inline void synthbuf(void)
{
long *outptr;
register int i;
static int currentSample=0;
int synthtemp[2];
#ifndef SYNC
if(lastswap==swap) return;
lastswap=swap;
outptr=(swap ? gmbuf : gmbuf+BUF_SIZE);
#else
outptr=gmbuf;
#endif
for(i=0; i<BUF_SIZE/2; i++)
{
synthSample(&synthtemp[0], &synthtemp[1]);
currentSample++;
*outptr=((synthtemp[0]&0xFFFF) << 16) | (synthtemp[1]&0xFFFF);
outptr++;
if(currentSample==numberOfSamples)
{
if(playState == STATE_PLAYING)
{
stepFlag=1;
}
currentSample=0;
}
}
}
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;
}
