forked from len0rd/rockbox
1f5fb99819
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6323 a1c6a512-1295-4272-9138-f99709370657
412 lines
8.8 KiB
C
412 lines
8.8 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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*
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* Copyright (C) 2005 Stepan Moskovchenko
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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extern struct plugin_api * rb;
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struct Event * getEvent(struct Track * tr, int evNum)
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{
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return tr->dataBlock + (evNum*sizeof(struct Event));
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}
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void readTextBlock(int file, char * buf)
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{
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char c = 0;
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do
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{
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c = readChar(file);
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} while(c == '\n' || c == ' ' || c=='\t');
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rb->lseek(file, -1, SEEK_CUR);
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int cp = 0;
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do
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{
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c = readChar(file);
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buf[cp] = c;
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cp++;
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} while (c != '\n' && c != ' ' && c != '\t' && !eof(file));
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buf[cp-1]=0;
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rb->lseek(file, -1, SEEK_CUR);
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}
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//Filename is the name of the config file
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//The MIDI file should have been loaded at this point
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int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
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{
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char patchUsed[128];
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char drumUsed[128];
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int a=0;
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for(a=0; a<MAX_VOICES; a++)
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{
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voices[a].cp=0;
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voices[a].vol=0;
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voices[a].ch=0;
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voices[a].isUsed=0;
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voices[a].note=0;
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}
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for(a=0; a<16; a++)
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{
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chVol[a]=100; //Default, not quite full blast..
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chPanLeft[a]=64; //Center
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chPanRight[a]=64; //Center
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chPat[a]=0; //Ac Gr Piano
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chPW[a]=64; // .. not .. bent ?
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}
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for(a=0; a<128; a++)
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{
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patchSet[a]=NULL;
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drumSet[a]=NULL;
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patchUsed[a]=0;
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drumUsed[a]=0;
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}
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//Always load the piano.
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//Some files will assume its loaded without specifically
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//issuing a Patch command... then we wonder why we can't hear anything
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patchUsed[0]=1;
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//Scan the file to see what needs to be loaded
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for(a=0; a<mf->numTracks; a++)
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{
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int ts=0;
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if(mf->tracks[a] == NULL)
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{
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printf("\nNULL TRACK !!!");
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rb->splash(HZ*2, true, "Null Track in loader.");
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return -1;
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}
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for(ts=0; ts<mf->tracks[a]->numEvents; ts++)
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{
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if((getEvent(mf->tracks[a], ts)->status) == (MIDI_NOTE_ON+9))
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drumUsed[getEvent(mf->tracks[a], ts)->d1]=1;
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if( (getEvent(mf->tracks[a], ts)->status & 0xF0) == MIDI_PRGM)
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{
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if(patchUsed[getEvent(mf->tracks[a], ts)->d1]==0)
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printf("\nI need to load patch %d.", getEvent(mf->tracks[a], ts)->d1);
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patchUsed[getEvent(mf->tracks[a], ts)->d1]=1;
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}
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}
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}
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int file = rb->open(filename, O_RDONLY);
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if(file == -1)
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{
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rb->splash(HZ*2, true, "Bad patch config.\nDid you install the patchset?");
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return -1;
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}
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char name[40];
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char fn[40];
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//Scan our config file and load the right patches as needed
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int c = 0;
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rb->snprintf(name, 40, "");
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for(a=0; a<128; a++)
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{
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while(readChar(file)!=' ' && !eof(file));
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readTextBlock(file, name);
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rb->snprintf(fn, 40, "/.rockbox/patchset/%s.pat", name);
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printf("\nLOADING: <%s> ", fn);
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if(patchUsed[a]==1)
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patchSet[a]=gusload(fn);
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// if(patchSet[a] == NULL)
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// return -1;
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while((c != '\n'))
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c = readChar(file);
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}
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rb->close(file);
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file = rb->open(drumConfig, O_RDONLY);
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if(file == -1)
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{
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rb->splash(HZ*2, true, "Bad drum config.\nDid you install the patchset?");
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return -1;
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}
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//Scan our config file and load the drum data
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int idx=0;
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char number[30];
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while(!eof(file))
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{
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readTextBlock(file, number);
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readTextBlock(file, name);
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rb->snprintf(fn, 40, "/.rockbox/patchset/%s.pat", name);
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idx = rb->atoi(number);
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if(idx == 0)
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break;
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if(drumUsed[idx]==1)
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drumSet[idx]=gusload(fn);
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// if(drumSet[idx] == NULL)
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// return -1;
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while((c != '\n') && (c != 255) && (!eof(file)))
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c = readChar(file);
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}
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rb->close(file);
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return 0;
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}
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int currentVoice IDATA_ATTR;
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struct SynthObject * so IDATA_ATTR;
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struct GWaveform * wf IDATA_ATTR;
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int s IDATA_ATTR;
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short s1 IDATA_ATTR;
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short s2 IDATA_ATTR;
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short sample IDATA_ATTR; //For synthSample
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unsigned int cpShifted IDATA_ATTR;
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unsigned char b1 IDATA_ATTR;
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unsigned char b2 IDATA_ATTR;
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inline int getSample(int s)
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{
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//16 bit samples
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if(wf->mode&1)
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{
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if(s<<1 >= wf->wavSize)
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{
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printf("\n!!!!! %d \t %d", s<<1, wf->wavSize);
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return 0;
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}
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// signed short a = ((short *)wf->data)[s];
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//Sign conversion moved into guspat.c
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b1=wf->data[s<<1]+((wf->mode & 2) << 6);
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b2=wf->data[(s<<1)|1]+((wf->mode & 2) << 6);
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return (b1 | (b2<<8)) ;
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//Get rid of this sometime
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}
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else
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{ //8-bit samples
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//Do we even have anything 8-bit in our set?
