rockbox/apps/plugins/midi2wav.c
Jens Arnold 4d6374c923 Get rid of the 'center' parameter for splashes. There were only 2 of almost 500 splashes which were not centered.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12807 a1c6a512-1295-4272-9138-f99709370657
2007-03-16 21:56:08 +00:00

238 lines
6.2 KiB
C

/***************************************************************************
* __________ __ ___.
* 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" */
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 */
struct plugin_api * rb;
enum plugin_status plugin_start(struct plugin_api* api, 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/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, "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;
}