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Fixed warnings, adapted to Rockbox coding style, optimized to 78% realtime.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6329 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Stepan Moskovchenko 2005-04-20 21:07:13 +00:00
parent 74e9d545ef
commit 9ec1ff8cf5
7 changed files with 1056 additions and 1115 deletions
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65
apps/plugins/midi/guspat.c
View file

@ -67,9 +67,12 @@ struct GWaveform * loadWaveform(int file)
wav->data=readData(file, wav->wavSize); wav->data=readData(file, wav->wavSize);
wav->numSamples = wav->wavSize / 2; wav->numSamples = wav->wavSize / 2;
int a=0; wav->startLoop = wav->startLoop >> 1;
wav->endLoop = wav->endLoop >> 1;
unsigned int a=0;
return wav; /* half baked 8 bit conversion UNFINISHED*/
/*
if(wav->mode & 1 == 0) //Whoops, 8 bit if(wav->mode & 1 == 0) //Whoops, 8 bit
{ {
wav->numSamples = wav->wavSize; wav->numSamples = wav->wavSize;
@ -81,7 +84,7 @@ struct GWaveform * loadWaveform(int file)
for(a=0; a<1000; a++) for(a=0; a<1000; a++)
printf("\n!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!"); printf("\n!!!!!!!!!!!!!!!!!!!!!!!!!!!");
for(a=wav->wavSize-1; a>0; a-=2) for(a=wav->wavSize-1; a>0; a-=2)
@ -91,39 +94,26 @@ struct GWaveform * loadWaveform(int file)
// int b1=wf->data[s]+((wf->mode & 2) << 6); // int b1=wf->data[s]+((wf->mode & 2) << 6);
// return b1<<8; // return b1<<8;
} }
*/
/*
//#if !defined(SIMULATOR) /* Iriver needs byteswapping.. big endian, go figure. Gus files are little endian */
for(a=0; a<wav->wavSize; a+=2)
#if !defined(SIMULATOR)
for(a=0; a<wav->numSamples; a++)
{ {
unsigned char tmp; ((unsigned short *) wav->data)[a] = SWAB16(((unsigned short *) wav->data)[a]);
tmp = wav->data[2*a];
wav->data[2*a] = wav->data[2*a+1];
wav->data[2*a+1] = tmp;
} }
//#endif #endif
/* Convert unsigned to signed by subtracting 32768 */
if(wav->mode & 2) if(wav->mode & 2)
{ {
for(a=0; a<wav->wavSize/2; a++) for(a=0; a<wav->numSamples; a++)
{ ((short *) wav->data)[a] = ((unsigned short *) wav->data)[a] - 32768;
((short *) wav->data)[a] = ((short *) wav->data)[a] - 32767;
}
}
*/
//If we have a 16 bit waveform
/* if(wav->mode & 1 && (wav->mode & 2))
{
for(a=0; a<wav->wavSize; a+=2) //Convert it to
{
wav->data[a]=wav->data[a]+(1 << 7);
wav->data[a|1]=wav->data[(a)|1]+(1 << 7);
} }
}
*/
return wav; return wav;
} }
@ -131,8 +121,9 @@ struct GWaveform * loadWaveform(int file)
int selectWaveform(struct GPatch * pat, int midiNote) int selectWaveform(struct GPatch * pat, int midiNote)
{ {
int tabFreq = gustable[midiNote]/100; //Comparison /* We divide by 100 here because everyone's freq formula is slightly different */
int a=0; unsigned int tabFreq = gustable[midiNote]/100; /* Comparison */
unsigned int a=0;
for(a=0; a<pat->numWaveforms; a++) for(a=0; a<pat->numWaveforms; a++)
{ {
if(pat->waveforms[a]->lowFreq/100 <= tabFreq && if(pat->waveforms[a]->lowFreq/100 <= tabFreq &&
@ -166,7 +157,7 @@ struct GPatch * gusload(char * filename)
gp->inst=readChar(file); gp->inst=readChar(file);
gp->voc=readChar(file); gp->voc=readChar(file);
gp->chan=readChar(file); gp->chan=readChar(file);
gp->numWaveforms=readWord(file); //readWord(file); gp->numWaveforms=readWord(file);
gp->vol=readWord(file); gp->vol=readWord(file);
gp->datSize=readDWord(file); gp->datSize=readDWord(file);
gp->res=readData(file, 36); gp->res=readData(file, 36);
@ -184,19 +175,7 @@ struct GPatch * gusload(char * filename)
gp->numWaves=readChar(file); gp->numWaves=readChar(file);
gp->layerRes=readData(file,40); gp->layerRes=readData(file,40);
/* printf("\n%s\n%s\n%s", gp->header, gp->gravisid, gp->desc);
printf("\nInst = %d", gp->inst);
printf("\nVoc = %d", gp->voc);
printf("\nChan = %d", gp->chan);
printf("\nWav = %d", gp->numWaveforms);
printf("\nVol = %d", gp->vol);
printf("\nSize = %d", gp->datSize);
printf("\n\ninstrID = %d", gp->instrID);
printf("\ninstrName = %s", gp->instrName);
// printf("\ninstrSize = %d", gp->instrSize);
// printf("\nlayers = %d", gp->layers);
*/
printf("\nFILE: %s", filename); printf("\nFILE: %s", filename);
printf("\nlayerSamples=%d", gp->numWaves); printf("\nlayerSamples=%d", gp->numWaves);

2
apps/plugins/midi/guspat.h
View file