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int b1=wf->data[s]+((wf->mode & 2) << 6);
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return b1<<8;
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}
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}
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inline void setPoint(struct SynthObject * so, int pt)
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{
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if(so->ch==9) //Drums, no ADSR
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{
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so->curOffset = 1<<27;
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so->curRate = 1;
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return;
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}
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if(so->wf==NULL)
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{
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printf("\nCrap... null waveform...");
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exit(1);
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}
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if(so->wf->envRate==NULL)
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{
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printf("\nWaveform has no envelope set");
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exit(1);
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}
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so->curPoint = pt;
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int r=0;
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int rate = so->wf->envRate[pt];
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r=3-((rate>>6) & 0x3); // Some blatant Timidity code for rate conversion...
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r*=3;
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r = (rate & 0x3f) << r;
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/*
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Okay. This is the rate shift. Timidity defaults to 9, and sets
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it to 10 if you use the fast decay option. Slow decay sounds better
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on some files, except on some other files... you get chords that aren't
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done decaying yet.. and they dont harmonize with the next chord and it
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sounds like utter crap. Yes, even Timitidy does that. So I'm going to
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default this to 10, and maybe later have an option to set it to 9
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for longer decays.
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*/
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so->curRate = r<<10;
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//Do this here because the patches assume a 44100 sampling rate
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//We've halved our sampling rate, ergo the ADSR code will be
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//called half the time. Ergo, double the rate to keep stuff
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//sounding right.
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so->curRate = so->curRate << 1;
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so->targetOffset = so->wf->envOffset[pt]<<(20);
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if(pt==0)
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so->curOffset = 0;
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}
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inline void stopVoice(struct SynthObject * so)
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{
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if(so->state == STATE_RAMPDOWN)
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return;
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so->state = STATE_RAMPDOWN;
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so->decay = 255;
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}
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inline signed short int synthVoice()
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{
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so = &voices[currentVoice];
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wf = so->wf;
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if(so->state != STATE_RAMPDOWN)
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{
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so->cp += so->delta;
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}
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cpShifted = so->cp >> 10;
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if( (cpShifted >= (wf->wavSize>>1)) && (so->state != STATE_RAMPDOWN))
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stopVoice(so);
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s2 = getSample((cpShifted)+1);
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if((wf->mode & (LOOP_REVERSE|LOOP_PINGPONG)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop>>1)))
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{
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if(wf->mode & LOOP_REVERSE)
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{
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so->cp = (wf->endLoop)<<9;
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cpShifted = so->cp >> 10;
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s2=getSample((cpShifted));
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} else
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{
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so->delta = -so->delta;
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so->loopDir = LOOPDIR_FORWARD;
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}
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}
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if((wf->mode & 28) && (so->cp>>10 >= wf->endLoop>>1))
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{
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so->loopState = STATE_LOOPING;
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if((wf->mode & (24)) == 0)
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{
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so->cp = (wf->startLoop)<<9;
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cpShifted = so->cp >> 10;
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s2=getSample((cpShifted));
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} else
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{
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so->delta = -so->delta;
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so->loopDir = LOOPDIR_REVERSE;
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}
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}
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//Better, working, linear interpolation
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s1=getSample((cpShifted));
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s = s1 + ((signed)((s2 - s1) * (so->cp & 1023))>>10);
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//ADSR COMMENT WOULD GO FROM HERE.........
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if(so->curRate == 0)
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stopVoice(so);
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if(so->ch != 9) //Stupid ADSR code... and don't do ADSR for drums
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{
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if(so->curOffset < so->targetOffset)
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{
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so->curOffset += (so->curRate);
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if(so -> curOffset > so->targetOffset && so->curPoint != 2)
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{
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if(so->curPoint != 5)
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setPoint(so, so->curPoint+1);
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else
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stopVoice(so);
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}
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} else
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{
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so->curOffset -= (so->curRate);
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if(so -> curOffset < so->targetOffset && so->curPoint != 2)
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{
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if(so->curPoint != 5)
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setPoint(so, so->curPoint+1);
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else
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stopVoice(so);
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}
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}
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}
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if(so->curOffset < 0)
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so->isUsed=0; //This is OK
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s = (s * (so->curOffset >> 22) >> 6);
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// ............. TO HERE
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if(so->state == STATE_RAMPDOWN)
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{
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so->decay--;
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if(so->decay == 0)
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so->isUsed=0;
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}
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s = s * so->decay; s = s >> 10;
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return s*((signed short int)so->vol*(signed short int)chVol[so->ch])>>14;
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}
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inline void synthSample(int * mixL, int * mixR)
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{
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// signed int leftMix=0, rightMix=0,
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*mixL = 0;
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*mixR = 0;
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for(currentVoice=0; currentVoice<MAX_VOICES; currentVoice++)
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{
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if(voices[currentVoice].isUsed==1)
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{
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sample = synthVoice(currentVoice);
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*mixL += (sample*chPanLeft[voices[currentVoice].ch])>>7;
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*mixR += (sample*chPanRight[voices[currentVoice].ch])>>7;
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}
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}
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//TODO: Automatic Gain Control, anyone?
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//Or, should this be implemented on the DSP's output volume instead?
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return; //No more ghetto lowpass filter.. linear intrpolation works well.
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}
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