume instead?
	return; //No more ghetto lowpass filter.. linear intrpolation works well.
	}