@ -16,7 +16,7 @@
* *
****************************************************************************/ ****************************************************************************/
//This came from one of the Gravis documents /* This came from one of the Gravis documents */
static const unsigned int gustable[]= static const unsigned int gustable[]=
{ {
8175, 8661, 9177, 9722, 10300, 10913, 11562, 12249, 12978, 13750, 14567, 15433, 8175, 8661, 9177, 9722, 10300, 10913, 11562, 12249, 12978, 13750, 14567, 15433,

37
apps/plugins/midi/midifile.c
View file

@ -29,7 +29,7 @@ struct MIDIfile * loadFile(char * filename)
struct MIDIfile * mf; struct MIDIfile * mf;
int file = rb->open (filename, O_RDONLY); int file = rb->open (filename, O_RDONLY);
if(file==0) if(file==-1)
{ {
bail("Could not open file\n"); bail("Could not open file\n");
} }
@ -69,8 +69,6 @@ struct MIDIfile * loadFile(char * filename)
printf("\nnumTracks=%d div=%d\nBegin reading track data\n", mf->numTracks, mf->div); printf("\nnumTracks=%d div=%d\nBegin reading track data\n", mf->numTracks, mf->div);
// return;
while(! eof(file) && track < mf->numTracks) while(! eof(file) && track < mf->numTracks)
{ {
@ -107,21 +105,21 @@ struct MIDIfile * loadFile(char * filename)
int rStatus = 0; int rStatus = 0;
//Returns 0 if done, 1 if keep going /* Returns 0 if done, 1 if keep going */
int readEvent(int file, void * dest) int readEvent(int file, void * dest)
{ {
struct Event dummy; struct Event dummy;
struct Event * ev = (struct Event *) dest; struct Event * ev = (struct Event *) dest;
if(ev == NULL) if(ev == NULL)
ev = &dummy; //If we are just counting events instead of loading them ev = &dummy; /* If we are just counting events instead of loading them */
ev->delta = readVarData(file); ev->delta = readVarData(file);
int t=readChar(file); int t=readChar(file);
if((t&0x80) == 0x80) //if not a running status event if((t&0x80) == 0x80) /* if not a running status event */
{ {
ev->status = t; ev->status = t;
if(t == 0xFF) if(t == 0xFF)
@ -129,7 +127,7 @@ int readEvent(int file, void * dest)
ev->d1 = readChar(file); ev->d1 = readChar(file);
ev->len = readVarData(file); ev->len = readVarData(file);
//Allocate and read in the data block /* Allocate and read in the data block */
if(dest != NULL) if(dest != NULL)
{ {
ev->evData = readData(file, ev->len); ev->evData = readData(file, ev->len);
@ -137,19 +135,20 @@ int readEvent(int file, void * dest)
} }
else else
{ {
//Don't allocate anything, just see how much it would tale /*
//To make memory usage efficient * Don't allocate anything, just see how much it would tale
* To make memory usage efficient
int a=0; */
unsigned int a=0;
for(a=0; a<ev->len; a++) for(a=0; a<ev->len; a++)
readChar(file); //Skip skip readChar(file); //Skip skip
} }
if(ev->d1 == 0x2F) if(ev->d1 == 0x2F)
{ {
return 0; //Termination meta-event return 0; /* Termination meta-event */
} }
} else //If part of a running status event } else /* If part of a running status event */
{ {
rStatus = t; rStatus = t;
ev->status = t; ev->status = t;
@ -161,7 +160,7 @@ int readEvent(int file, void * dest)
} else } else
ev->d2 = 127; ev->d2 = 127;
} }
} else //Running Status } else /* Running Status */
{ {
ev->status = rStatus; ev->status = rStatus;
ev->d1 = t; ev->d1 = t;
@ -189,9 +188,9 @@ struct Track * readTrack(int file)
int pos = rb->lseek(file, 0, SEEK_CUR); int pos = rb->lseek(file, 0, SEEK_CUR);
while(readEvent(file, NULL)) //Memory saving technique while(readEvent(file, NULL)) /* Memory saving technique */
numEvents++; //Attempt to read in events, count how many numEvents++; /* Attempt to read in events, count how many */
//THEN allocate memory and read them in /* THEN allocate memory and read them in */
rb->lseek(file, pos, SEEK_SET); rb->lseek(file, pos, SEEK_SET);
int trackSize = (numEvents+1) * sizeof(struct Event); int trackSize = (numEvents+1) * sizeof(struct Event);
@ -252,7 +251,7 @@ int readTwoBytes(int file)
return data; return data;
} }
//This came from the MIDI file format guide /* This came from the MIDI file format guide */
int readVarData(int file) int readVarData(int file)
{ {
unsigned int value; unsigned int value;
@ -270,8 +269,6 @@ int readVarData(int file)
/* /*
//This function should not be needed because we
