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Fixed warnings, adapted to Rockbox coding style, optimized to 78% realtime.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6329 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Stepan Moskovchenko 2005-04-20 21:07:13 +00:00
parent 74e9d545ef
commit 9ec1ff8cf5
7 changed files with 1056 additions and 1115 deletions
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231
apps/plugins/midi2wav.c
View file

@ -20,21 +20,22 @@
#define MAX_VOICES 100
//Only define LOCAL_DSP on Simulator or else we're asking for trouble
/* 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
/*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
/* 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"
@ -53,9 +54,9 @@ long bpm;
int fd=-1; //File descriptor where the output is written
int fd=-1; /* File descriptor where the output is written */
extern long tempo; //The sequencer keeps track of this
extern long tempo; /* The sequencer keeps track of this */
struct plugin_api * rb;
@ -66,27 +67,28 @@ 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;
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;
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 ... !
signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
int currentSample IDATA_ATTR;
@ -98,132 +100,133 @@ int outputSampleTwo IDATA_ATTR;
int midimain(void * filename)
{
printf("\nHello.\n");
printf("\nHello.\n");
rb->splash(HZ/5, true, "LOADING MIDI");
rb->splash(HZ/5, true, "LOADING MIDI");
struct MIDIfile * mf = loadFile(filename);
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;
}
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.
/*
* 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;
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 = 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 = 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;
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
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;
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, " Starting Playback ");
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.
*/
// 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");
printf("\nOkay, starting sequencing");
currentSample=0; /* Sample counting variable */
outputBufferPosition = 0;
currentSample=0; //Sample counting variable
outputBufferPosition = 0;
bpm=mf->div*1000000/tempo;
numberOfSamples=SAMPLE_RATE/bpm;
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))
{
/* 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++)
{
//Some annoying math to compute the number of samples
//to syntehsize per each MIDI tick.
//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(currentSample=0; currentSample<numberOfSamples; currentSample++)
{
synthSample(&outputSampleOne, &outputSampleTwo);
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++;
/*
* 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]=outputSampleTwo&0XFF; // Low byte first
outputBufferPosition++;
outputBuffer[outputBufferPosition]=outputSampleTwo>>8; //High byte second
outputBufferPosition++;
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;
}
}
}
/*
* 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;
}
}
}
// unloadFile(mf);
printf("\n");
printf("\n");
#if !defined(LOCAL_DSP)
close_wav(&file_info);
close_wav(&file_info);
#else
rb->close(fd);
rb->close(fd);
#endif
return 0;
return 0;
}