apps/plugins/midi2wav.c

   1 /***************************************************************************
   2  *             __________               __   ___.
   3  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
   4  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
   5  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
   6  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
   7  *                     \/            \/     \/    \/            \/
   8  *
   9  * Copyright (C) 2005 Stepan Moskovchenko
  10  *
  11  * This program is free software; you can redistribute it and/or
  12  * modify it under the terms of the GNU General Public License
  13  * as published by the Free Software Foundation; either version 2
  14  * of the License, or (at your option) any later version.
  15  *
  16  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  17  * KIND, either express or implied.
  18  *
  19  ****************************************************************************/
  20
  21 #define SAMPLE_RATE 22050
  22 #define MAX_VOICES 100
  23
  24
  25 /* Only define LOCAL_DSP on Simulator or else we're asking for trouble  */
  26 #if defined(SIMULATOR)
  27     /*Enable this to write to the soundcard via a /dsv/dsp symlink in */
  28     /*#define LOCAL_DSP */
  29 #endif
  30
  31
  32 #if defined(LOCAL_DSP)
  33 /*  This is for writing to the DSP directly from the Simulator */
  34 #include <stdio.h>
  35 #include <stdlib.h>
  36 #include <linux/soundcard.h>
  37 #include <sys/ioctl.h>
  38 #endif
  39
  40 #include "../../firmware/export/system.h"
  41
  42 #include "../../plugin.h"
  43
  44 /*#include "../codecs/lib/xxx2wav.h" */
  45
  46 PLUGIN_HEADER
  47
  48 int numberOfSamples IDATA_ATTR;
  49 long bpm;
  50
  51 #include "midi/midiutil.c"
  52 #include "midi/guspat.h"
  53 #include "midi/guspat.c"
  54 #include "midi/sequencer.c"
  55 #include "midi/midifile.c"
  56 #include "midi/synth.c"
  57
  58
  59 int fd=-1; /* File descriptor where the output is written */
  60
  61 extern long tempo; /* The sequencer keeps track of this */
  62
  63 const struct plugin_api * rb;
  64
  65
  66 enum plugin_status plugin_start(const struct plugin_api* api, const void* parameter)
  67 {
  68     (void)parameter;
  69     rb = api;
  70
  71     if(parameter == NULL)
  72     {
  73         rb->splash(HZ*2, "Play .MID file");
  74         return PLUGIN_OK;
  75     }
  76
  77     rb->splash(HZ, parameter);
  78     if(midimain(parameter) == -1)
  79     {
  80         return PLUGIN_ERROR;
  81     }
  82     rb->splash(HZ*3, "FINISHED PLAYING");
  83     /* Return PLUGIN_USB_CONNECTED to force a file-tree refresh */
  84     return PLUGIN_USB_CONNECTED;
  85 }
  86
  87 signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
  88
  89
  90 int currentSample IDATA_ATTR;
  91 int outputBufferPosition IDATA_ATTR;
  92 int outputSampleOne IDATA_ATTR;
  93 int outputSampleTwo IDATA_ATTR;
  94
  95
  96 int midimain(void * filename)
  97 {
  98
  99     printf("\nHello.\n");
 100
 101     rb->splash(HZ/5, "LOADING MIDI");
 102
 103     struct MIDIfile * mf = loadFile(filename);
 104
 105     rb->splash(HZ/5, "LOADING PATCHES");
 106     if (initSynth(mf, ROCKBOX_DIR "/patchset/patchset.cfg",
 107         ROCKBOX_DIR "/patchset/drums.cfg") == -1)
 108     {
 109         return -1;
 110     }
 111
 112 /*
 113  *  This lets you hear the music through the sound card if you are on Simulator
 114  *  Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
 115  *  your sound device is.
 116  */
 117
 118 #if defined(LOCAL_DSP)
 119     fd=rb->open("/dsp.raw", O_WRONLY);
 120     int arg, status;
 121     int bit, samp, ch;
 122
 123     arg = 16;          /* sample size */
 124     status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
 125     status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
 126     bit=arg;
 127
 128
 129     arg = 2;        /* Number of channels, 1=mono */
 130     status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
 131     status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
 132     ch=arg;
 133
 134     arg = SAMPLE_RATE; /* Yeah. sampling rate */
 135     status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
 136     status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
 137     samp=arg;
 138 #else
 139
 140 /* xxx2wav stuff, removed for now, will move to the real way of outputting sound soon */
 141 /*
 142     file_info_struct file_info;
 143     file_info.samplerate = SAMPLE_RATE;
 144     file_info.infile = fd;
 145     file_info.channels = 2;
 146     file_info.bitspersample = 16;
 147     local_init("/miditest.tmp", "/miditest.wav", &file_info, rb);
 148     fd = file_info.outfile;
 149 */
 150 #endif
 151
 152
 153     rb->splash(HZ/5, "I hope this works...");
 154
 155
 156
 157
 158     /*
 159      * tick() will do one MIDI clock tick. Then, there's a loop here that
 160      * will generate the right number of samples per MIDI tick. The whole
 161      * MIDI playback is timed in terms of this value.. there are no forced
 162      * delays or anything. It just produces enough samples for each tick, and
 163      * the playback of these samples is what makes the timings right.
 164      *
 165      * This seems to work quite well.
 166      */
 167
 168     printf("\nOkay, starting sequencing");
 169
 170
 171     currentSample=0;            /* Sample counting variable */
 172     outputBufferPosition = 0;
 173
 174
 175     bpm=mf->div*1000000/tempo;
 176     numberOfSamples=SAMPLE_RATE/bpm;
 177
 178
 179
 180     /* Tick() will return 0 if there are no more events left to play */
 181     while(tick(mf))
 182     {
 183         /*
 184          * Tempo recalculation moved to sequencer.c to be done on a tempo event only
 185          *
 186          */
 187         for(currentSample=0; currentSample<numberOfSamples; currentSample++)
 188         {
 189
 190             synthSample(&outputSampleOne, &outputSampleTwo);
 191
 192
 193             /*
 194              * 16-bit audio because, well, it's better
 195              * But really because ALSA's OSS emulation sounds extremely
 196              * noisy and distorted when in 8-bit mode. I still do not know
 197              * why this happens.
 198              */
 199
 200             outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; /* Low byte first */
 201             outputBufferPosition++;
 202             outputBuffer[outputBufferPosition]=outputSampleOne>>8;   /*High byte second */
 203             outputBufferPosition++;
 204
 205             outputBuffer[outputBufferPosition]=outputSampleTwo&0XFF; /* Low byte first */
 206             outputBufferPosition++;
 207             outputBuffer[outputBufferPosition]=outputSampleTwo>>8;   /*High byte second */
 208             outputBufferPosition++;
 209
 210
 211             /*
 212              * As soon as we produce 2000 bytes of sound,
 213              * write it to the sound card. Why 2000? I have
 214              * no idea. It's 1 AM and I am dead tired.
 215              */
 216             if(outputBufferPosition>=2000)
 217             {
 218                 rb->write(fd, outputBuffer, 2000);
 219                 outputBufferPosition=0;
 220             }
 221         }
 222     }
 223
 224     printf("\n");
 225
 226 #if !defined(LOCAL_DSP)
 227
 228 /* again, xxx2wav stuff, removed for now */
 229
 230     /* close_wav(&file_info); */
 231 #else
 232     rb->close(fd);
 233 #endif
 234     return 0;
 235 }