//can just release the whole memory buffer at once
void unloadFile(struct MIDIfile * mf) void unloadFile(struct MIDIfile * mf)
{ {
if(mf == NULL) if(mf == NULL)

36
apps/plugins/midi/midiutil.c
View file

@ -63,27 +63,17 @@
extern struct plugin_api * rb; extern struct plugin_api * rb;
int chVol[16] IDATA_ATTR; //Channel volume int chVol[16] IDATA_ATTR; /* Channel volume */
int chPanLeft[16] IDATA_ATTR; //Channel panning int chPanLeft[16] IDATA_ATTR; /* Channel panning */
int chPanRight[16] IDATA_ATTR; int chPanRight[16] IDATA_ATTR;
int chPat[16]; //Channel patch int chPat[16]; /* Channel patch */
int chPW[16]; //Channel pitch wheel, MSB only int chPW[16]; /* Channel pitch wheel, MSB only */
/*
unsigned char chVol[16]; //Channel volume
unsigned char chPanLeft[16]; //Channel panning
unsigned char chPanRight[16];
unsigned char chPat[16]; //Channel patch
unsigned char chPW[16]; //Channel pitch wheel, MSB only
*/
struct GPatch * gusload(char *); struct GPatch * gusload(char *);
struct GPatch * patchSet[128]; struct GPatch * patchSet[128];
struct GPatch * drumSet[128]; struct GPatch * drumSet[128];
//char myarray[80] IDATA_ATTR;
struct Event struct Event
{ {
unsigned int delta; unsigned int delta;
@ -96,8 +86,8 @@ struct Track
{ {
unsigned int size; unsigned int size;
unsigned int numEvents; unsigned int numEvents;
unsigned int delta; //For sequencing unsigned int delta; /* For sequencing */
unsigned int pos; //For sequencing unsigned int pos; /* For sequencing */
void * dataBlock; void * dataBlock;
}; };
@ -105,18 +95,16 @@ struct Track
struct MIDIfile struct MIDIfile
{ {
int Length; int Length;
//int Format; //We don't really care what type it is
unsigned short numTracks; unsigned short numTracks;
unsigned short div; //Time division, X ticks per millisecond unsigned short div; /* Time division, X ticks per millisecond */
struct Track * tracks[48]; struct Track * tracks[48];
unsigned char patches[128]; unsigned char patches[128];
int numPatches; int numPatches;
}; };
/* /*
struct SynthObject struct SynthObject
{ {
// int tmp;
struct GWaveform * wf; struct GWaveform * wf;
unsigned int delta; unsigned int delta;
unsigned int decay; unsigned int decay;
@ -127,13 +115,13 @@ struct SynthObject
unsigned int curPoint; unsigned int curPoint;
}; };
*/ */
struct SynthObject struct SynthObject
{ {
// int tmp;
struct GWaveform * wf; struct GWaveform * wf;
int delta; int delta;
int decay; int decay;
int cp; unsigned int cp;
int state, loopState, loopDir; int state, loopState, loopDir;
int note, vol, ch, isUsed; int note, vol, ch, isUsed;
int curRate, curOffset, targetOffset; int curRate, curOffset, targetOffset;
@ -183,7 +171,7 @@ void *alloc(int size)
} }
//Rick's code /* Rick's code */
/* /*
void *alloc(int size) void *alloc(int size)
{ {
@ -242,5 +230,5 @@ void printf(char *fmt, ...) {fmt=fmt; }
void exit(int code) void exit(int code)
{ {
code = code; //Stub function, kill warning for now code = code; /* Stub function, kill warning for now */
} }

56
apps/plugins/midi/sequencer.c
View file

@ -44,10 +44,10 @@ void setPatch(int ch, int pat)
/* /*
Pitch Bend table, Computed by * Pitch Bend table, Computed by
for i=0:127, fprintf('%d,', round(2^16*2^((i-64)/384))); end * for i=0:127, fprintf('%d,', round(2^16*2^((i-64)/384))); end
(When typed into Matlab) * (When typed into Matlab)
16 bit pitch bend table * 16 bit pitch bend table
*/ */
long pitchTbl[]= long pitchTbl[]=
{ {
@ -67,8 +67,8 @@ void findDelta(struct SynthObject * so, int ch, int note)
struct GWaveform * wf = patchSet[chPat[ch]]->waveforms[patchSet[chPat[ch]]->noteTable[note]]; struct GWaveform * wf = patchSet[chPat[ch]]->waveforms[patchSet[chPat[ch]]->noteTable[note]];
so->wf=wf; so->wf=wf;
so->delta = (((gustable[note]<<10) / wf->rootFreq) * wf->sampRate / SAMPLE_RATE); so->delta = (((gustable[note]<<10) / (wf->rootFreq)) * wf->sampRate / (SAMPLE_RATE));
so->delta = so->delta * pitchTbl[chPW[ch]] >> 16; so->delta = (so->delta * pitchTbl[chPW[ch]])>> 16;
} }
void setPW(int ch, int msb) void setPW(int ch, int msb)
@ -88,12 +88,14 @@ void setPW(int ch, int msb)
void pressNote(int ch, int note, int vol) void pressNote(int ch, int note, int vol)
{ {
//Silences all channels but one, for easy debugging, for me.
