1 files changed, 168 insertions, 0 deletions

diff --git a/apps/plugins/midi2wav.c b/apps/plugins/midi2wav.c
new file mode 100644
index 0000000000..f8368535d1
--- /dev/null
+++ b/apps/plugins/midi2wav.c
@@ -0,0 +1,168 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ *
+ * Copyright (C) 2005 Stepan Moskovchenko
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#define SAMPLE_RATE 48000
+#define MAX_VOICES 100
+
+/*      This is for writing to the DSP directly from the Simulator
+#include <stdio.h>
+#include <stdlib.h>
+#include <linux/soundcard.h>
+#include <sys/ioctl.h>
+*/
+
+#include "../../plugin.h"
+#include "midi/midiutil.c"
+#include "midi/guspat.h"
+#include "midi/guspat.c"
+#include "midi/sequencer.c"
+#include "midi/midifile.c"
+#include "midi/synth.c"
+
+
+
+//#include "lib/xxx2wav.h"
+
+int fd=-1; //File descriptor, for opening /dev/dsp and writing to it
+
+extern long tempo; //The sequencer keeps track of this
+
+
+struct plugin_api * rb;
+
+
+
+enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
+{
+    TEST_PLUGIN_API(api);
+    (void)parameter;
+    rb = api;
+    rb->splash(HZ*2, true, "MIDI");
+    midimain();
+    rb->splash(HZ*2, true, "FINISHED PLAYING");
+    return PLUGIN_OK;
+}
+
+
+int midimain()
+{
+        rb->splash(HZ*2, true, "OPENED DSP");
+        fd=rb->open("/dsp.raw", O_WRONLY|O_CREAT);
+/*
+        int arg, status;
+        int bit, samp, ch;
+
+        arg = 16;          // sample size
+        status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
+        status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
+        bit=arg;
+
+
+        arg = 2;        //Number of channels, 1=mono
+        status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
+        status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
+        ch=arg;
+
+        arg = SAMPLE_RATE; //Yeah. sampling rate
+        status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
+        status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
+        samp=arg;
+*/
+
+        printf("\nHello.\n");
+//      initSound();    //Open the computer's sound card
+        int a=0;
+
+        rb->splash(HZ*2, true, "LOADING MIDI");
+
+        struct MIDIfile * mf = loadFile("/test.mid");
+
+        rb->splash(HZ*2, true, "LOADED MIDI");
+        long bpm, nsmp, l;
+
+        int bp=0;
+
+        rb->splash(HZ*2, true, "LOADING PATCHES");
+        initSynth(mf, "/iriver2.cfg", "/drums.cfg");    //Initialize the MIDI syntehsizer
+        rb->splash(HZ*2, true, "START PLAYING");
+
+        signed char buf[3000];
+
+        // tick() will do one MIDI clock tick. Then, there's a loop here that
+        // will generate the right number of samples per MIDI tick. The whole
+        // MIDI playback is timed in terms of this value.. there are no forced
+        // delays or anything. It just produces enough samples for each tick, and
+        // the playback of these samples is what makes the timings right.
+        //
+        // This seems to work quite well.
+
+
+        printf("\nOkay, starting sequencing");
+
+        //Tick() will return 0 if there are no more events left to play
+        while(tick(mf))
+        {
+
+                //Some annoying math to compute the number of samples
+                //to syntehsize per each MIDI tick.
+                bpm=mf->div*1000000/tempo;
+                nsmp=SAMPLE_RATE/bpm;
+
+                //Yes we need to do this math each time because the tempo
+                //could have changed.
+
+                // On second thought, this can be moved to the event that
+                //recalculates the tempo, to save a little bit of CPU time.
+                for(l=0; l<nsmp; l++)
+                {
+                        int s1, s2;
+
+                        synthSample(&s1, &s2);
+
+
+                        //16-bit audio because, well, it's better
+                        // But really because ALSA's OSS emulation sounds extremely
+                        //noisy and distorted when in 8-bit mode. I still do not know
+                        //why this happens.
+                        buf[bp]=s1&0XFF; // Low byte first
+                        bp++;
+                        buf[bp]=s1>>8;   //High byte second
+                        bp++;
+
+                        buf[bp]=s2&0XFF; // Low byte first
+                        bp++;
+                        buf[bp]=s2>>8;   //High byte second
+                        bp++;
+
+
+                        //As soon as we produce 2000 bytes of sound,
+                        //write it to the sound card. Why 2000? I have
+                        //no idea. It's 1 AM and I am dead tired.
+                        if(bp>=2000)
+                        {
+                                rb->write(fd, buf, 2000);
+                                bp=0;
+                        }
+                }
+        }
+
+//      unloadFile(mf);
+        printf("\n");
+        rb->close(fd);
+        return 0;
+}