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. */
 }