/* /*
if(ch == 0) return; if(ch == 0) return;
// if(ch == 1) return; if(ch == 1) return;
if(ch == 2) return; if(ch == 2) return;
if(ch == 3) return; if(ch == 3) return;
if(ch == 4) return; // if(ch == 4) return;
if(ch == 5) return; if(ch == 5) return;
if(ch == 6) return; if(ch == 6) return;
if(ch == 7) return; if(ch == 7) return;
@ -121,40 +123,29 @@ void pressNote(int ch, int note, int vol)
printf("\nVOICE DUMP: "); printf("\nVOICE DUMP: ");
for(a=0; a<48; a++) for(a=0; a<48; a++)
printf("\n#%d Ch=%d Note=%d curRate=%d curOffset=%d curPoint=%d targetOffset=%d", a, voices[a].ch, voices[a].note, voices[a].curRate, voices[a].curOffset, voices[a].curPoint, voices[a].targetOffset); printf("\n#%d Ch=%d Note=%d curRate=%d curOffset=%d curPoint=%d targetOffset=%d", a, voices[a].ch, voices[a].note, voices[a].curRate, voices[a].curOffset, voices[a].curPoint, voices[a].targetOffset);
return; //None avaolable return; /* None available */
} }
voices[a].ch=ch; voices[a].ch=ch;
voices[a].note=note; voices[a].note=note;
voices[a].vol=vol; voices[a].vol=vol;
voices[a].cp=0; voices[a].cp=0;
voices[a].state=STATE_ATTACK; voices[a].state=STATE_ATTACK;
// voices[a].pstate=STATE_ATTACK;
voices[a].decay=255; voices[a].decay=255;
voices[a].loopState=STATE_NONLOOPING; voices[a].loopState=STATE_NONLOOPING;
voices[a].loopDir = LOOPDIR_FORWARD; voices[a].loopDir = LOOPDIR_FORWARD;
/* /*
OKAY. Gt = Gus Table value * OKAY. Gt = Gus Table value
rf = Root Frequency of wave * rf = Root Frequency of wave
SR = sound sampling rate * SR = sound sampling rate
sr = WAVE sampling rate * sr = WAVE sampling rate
*/ */
/*
unsigned int gt = gustable[note];
unsigned int rf = wf->rootFreq;
unsigned int SR = SAMPLE_RATE;
unsigned int sr = wf->sampRate;
voices[a].delta=((((gt<<10) / rf) * sr / SR));
*/
if(ch!=9) if(ch!=9)
{ {
findDelta(&voices[a], ch, note); findDelta(&voices[a], ch, note);
//Turn it on /* Turn it on */
voices[a].isUsed=1; voices[a].isUsed=1;
setPoint(&voices[a], 0); setPoint(&voices[a], 0);
} else } else
@ -170,7 +161,8 @@ void pressNote(int ch, int note, int vol)
if(wf->mode & 28) if(wf->mode & 28)
printf("\nWoah, a drum patch has a loop. Stripping the loop..."); printf("\nWoah, a drum patch has a loop. Stripping the loop...");
wf->mode = wf->mode & (255-28); wf->mode = wf->mode & (255-28);
//Turn it on
/* Turn it on */
voices[a].isUsed=1; voices[a].isUsed=1;
setPoint(&voices[a], 0); setPoint(&voices[a], 0);
@ -184,8 +176,9 @@ void pressNote(int ch, int note, int vol)
void releaseNote(int ch, int note) void releaseNote(int ch, int note)
{ {
if(ch==9) // && note != 27 && note != 31 && note != 28) if(ch==9)
return; return;
int a=0; int a=0;
for(a=0; a<MAX_VOICES; a++) for(a=0; a<MAX_VOICES; a++)
{ {
@ -193,10 +186,6 @@ void releaseNote(int ch, int note)
{ {
if((voices[a].wf->mode & 28)) if((voices[a].wf->mode & 28))
{ {
// voices[a].tmp=40;
// voices[a].state = STATE_RELEASE; //Ramp down
// voices[a].state = STATE_RAMPDOWN; //Ramp down
setPoint(&voices[a], 3); setPoint(&voices[a], 3);
} }
} }
@ -205,8 +194,6 @@ void releaseNote(int ch, int note)
void sendEvent(struct Event * ev) void sendEvent(struct Event * ev)
{ {
// printf("\nEVENT S=%2x D1=%2x D2=%2x", ev->status, ev->d1, ev->d2);
if( ((ev->status & 0xF0) == MIDI_CONTROL) && (ev->d1 == CTRL_VOLUME) ) if( ((ev->status & 0xF0) == MIDI_CONTROL) && (ev->d1 == CTRL_VOLUME) )
{ {
setVol((ev->status & 0xF), ev->d2); setVol((ev->status & 0xF), ev->d2);
@ -231,7 +218,7 @@ void sendEvent(struct Event * ev)
return; return;
} }
if(((ev->status & 0xF0) == MIDI_NOTE_ON) && (ev->d2 == 0)) //Release by vol=0 if(((ev->status & 0xF0) == MIDI_NOTE_ON) && (ev->d2 == 0)) /* Release by vol=0 */
{ {
releaseNote(ev->status & 0x0F, ev->d1); releaseNote(ev->status & 0x0F, ev->d1);
return; return;
@ -282,7 +269,6 @@ int tick(struct MIDIfile * mf)
tracksAdv++; tracksAdv++;
