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-rw-r--r--apps/plugins/midi/synth.c650
1 files changed, 319 insertions, 331 deletions
diff --git a/apps/plugins/midi/synth.c b/apps/plugins/midi/synth.c
index bafaf1c7ca..50becf32a4 100644
--- a/apps/plugins/midi/synth.c
+++ b/apps/plugins/midi/synth.c
@@ -20,158 +20,164 @@ extern struct plugin_api * rb;
struct Event * getEvent(struct Track * tr, int evNum)
{
- return tr->dataBlock + (evNum*sizeof(struct Event));
+ 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);
+ 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
+/* 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;
+ 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++)
+ {
+ unsigned 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) /* There was an error loading it */
+ 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) /* Error loading patch */
+ return -1;
+ }
+
+ while((c != '\n') && (c != 255) && (!eof(file)))
+ c = readChar(file);
+ }
+ rb->close(file);
+ return 0;
}
@@ -182,7 +188,7 @@ struct GWaveform * wf IDATA_ATTR;
int s IDATA_ATTR;
short s1 IDATA_ATTR;
short s2 IDATA_ATTR;
-short sample IDATA_ATTR; //For synthSample
+short sample IDATA_ATTR; /* For synthSample */
unsigned int cpShifted IDATA_ATTR;
unsigned char b1 IDATA_ATTR;
@@ -191,31 +197,9 @@ 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;
- }
+ /* Sign conversion moved to guspat.c */
+ /* 8bit conversion NOT YET IMPLEMENTED in guspat.c */
+ return ((short *) wf->data)[s];
}
@@ -223,190 +207,194 @@ inline int getSample(int s)
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;
+ 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;
+ if(so->state == STATE_RAMPDOWN)
+ return;
+ so->state = STATE_RAMPDOWN;
+ so->decay = 255;
}
inline signed short int synthVoice()
{
- so = &voices[currentVoice];
- wf = so->wf;
+ so = &voices[currentVoice];
+ wf = so->wf;
- if(so->state != STATE_RAMPDOWN)
- {
- so->cp += so->delta;
- }
+ if(so->state != STATE_RAMPDOWN)
+ {
+ so->cp += so->delta;
+ }
- cpShifted = so->cp >> 10;
+ cpShifted = so->cp >> 10;
- if( (cpShifted >= (wf->wavSize>>1)) && (so->state != STATE_RAMPDOWN))
- stopVoice(so);
+ if( (cpShifted > (wf->numSamples) && (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;
+ /* LOOP_REVERSE|LOOP_PINGPONG = 24 */
+ if((wf->mode & (24)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop)))
+ {
+ if(wf->mode & LOOP_REVERSE)
+ {
+ so->cp = (wf->endLoop)<<10;
+ cpShifted = wf->endLoop;
+ s2=getSample((cpShifted));
+ } else
+ {
+ so->delta = -so->delta;
+ so->loopDir = LOOPDIR_FORWARD;
+ }
+ }
+
+ if((wf->mode & 28) && (cpShifted >= wf->endLoop))
+ {
+ so->loopState = STATE_LOOPING;
+ if((wf->mode & (24)) == 0)
+ {
+ so->cp = (wf->startLoop)<<10;
+ cpShifted = wf->startLoop;
+ 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 because offset faded it out already */
+
+
+ s = (s * (so->curOffset >> 22) >> 8);
+
+/* ............. TO HERE */
+
+
+ if(so->state == STATE_RAMPDOWN)
+ {
+ so->decay--;
+ if(so->decay == 0)
+ so->isUsed=0;
+ s = (s * so->decay) >> 8;
+ }
+
+ 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.
+ *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. */
}
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