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 int fd=-1; /* File descriptor where the output is written */
  58
  59 extern long tempo; /* The sequencer keeps track of this */
  60
  61 const struct plugin_api * rb;
  62
  63
  64 enum plugin_status plugin_start(const struct plugin_api* api, const void* parameter)
  65 {
  66     (void)parameter;
  67     rb = api;
  68
  69     if(parameter == NULL)
  70     {
  71         rb->splash(HZ*2, "Play .MID file");
  72         return PLUGIN_OK;
  73     }
  74
  75     rb->splash(HZ, parameter);
  76     if(midimain(parameter) == -1)
  77     {
  78         return PLUGIN_ERROR;
  79     }
  80     rb->splash(HZ*3, "FINISHED PLAYING");
  81     /* Return PLUGIN_USB_CONNECTED to force a file-tree refresh */
  82     return PLUGIN_USB_CONNECTED;
  83 }
  84
  85 signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
  86
  87
  88 int currentSample IDATA_ATTR;
  89 int outputBufferPosition IDATA_ATTR;
  90 int outputSampleOne IDATA_ATTR;
  91 int outputSampleTwo IDATA_ATTR;
  92
  93
  94 int midimain(void * filename)
  95 {
  96
  97     printf("\nHello.\n");
  98
  99     rb->splash(HZ/5, "LOADING MIDI");
 100
 101     struct MIDIfile * mf = loadFile(filename);
 102
 103     rb->splash(HZ/5, "LOADING PATCHES");
 104     if (initSynth(mf, ROCKBOX_DIR "/patchset/patchset.cfg",
 105         ROCKBOX_DIR "/patchset/drums.cfg") == -1)
 106     {
 107         return -1;
 108     }
 109
 110 /*
 111  *  This lets you hear the music through the sound card if you are on Simulator
 112  *  Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
 113  *  your sound device is.
 114  */
 115
 116 #if defined(LOCAL_DSP)
 117     fd=rb->open("/dsp.raw", O_WRONLY);
 118     int arg, status;
 119     int bit, samp, ch;
 120
 121     arg = 16;          /* sample size */
 122     status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
 123     status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
 124     bit=arg;
 125
 126
 127     arg = 2;        /* Number of channels, 1=mono */
 128     status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
 129     status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
 130     ch=arg;
 131
 132     arg = SAMPLE_RATE; /* Yeah. sampling rate */
 133     status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
 134     status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
 135     samp=arg;
 136 #else
 137
 138 /* xxx2wav stuff, removed for now, will move to the real way of outputting sound soon */
 139 /*
 140     file_info_struct file_info;
 141     file_info.samplerate = SAMPLE_RATE;
 142     file_info.infile = fd;
 143     file_info.channels = 2;
 144     file_info.bitspersample = 16;
 145     local_init("/miditest.tmp", "/miditest.wav", &file_info, rb);
 146     fd = file_info.outfile;
 147 */
 148 #endif
 149
 150
 151     rb->splash(HZ/5, "I hope this works...");
 152
 153
 154
 155
 156     /*
 157      * tick() will do one MIDI clock tick. Then, there's a loop here that
 158      * will generate the right number of samples per MIDI tick. The whole
 159      * MIDI playback is timed in terms of this value.. there are no forced
 160      * delays or anything. It just produces enough samples for each tick, and
 161      * the playback of these samples is what makes the timings right.
 162      *
 163      * This seems to work quite well.
 164      */
 165
 166     printf("\nOkay, starting sequencing");
 167
 168
 169     currentSample=0;            /* Sample counting variable */
 170     outputBufferPosition = 0;
 171
 172
 173     bpm=mf->div*1000000/tempo;
 174     numberOfSamples=SAMPLE_RATE/bpm;
 175
 176
 177
 178     /* Tick() will return 0 if there are no more events left to play */
 179     while(tick(mf))
 180     {
 181         /*
 182          * Tempo recalculation moved to sequencer.c to be done on a tempo event only
 183          *
 184          */
 185         for(currentSample=0; currentSample<numberOfSamples; currentSample++)
 186         {
 187
 188             synthSample(&outputSampleOne, &outputSampleTwo);
 189
 190
 191             /*
 192              * 16-bit audio because, well, it's better
 193              * But really because ALSA's OSS emulation sounds extremely
 194              * noisy and distorted when in 8-bit mode. I still do not know
 195              * why this happens.
 196              */
 197
 198             outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; /* Low byte first */
 199             outputBufferPosition++;
 200             outputBuffer[outputBufferPosition]=outputSampleOne>>8;   /*High byte second */
 201             outputBufferPosition++;
 202
 203             outputBuffer[outputBufferPosition]=outputSampleTwo&0XFF; /* Low byte first */
 204             outputBufferPosition++;
 205             outputBuffer[outputBufferPosition]=outputSampleTwo>>8;   /*High byte second */
 206             outputBufferPosition++;
 207
 208
 209             /*
 210              * As soon as we produce 2000 bytes of sound,
 211              * write it to the sound card. Why 2000? I have
 212              * no idea. It's 1 AM and I am dead tired.
 213              */
 214             if(outputBufferPosition>=2000)
 215             {
 216                 rb->write(fd, outputBuffer, 2000);
 217                 outputBufferPosition=0;
 218             }
 219         }
 220     }
 221
 222     printf("\n");
 223
 224 #if !defined(LOCAL_DSP)
 225
 226 /* again, xxx2wav stuff, removed for now */
 227
 228     /* close_wav(&file_info); */
 229 #else
 230     rb->close(fd);
 231 #endif
 232     return 0;
 233 }