while(getEvent(tr, tr->pos)->delta <= tr->delta) while(getEvent(tr, tr->pos)->delta <= tr->delta)
{ {
// printf("\nDelta = %d", tr->delta);
struct Event * e = getEvent(tr, tr->pos); struct Event * e = getEvent(tr, tr->pos);
if(e->status != 0xFF) if(e->status != 0xFF)

140
apps/plugins/midi/synth.c
View file

@ -45,8 +45,8 @@ void readTextBlock(int file, char * buf)
//Filename is the name of the config file /* Filename is the name of the config file */
//The MIDI file should have been loaded at this point /* The MIDI file should have been loaded at this point */
int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig) int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
{ {
char patchUsed[128]; char patchUsed[128];
@ -63,11 +63,11 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
for(a=0; a<16; a++) for(a=0; a<16; a++)
{ {
chVol[a]=100; //Default, not quite full blast.. chVol[a]=100; /* Default, not quite full blast.. */
chPanLeft[a]=64; //Center chPanLeft[a]=64; /* Center */
chPanRight[a]=64; //Center chPanRight[a]=64; /* Center */
chPat[a]=0; //Ac Gr Piano chPat[a]=0; /* Ac Gr Piano */
chPW[a]=64; // .. not .. bent ? chPW[a]=64; /* .. not .. bent ? */
} }
for(a=0; a<128; a++) for(a=0; a<128; a++)
{ {
@ -77,15 +77,17 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
drumUsed[a]=0; drumUsed[a]=0;
} }
//Always load the piano. /*
//Some files will assume its loaded without specifically * Always load the piano.
//issuing a Patch command... then we wonder why we can't hear anything * Some files will assume its loaded without specifically
* issuing a Patch command... then we wonder why we can't hear anything
*/
patchUsed[0]=1; patchUsed[0]=1;
//Scan the file to see what needs to be loaded /* Scan the file to see what needs to be loaded */
for(a=0; a<mf->numTracks; a++) for(a=0; a<mf->numTracks; a++)
{ {
int ts=0; unsigned int ts=0;
if(mf->tracks[a] == NULL) if(mf->tracks[a] == NULL)
{ {
@ -119,7 +121,7 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
char name[40]; char name[40];
char fn[40]; char fn[40];
//Scan our config file and load the right patches as needed /* Scan our config file and load the right patches as needed */
int c = 0; int c = 0;
rb->snprintf(name, 40, ""); rb->snprintf(name, 40, "");
for(a=0; a<128; a++) for(a=0; a<128; a++)
@ -131,10 +133,12 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
printf("\nLOADING: <%s> ", fn); printf("\nLOADING: <%s> ", fn);
if(patchUsed[a]==1) if(patchUsed[a]==1)
{
patchSet[a]=gusload(fn); patchSet[a]=gusload(fn);
// if(patchSet[a] == NULL) if(patchSet[a] == NULL) /* There was an error loading it */
// return -1; return -1;
}
while((c != '\n')) while((c != '\n'))
c = readChar(file); c = readChar(file);
@ -148,7 +152,7 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
return -1; return -1;
} }
//Scan our config file and load the drum data /* Scan our config file and load the drum data */
int idx=0; int idx=0;
char number[30]; char number[30];
while(!eof(file)) while(!eof(file))
@ -162,10 +166,12 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
break; break;
if(drumUsed[idx]==1) if(drumUsed[idx]==1)
{
drumSet[idx]=gusload(fn); drumSet[idx]=gusload(fn);
// if(drumSet[idx] == NULL) if(drumSet[idx] == NULL) /* Error loading patch */
// return -1; return -1;
}
while((c != '\n') && (c != 255) && (!eof(file))) while((c != '\n') && (c != 255) && (!eof(file)))
c = readChar(file); c = readChar(file);
@ -182,7 +188,7 @@ struct GWaveform * wf IDATA_ATTR;
int s IDATA_ATTR; int s IDATA_ATTR;
short s1 IDATA_ATTR; short s1 IDATA_ATTR;
short s2 IDATA_ATTR; short s2 IDATA_ATTR;
short sample IDATA_ATTR; //For synthSample short sample IDATA_ATTR; /* For synthSample */
unsigned int cpShifted IDATA_ATTR; unsigned int cpShifted IDATA_ATTR;
unsigned char b1 IDATA_ATTR; unsigned char b1 IDATA_ATTR;
