2ed84ec2ca
In revision 1.4 Steven made currentVoice global, so the function declaration should be: synthVoice(void), wich fixes a warning. And the call should be just synthVoice() wich fixes a hidden error. Hope I saw this right... please correct me if this isn't the way you ment is Seven. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6708 a1c6a512-1295-4272-9138-f99709370657
400 lines
10 KiB
C
400 lines
10 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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/*
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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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*/
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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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unsigned 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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{
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patchSet[a]=gusload(fn);
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if(patchSet[a] == NULL) /* There was an error loading it */
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return -1;
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}
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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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{
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drumSet[idx]=gusload(fn);
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if(drumSet[idx] == NULL) /* Error loading patch */
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return -1;
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}
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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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/* Sign conversion moved to guspat.c */
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/* 8bit conversion NOT YET IMPLEMENTED in guspat.c */
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return ((short *) wf->data)[s];
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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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/*
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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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*/
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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(void)
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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->numSamples) && (so->state != STATE_RAMPDOWN)))
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{
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stopVoice(so);
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}
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s2 = getSample((cpShifted)+1);
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/* LOOP_REVERSE|LOOP_PINGPONG = 24 */
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if((wf->mode & (24)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop)))
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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)<<10;
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cpShifted = wf->endLoop;
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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) && (cpShifted >= wf->endLoop))
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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)<<10;
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cpShifted = wf->startLoop;
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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 because offset faded it out already */
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s = (s * (so->curOffset >> 22) >> 8);
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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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s = (s * so->decay) >> 8;
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}
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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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*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();
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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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