apps/plugins/midi2wav.c

   1 /***************************************************************************
   2  *             __________               __   ___.
   3  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
   4  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
   5  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
   6  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
   7  *                     \/            \/     \/    \/            \/
   8  *
   9  * Copyright (C) 2005 Stepan Moskovchenko
  10  *
  11  * All files in this archive are subject to the GNU General Public License.
  12  * See the file COPYING in the source tree root for full license agreement.
  13  *
  14  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  15  * KIND, either express or implied.
  16  *
  17  ****************************************************************************/
  18
  19 #define SAMPLE_RATE 22050
  20 #define MAX_VOICES 100
  21
  22
  23 /* Only define LOCAL_DSP on Simulator or else we're asking for trouble  */
  24 #if defined(SIMULATOR)
  25     /*Enable this to write to the soundcard via a /dsv/dsp symlink in */
  26     /*#define LOCAL_DSP */
  27 #endif
  28
  29
  30 #if defined(LOCAL_DSP)
  31 /*  This is for writing to the DSP directly from the Simulator */
  32 #include <stdio.h>
  33 #include <stdlib.h>
  34 #include <linux/soundcard.h>
  35 #include <sys/ioctl.h>
  36 #endif
  37
  38 #include "../../firmware/export/system.h"
  39
  40 #include "../../plugin.h"
  41
  42 /*#include "../codecs/lib/xxx2wav.h" */
  43
  44 PLUGIN_HEADER
  45
  46 int numberOfSamples IDATA_ATTR;
  47 long bpm;
  48
  49 #include "midi/midiutil.c"
  50 #include "midi/guspat.h"
  51 #include "midi/guspat.c"
  52 #include "midi/sequencer.c"
  53 #include "midi/midifile.c"
  54 #include "midi/synth.c"
  55
  56
  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
  64 struct plugin_api * rb;
  65
  66
  67
  68
  69
  70 enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
  71 {
  72     (void)parameter;
  73     rb = api;
  74
  75     if(parameter == NULL)
  76     {
  77         rb->splash(HZ*2, "Play .MID file");
  78         return PLUGIN_OK;
  79     }
  80
  81     rb->splash(HZ, parameter);
  82     if(midimain(parameter) == -1)
  83     {
  84         return PLUGIN_ERROR;
  85     }
  86     rb->splash(HZ*3, "FINISHED PLAYING");
  87     /* Return PLUGIN_USB_CONNECTED to force a file-tree refresh */
  88     return PLUGIN_USB_CONNECTED;
  89 }
  90
  91 signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
  92
  93
  94 int currentSample IDATA_ATTR;
  95 int outputBufferPosition IDATA_ATTR;
  96 int outputSampleOne IDATA_ATTR;
  97 int outputSampleTwo IDATA_ATTR;
  98
  99
 100 int midimain(void * filename)
 101 {
 102
 103     printf("\nHello.\n");
 104
 105     rb->splash(HZ/5, "LOADING MIDI");
 106
 107     struct MIDIfile * mf = loadFile(filename);
 108
 109     rb->splash(HZ/5, "LOADING PATCHES");
 110     if (initSynth(mf, ROCKBOX_DIR "/patchset/patchset.cfg",
 111         ROCKBOX_DIR "/patchset/drums.cfg") == -1)
 112     {
 113         return -1;
 114     }
 115
 116 /*
 117  *  This lets you hear the music through the sound card if you are on Simulator
 118  *  Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
 119  *  your sound device is.
 120  */
 121
 122 #if defined(LOCAL_DSP)
 123     fd=rb->open("/dsp.raw", O_WRONLY);
 124     int arg, status;
 125     int bit, samp, ch;
 126
 127     arg = 16;          /* sample size */
 128     status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
 129     status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
 130     bit=arg;
 131
 132
 133     arg = 2;        /* Number of channels, 1=mono */
 134     status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
 135     status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
 136     ch=arg;
 137
 138     arg = SAMPLE_RATE; /* Yeah. sampling rate */
 139     status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
 140     status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
 141     samp=arg;
 142 #else
 143
 144 /* xxx2wav stuff, removed for now, will move to the real way of outputting sound soon */
 145 /*
 146     file_info_struct file_info;
 147     file_info.samplerate = SAMPLE_RATE;
 148     file_info.infile = fd;
 149     file_info.channels = 2;
 150     file_info.bitspersample = 16;
 151     local_init("/miditest.tmp", "/miditest.wav", &file_info, rb);
 152     fd = file_info.outfile;
 153 */
 154 #endif
 155
 156
 157     rb->splash(HZ/5, "I hope this works...");
 158
 159
 160
 161
 162     /*
 163      * tick() will do one MIDI clock tick. Then, there's a loop here that
 164      * will generate the right number of samples per MIDI tick. The whole
 165      * MIDI playback is timed in terms of this value.. there are no forced
 166      * delays or anything. It just produces enough samples for each tick, and
 167      * the playback of these samples is what makes the timings right.
 168      *
 169      * This seems to work quite well.
 170      */
 171
 172     printf("\nOkay, starting sequencing");
 173
 174
 175     currentSample=0;            /* Sample counting variable */
 176     outputBufferPosition = 0;
 177
 178
 179     bpm=mf->div*1000000/tempo;
 180     numberOfSamples=SAMPLE_RATE/bpm;
 181
 182
 183
 184     /* Tick() will return 0 if there are no more events left to play */
 185     while(tick(mf))
 186     {
 187         /*
 188          * Tempo recalculation moved to sequencer.c to be done on a tempo event only
 189          *
 190          */
 191         for(currentSample=0; currentSample<numberOfSamples; currentSample++)
 192         {
 193
 194             synthSample(&outputSampleOne, &outputSampleTwo);
 195
 196
 197             /*
 198              * 16-bit audio because, well, it's better
 199              * But really because ALSA's OSS emulation sounds extremely
 200              * noisy and distorted when in 8-bit mode. I still do not know
 201              * why this happens.
 202              */
 203
 204             outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; /* Low byte first */
 205             outputBufferPosition++;
 206             outputBuffer[outputBufferPosition]=outputSampleOne>>8;   /*High byte second */
 207             outputBufferPosition++;
 208
 209             outputBuffer[outputBufferPosition]=outputSampleTwo&0XFF; /* Low byte first */
 210             outputBufferPosition++;
 211             outputBuffer[outputBufferPosition]=outputSampleTwo>>8;   /*High byte second */
 212             outputBufferPosition++;
 213
 214
 215             /*
 216              * As soon as we produce 2000 bytes of sound,
 217              * write it to the sound card. Why 2000? I have
 218              * no idea. It's 1 AM and I am dead tired.
 219              */
 220             if(outputBufferPosition>=2000)
 221             {
 222                 rb->write(fd, outputBuffer, 2000);
 223                 outputBufferPosition=0;
 224             }
 225         }
 226     }
 227
 228     printf("\n");
 229
 230 #if !defined(LOCAL_DSP)
 231
 232 /* again, xxx2wav stuff, removed for now */
 233
 234     /* close_wav(&file_info); */
 235 #else
 236     rb->close(fd);
 237 #endif
 238     return 0;
 239 }