@ -191,31 +197,9 @@ unsigned char b2 IDATA_ATTR;
inline int getSample(int s) inline int getSample(int s)
{ {
/* Sign conversion moved to guspat.c */
//16 bit samples /* 8bit conversion NOT YET IMPLEMENTED in guspat.c */
if(wf->mode&1) return ((short *) wf->data)[s];
{
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;
}
} }
@ -223,7 +207,7 @@ inline int getSample(int s)
inline void setPoint(struct SynthObject * so, int pt) inline void setPoint(struct SynthObject * so, int pt)
{ {
if(so->ch==9) //Drums, no ADSR if(so->ch==9) /* Drums, no ADSR */
{ {
so->curOffset = 1<<27; so->curOffset = 1<<27;
so->curRate = 1; so->curRate = 1;
@ -246,25 +230,27 @@ inline void setPoint(struct SynthObject * so, int pt)
int r=0; int r=0;
int rate = so->wf->envRate[pt]; int rate = so->wf->envRate[pt];
r=3-((rate>>6) & 0x3); // Some blatant Timidity code for rate conversion... r=3-((rate>>6) & 0x3); /* Some blatant Timidity code for rate conversion... */
r*=3; r*=3;
r = (rate & 0x3f) << r; r = (rate & 0x3f) << r;
/* /*
Okay. This is the rate shift. Timidity defaults to 9, and sets * 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 * 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 * 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 * 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 * 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 * default this to 10, and maybe later have an option to set it to 9
for longer decays. * for longer decays.
*/ */
so->curRate = r<<10; 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 * Do this here because the patches assume a 44100 sampling rate
//called half the time. Ergo, double the rate to keep stuff * We've halved our sampling rate, ergo the ADSR code will be
//sounding right. * called half the time. Ergo, double the rate to keep stuff
* sounding right.
*/
so->curRate = so->curRate << 1; so->curRate = so->curRate << 1;
@ -297,17 +283,20 @@ inline signed short int synthVoice()
cpShifted = so->cp >> 10; cpShifted = so->cp >> 10;
if( (cpShifted >= (wf->wavSize>>1)) && (so->state != STATE_RAMPDOWN)) if( (cpShifted > (wf->numSamples) && (so->state != STATE_RAMPDOWN)))
{
stopVoice(so); stopVoice(so);
}
s2 = getSample((cpShifted)+1); s2 = getSample((cpShifted)+1);
if((wf->mode & (LOOP_REVERSE|LOOP_PINGPONG)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop>>1))) /* LOOP_REVERSE|LOOP_PINGPONG = 24 */
if((wf->mode & (24)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop)))
{ {
if(wf->mode & LOOP_REVERSE) if(wf->mode & LOOP_REVERSE)
{ {
so->cp = (wf->endLoop)<<9; so->cp = (wf->endLoop)<<10;
cpShifted = so->cp >> 10; cpShifted = wf->endLoop;
s2=getSample((cpShifted)); s2=getSample((cpShifted));
} else } else
{ {
@ -316,13 +305,13 @@ inline signed short int synthVoice()
} }
} }
if((wf->mode & 28) && (so->cp>>10 >= wf->endLoop>>1)) if((wf->mode & 28) && (cpShifted >= wf->endLoop))
{ {
so->loopState = STATE_LOOPING; so->loopState = STATE_LOOPING;
if((wf->mode & (24)) == 0) if((wf->mode & (24)) == 0)
{ {
so->cp = (wf->startLoop)<<9; so->cp = (wf->startLoop)<<10;
cpShifted = so->cp >> 10; cpShifted = wf->startLoop;
s2=getSample((cpShifted)); s2=getSample((cpShifted));
} else } else
{ {
@ -331,18 +320,18 @@ inline signed short int synthVoice()
} }
} }
//Better, working, linear interpolation /* Better, working, linear interpolation */
s1=getSample((cpShifted)); s1=getSample((cpShifted));
s = s1 + ((signed)((s2 - s1) * (so->cp & 1023))>>10); s = s1 + ((signed)((s2 - s1) * (so->cp & 1023))>>10);
//ADSR COMMENT WOULD GO FROM HERE......... /* ADSR COMMENT WOULD GO FROM HERE.........*/
if(so->curRate == 0) if(so->curRate == 0)
stopVoice(so); stopVoice(so);
if(so->ch != 9) //Stupid ADSR code... and don't do ADSR for drums if(so->ch != 9) /* Stupid ADSR code... and don't do ADSR for drums */
{ {
if(so->curOffset < so->targetOffset) if(so->curOffset < so->targetOffset)
