Add this back in, for now. Will turn into real codec later, when plugins support the codec api.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7290 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 238 insertions, 0 deletions

diff --git a/apps/plugins/midi2wav.c b/apps/plugins/midi2wav.c
new file mode 100644
index 0000000..1a87fa8
--- /dev/null
+++ b/apps/plugins/midi2wav.c
@@ -0,0 +1,238 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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 22050
+#define MAX_VOICES 100
+
+
+/* Only define LOCAL_DSP on Simulator or else we're asking for trouble  */
+#if defined(SIMULATOR)
+    /*Enable this to write to the soundcard via a /dsv/dsp symlink in */
+    //#define LOCAL_DSP
+#endif
+
+
+#if defined(LOCAL_DSP)
+/*  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>
+#endif
+
+#include "../../firmware/export/system.h"
+
+#include "../../plugin.h"
+
+//#include "../codecs/lib/xxx2wav.h"
+
+int numberOfSamples IDATA_ATTR;
+long bpm;
+
+#include "midi/midiutil.c"
+#include "midi/guspat.h"
+#include "midi/guspat.c"
+#include "midi/sequencer.c"
+#include "midi/midifile.c"
+#include "midi/synth.c"
+
+
+
+
+int fd=-1; /* File descriptor where the output is written */
+
+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);
+    rb = api;
+    TEST_PLUGIN_API(api);
+    (void)parameter;
+    rb = api;
+
+    if(parameter == NULL)
+    {
+        rb->splash(HZ*2, true, " Play .MID file ");
+        return PLUGIN_OK;
+    }
+
+    rb->splash(HZ, true, parameter);
+    if(midimain(parameter) == -1)
+    {
+        return PLUGIN_ERROR;
+    }
+    rb->splash(HZ*3, true, "FINISHED PLAYING");
+    return PLUGIN_OK;
+}
+
+signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
+
+
+int currentSample IDATA_ATTR;
+int outputBufferPosition IDATA_ATTR;
+int outputSampleOne IDATA_ATTR;
+int outputSampleTwo IDATA_ATTR;
+
+
+int midimain(void * filename)
+{
+
+    printf("\nHello.\n");
+
+    rb->splash(HZ/5, true, "LOADING MIDI");
+
+    struct MIDIfile * mf = loadFile(filename);
+
+    rb->splash(HZ/5, true, "LOADING PATCHES");
+    if (initSynth(mf, "/.rockbox/patchset/patchset.cfg", "/.rockbox/patchset/drums.cfg") == -1)
+    {
+        return -1;
+    }
+
+/*
+ *  This lets you hear the music through the sound card if you are on Simulator
+ *  Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
+ *  your sound device is.
+ */
+
+#if defined(LOCAL_DSP)
+    fd=rb->open("/dsp.raw", O_WRONLY);
+    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;
+#else
+
+/* xxx2wav stuff, removed for now, will move to the real way of outputting sound soon */
+/*
+    file_info_struct file_info;
+    file_info.samplerate = SAMPLE_RATE;
+    file_info.infile = fd;
+    file_info.channels = 2;
+    file_info.bitspersample = 16;
+    local_init("/miditest.tmp", "/miditest.wav", &file_info, rb);
+    fd = file_info.outfile;
+*/
+#endif
+
+
+    rb->splash(HZ/5, true, "  I hope this works...  ");
+
+
+
+
+    /*
+     * 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");
+
+
+    currentSample=0;            /* Sample counting variable */
+    outputBufferPosition = 0;
+
+
+    bpm=mf->div*1000000/tempo;
+    numberOfSamples=SAMPLE_RATE/bpm;
+
+
+
+    /* Tick() will return 0 if there are no more events left to play */
+    while(tick(mf))
+    {
+        /*
+         * Tempo recalculation moved to sequencer.c to be done on a tempo event only
+         *
+         */
+        for(currentSample=0; currentSample<numberOfSamples; currentSample++)
+        {
+
+            synthSample(&outputSampleOne, &outputSampleTwo);
+
+
+            /*
+             * 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.
+             */
+
+            outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; // Low byte first
+            outputBufferPosition++;
+            outputBuffer[outputBufferPosition]=outputSampleOne>>8;   //High byte second
+            outputBufferPosition++;
+
+            outputBuffer[outputBufferPosition]=outputSampleTwo&0XFF; // Low byte first
+            outputBufferPosition++;
+            outputBuffer[outputBufferPosition]=outputSampleTwo>>8;   //High byte second
+            outputBufferPosition++;
+
+
+            /*
+             * 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(outputBufferPosition>=2000)
+            {
+                rb->write(fd, outputBuffer, 2000);
+                outputBufferPosition=0;
+            }
+        }
+    }
+
+    printf("\n");
+
+#if !defined(LOCAL_DSP)
+
+/* again, xxx2wav stuff, removed for now */
+
+    /* close_wav(&file_info); */
+#else
+    rb->close(fd);
+#endif
+    return 0;
+}