2acc0ac542
later. We still need to hunt down snippets used that are not. 1324 modified files... http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
235 lines
6.3 KiB
C
235 lines
6.3 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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*
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* Copyright (C) 2005 Stepan Moskovchenko
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#define SAMPLE_RATE 22050
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#define MAX_VOICES 100
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/* Only define LOCAL_DSP on Simulator or else we're asking for trouble */
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#if defined(SIMULATOR)
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/*Enable this to write to the soundcard via a /dsv/dsp symlink in */
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/*#define LOCAL_DSP */
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#endif
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#if defined(LOCAL_DSP)
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/* This is for writing to the DSP directly from the Simulator */
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#include <stdio.h>
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#include <stdlib.h>
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#include <linux/soundcard.h>
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#include <sys/ioctl.h>
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#endif
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#include "../../firmware/export/system.h"
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#include "../../plugin.h"
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/*#include "../codecs/lib/xxx2wav.h" */
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PLUGIN_HEADER
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int numberOfSamples IDATA_ATTR;
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long bpm;
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#include "midi/midiutil.c"
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#include "midi/guspat.h"
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#include "midi/guspat.c"
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#include "midi/sequencer.c"
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#include "midi/midifile.c"
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#include "midi/synth.c"
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int fd=-1; /* File descriptor where the output is written */
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extern long tempo; /* The sequencer keeps track of this */
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const struct plugin_api * rb;
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enum plugin_status plugin_start(const struct plugin_api* api, const void* parameter)
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{
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(void)parameter;
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rb = api;
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if(parameter == NULL)
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{
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rb->splash(HZ*2, "Play .MID file");
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return PLUGIN_OK;
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}
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rb->splash(HZ, parameter);
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if(midimain(parameter) == -1)
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{
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return PLUGIN_ERROR;
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}
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rb->splash(HZ*3, "FINISHED PLAYING");
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/* Return PLUGIN_USB_CONNECTED to force a file-tree refresh */
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return PLUGIN_USB_CONNECTED;
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}
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signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
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int currentSample IDATA_ATTR;
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int outputBufferPosition IDATA_ATTR;
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int outputSampleOne IDATA_ATTR;
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int outputSampleTwo IDATA_ATTR;
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int midimain(void * filename)
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{
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printf("\nHello.\n");
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rb->splash(HZ/5, "LOADING MIDI");
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struct MIDIfile * mf = loadFile(filename);
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rb->splash(HZ/5, "LOADING PATCHES");
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if (initSynth(mf, ROCKBOX_DIR "/patchset/patchset.cfg",
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ROCKBOX_DIR "/patchset/drums.cfg") == -1)
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{
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return -1;
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}
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/*
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* This lets you hear the music through the sound card if you are on Simulator
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* Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
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* your sound device is.
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*/
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#if defined(LOCAL_DSP)
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fd=rb->open("/dsp.raw", O_WRONLY);
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int arg, status;
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int bit, samp, ch;
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arg = 16; /* sample size */
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status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
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status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
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bit=arg;
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arg = 2; /* Number of channels, 1=mono */
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status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
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status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
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ch=arg;
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arg = SAMPLE_RATE; /* Yeah. sampling rate */
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status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
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status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
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samp=arg;
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#else
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/* xxx2wav stuff, removed for now, will move to the real way of outputting sound soon */
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/*
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file_info_struct file_info;
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file_info.samplerate = SAMPLE_RATE;
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file_info.infile = fd;
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file_info.channels = 2;
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file_info.bitspersample = 16;
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local_init("/miditest.tmp", "/miditest.wav", &file_info, rb);
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fd = file_info.outfile;
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*/
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#endif
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rb->splash(HZ/5, "I hope this works...");
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/*
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* tick() will do one MIDI clock tick. Then, there's a loop here that
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* will generate the right number of samples per MIDI tick. The whole
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* MIDI playback is timed in terms of this value.. there are no forced
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* delays or anything. It just produces enough samples for each tick, and
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* the playback of these samples is what makes the timings right.
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*
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* This seems to work quite well.
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*/
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printf("\nOkay, starting sequencing");
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currentSample=0; /* Sample counting variable */
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outputBufferPosition = 0;
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bpm=mf->div*1000000/tempo;
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numberOfSamples=SAMPLE_RATE/bpm;
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/* Tick() will return 0 if there are no more events left to play */
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while(tick(mf))
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{
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/*
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* Tempo recalculation moved to sequencer.c to be done on a tempo event only
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*
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*/
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for(currentSample=0; currentSample<numberOfSamples; currentSample++)
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{
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synthSample(&outputSampleOne, &outputSampleTwo);
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/*
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* 16-bit audio because, well, it's better
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* But really because ALSA's OSS emulation sounds extremely
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* noisy and distorted when in 8-bit mode. I still do not know
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* why this happens.
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*/
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outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; /* Low byte first */
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outputBufferPosition++;
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outputBuffer[outputBufferPosition]=outputSampleOne>>8; /*High byte second */
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outputBufferPosition++;
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outputBuffer[outputBufferPosition]=outputSampleTwo&0XFF; /* Low byte first */
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outputBufferPosition++;
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outputBuffer[outputBufferPosition]=outputSampleTwo>>8; /*High byte second */
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outputBufferPosition++;
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/*
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* As soon as we produce 2000 bytes of sound,
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* write it to the sound card. Why 2000? I have
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* no idea. It's 1 AM and I am dead tired.
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*/
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if(outputBufferPosition>=2000)
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{
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rb->write(fd, outputBuffer, 2000);
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outputBufferPosition=0;
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}
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}
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}
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printf("\n");
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#if !defined(LOCAL_DSP)
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/* again, xxx2wav stuff, removed for now */
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/* close_wav(&file_info); */
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#else
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rb->close(fd);
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#endif
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return 0;
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}
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