{ {
@ -370,12 +359,12 @@ inline signed short int synthVoice()
} }
if(so->curOffset < 0) if(so->curOffset < 0)
so->isUsed=0; //This is OK so->isUsed=0; /* This is OK because offset faded it out already */
s = (s * (so->curOffset >> 22) >> 6); s = (s * (so->curOffset >> 22) >> 8);
// ............. TO HERE /* ............. TO HERE */
if(so->state == STATE_RAMPDOWN) if(so->state == STATE_RAMPDOWN)
@ -383,17 +372,15 @@ inline signed short int synthVoice()
so->decay--; so->decay--;
if(so->decay == 0) if(so->decay == 0)
so->isUsed=0; so->isUsed=0;
s = (s * so->decay) >> 8;
} }
s = s * so->decay; s = s >> 10;
return s*((signed short int)so->vol*(signed short int)chVol[so->ch])>>14; return s*((signed short int)so->vol*(signed short int)chVol[so->ch])>>14;
} }
inline void synthSample(int * mixL, int * mixR) inline void synthSample(int * mixL, int * mixR)
{ {
// signed int leftMix=0, rightMix=0,
*mixL = 0; *mixL = 0;
*mixR = 0; *mixR = 0;
for(currentVoice=0; currentVoice<MAX_VOICES; currentVoice++) for(currentVoice=0; currentVoice<MAX_VOICES; currentVoice++)
@ -406,7 +393,8 @@ inline void synthSample(int * mixL, int * mixR)
} }
} }
//TODO: Automatic Gain Control, anyone? /* TODO: Automatic Gain Control, anyone? */
//Or, should this be implemented on the DSP's output volume instead? /* Or, should this be implemented on the DSP's output volume instead? */
return; //No more ghetto lowpass filter.. linear intrpolation works well.
return; /* No more ghetto lowpass filter.. linear intrpolation works well. */
} }

85
apps/plugins/midi2wav.c
View file

@ -20,21 +20,22 @@
#define MAX_VOICES 100 #define MAX_VOICES 100
//Only define LOCAL_DSP on Simulator or else we're asking for trouble /* Only define LOCAL_DSP on Simulator or else we're asking for trouble */
#if defined(SIMULATOR) #if defined(SIMULATOR)
//Enable this to write to the soundcard via a /dsv/dsp symlink in / /*Enable this to write to the soundcard via a /dsv/dsp symlink in */
// #define LOCAL_DSP // #define LOCAL_DSP
#endif #endif
#if defined(LOCAL_DSP) #if defined(LOCAL_DSP)
// This is for writing to the DSP directly from the Simulator /* This is for writing to the DSP directly from the Simulator */
#include <stdio.h> #include <stdio.h>
#include <stdlib.h> #include <stdlib.h>
#include <linux/soundcard.h> #include <linux/soundcard.h>
#include <sys/ioctl.h> #include <sys/ioctl.h>
#endif #endif
#include "../../firmware/export/system.h"
#include "../../plugin.h" #include "../../plugin.h"
@ -53,9 +54,9 @@ long bpm;
int fd=-1; //File descriptor where the output is written int fd=-1; /* File descriptor where the output is written */
extern long tempo; //The sequencer keeps track of this extern long tempo; /* The sequencer keeps track of this */
struct plugin_api * rb; struct plugin_api * rb;
@ -77,6 +78,7 @@ enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
rb->splash(HZ*2, true, " Play .MID file "); rb->splash(HZ*2, true, " Play .MID file ");
return PLUGIN_OK; return PLUGIN_OK;
} }
rb->splash(HZ, true, parameter); rb->splash(HZ, true, parameter);
if(midimain(parameter) == -1) if(midimain(parameter) == -1)
{ {
@ -86,7 +88,7 @@ enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
return PLUGIN_OK; return PLUGIN_OK;
} }
signed char outputBuffer[3000] IDATA_ATTR; //signed char.. gonna run out of iram ... ! signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
int currentSample IDATA_ATTR; int currentSample IDATA_ATTR;
@ -110,27 +112,29 @@ int midimain(void * filename)
return -1; return -1;
} }
//This lets you hear the music through the sound card if you are on Simulator /*
//Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever * This lets you hear the music through the sound card if you are on Simulator
//your sound device is. * Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
* your sound device is.
