Removing deprecated code.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19068 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 0 insertions, 235 deletions

diff --git a/apps/plugins/midi2wav.c b/apps/plugins/midi2wav.c
deleted file mode 100644
index cee9d28b20..0000000000
--- a/apps/plugins/midi2wav.c
+++ /dev/null
@@ -1,235 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
- *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
- *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
- *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
- *                     \/            \/     \/    \/            \/
- *
- * Copyright (C) 2005 Stepan Moskovchenko
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * 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" */
-
-PLUGIN_HEADER
-
-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 */
-
-const struct plugin_api * rb;
-
-
-enum plugin_status plugin_start(const struct plugin_api* api, const void* parameter)
-{
-    (void)parameter;
-    rb = api;
-
-    if(parameter == NULL)
-    {
-        rb->splash(HZ*2, "Play .MID file");
-        return PLUGIN_OK;
-    }
-
-    rb->splash(HZ, parameter);
-    if(midimain(parameter) == -1)
-    {
-        return PLUGIN_ERROR;
-    }
-    rb->splash(HZ*3, "FINISHED PLAYING");
-    /* Return PLUGIN_USB_CONNECTED to force a file-tree refresh */
-    return PLUGIN_USB_CONNECTED;
-}
-
-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, "LOADING MIDI");
-
-    struct MIDIfile * mf = loadFile(filename);
-
-    rb->splash(HZ/5, "LOADING PATCHES");
-    if (initSynth(mf, ROCKBOX_DIR "/patchset/patchset.cfg",
-        ROCKBOX_DIR "/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, "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;
-}