*/
#if defined(LOCAL_DSP) #if defined(LOCAL_DSP)
fd=rb->open("/dsp.raw", O_WRONLY); fd=rb->open("/dsp.raw", O_WRONLY);
int arg, status; int arg, status;
int bit, samp, ch; int bit, samp, ch;
arg = 16; // sample size arg = 16; /* sample size */
status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg); status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
status = ioctl(fd, SOUND_PCM_READ_BITS, &arg); status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
bit=arg; bit=arg;
arg = 2; //Number of channels, 1=mono arg = 2; /* Number of channels, 1=mono */
status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg); status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg); status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
ch=arg; ch=arg;
arg = SAMPLE_RATE; //Yeah. sampling rate arg = SAMPLE_RATE; /* Yeah. sampling rate */
status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg); status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
status = ioctl(fd, SOUND_PCM_READ_RATE, &arg); status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
samp=arg; samp=arg;
@ -145,25 +149,25 @@ int midimain(void * filename)
#endif #endif
rb->splash(HZ/5, true, " Starting Playback "); rb->splash(HZ/5, true, " I hope this works... ");
/*
// tick() will do one MIDI clock tick. Then, there's a loop here that * tick() will do one MIDI clock tick. Then, there's a loop here that
// will generate the right number of samples per MIDI tick. The whole * will generate the right number of samples per MIDI tick. The whole
// MIDI playback is timed in terms of this value.. there are no forced * MIDI playback is timed in terms of this value.. there are no forced
// delays or anything. It just produces enough samples for each tick, and * delays or anything. It just produces enough samples for each tick, and
// the playback of these samples is what makes the timings right. * the playback of these samples is what makes the timings right.
// *
// This seems to work quite well. * This seems to work quite well.
*/
printf("\nOkay, starting sequencing"); printf("\nOkay, starting sequencing");
currentSample=0; //Sample counting variable currentSample=0; /* Sample counting variable */
outputBufferPosition = 0; outputBufferPosition = 0;
@ -172,28 +176,26 @@ int midimain(void * filename)
//Tick() will return 0 if there are no more events left to play /* Tick() will return 0 if there are no more events left to play */
while(tick(mf)) while(tick(mf))
{ {
/*
//Some annoying math to compute the number of samples * Tempo recalculation moved to sequencer.c to be done on a tempo event only
//to syntehsize per each MIDI tick. *
*/
//Yes we need to do this math each time because the tempo
//could have changed.
// On second thought, this can be moved to the event that
//recalculates the tempo, to save a little bit of CPU time.
for(currentSample=0; currentSample<numberOfSamples; currentSample++) for(currentSample=0; currentSample<numberOfSamples; currentSample++)
{ {
synthSample(&outputSampleOne, &outputSampleTwo); synthSample(&outputSampleOne, &outputSampleTwo);
//16-bit audio because, well, it's better /*
// But really because ALSA's OSS emulation sounds extremely * 16-bit audio because, well, it's better
//noisy and distorted when in 8-bit mode. I still do not know * But really because ALSA's OSS emulation sounds extremely
//why this happens. * noisy and distorted when in 8-bit mode. I still do not know
* why this happens.
*/
outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; // Low byte first outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; // Low byte first
outputBufferPosition++; outputBufferPosition++;
outputBuffer[outputBufferPosition]=outputSampleOne>>8; //High byte second outputBuffer[outputBufferPosition]=outputSampleOne>>8; //High byte second
@ -205,9 +207,11 @@ int midimain(void * filename)
outputBufferPosition++; outputBufferPosition++;
//As soon as we produce 2000 bytes of sound, /*
//write it to the sound card. Why 2000? I have * As soon as we produce 2000 bytes of sound,
//no idea. It's 1 AM and I am dead tired. * write it to the sound card. Why 2000? I have
* no idea. It's 1 AM and I am dead tired.
*/
if(outputBufferPosition>=2000) if(outputBufferPosition>=2000)
{ {
rb->write(fd, outputBuffer, 2000); rb->write(fd, outputBuffer, 2000);
@ -216,7 +220,6 @@ int midimain(void * filename)
} }
} }
// unloadFile(mf);
printf("\n"); printf("\n");
#if !defined(LOCAL_DSP) #if !defined(LOCAL_DSP)