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Some shifting optimizations. Working code. 50% realtime.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6323 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Stepan Moskovchenko 2005-04-19 15:57:07 +00:00
parent c3d0a229cc
commit 1f5fb99819
6 changed files with 238 additions and 201 deletions
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65
apps/plugins/midi2wav.c
View file

@ -16,7 +16,7 @@
*
****************************************************************************/
#define SAMPLE_RATE 48000
#define SAMPLE_RATE 22050
#define MAX_VOICES 100
@ -37,6 +37,12 @@
#include "../../plugin.h"
#include "lib/xxx2wav.h"
int numberOfSamples IDATA_ATTR;
long bpm;
#include "midi/midiutil.c"
#include "midi/guspat.h"
#include "midi/guspat.c"
@ -46,7 +52,6 @@
#include "lib/xxx2wav.h"
int fd=-1; //File descriptor where the output is written
@ -58,6 +63,7 @@ struct plugin_api * rb;
enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
{
TEST_PLUGIN_API(api);
@ -80,6 +86,14 @@ enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
return PLUGIN_OK;
}
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)
{
@ -89,9 +103,6 @@ int midimain(void * filename)
rb->splash(HZ/5, true, "LOADING MIDI");
struct MIDIfile * mf = loadFile(filename);
long bpm, nsmp, l;
int bp=0;
rb->splash(HZ/5, true, "LOADING PATCHES");
if (initSynth(mf, "/.rockbox/patchset/patchset.cfg", "/.rockbox/patchset/drums.cfg") == -1)
@ -125,7 +136,7 @@ int midimain(void * filename)
samp=arg;
#else
file_info_struct file_info;
file_info.samplerate = 48000;
file_info.samplerate = SAMPLE_RATE;
file_info.infile = fd;
file_info.channels = 2;
file_info.bitspersample = 16;
@ -134,12 +145,11 @@ int midimain(void * filename)
#endif
rb->splash(HZ/5, true, " START PLAYING ");
rb->splash(HZ/5, true, " Starting Playback ");
signed char buf[3000];
// 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
@ -152,49 +162,56 @@ int midimain(void * filename)
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))
{
//Some annoying math to compute the number of samples
//to syntehsize per each MIDI tick.
bpm=mf->div*1000000/tempo;
nsmp=SAMPLE_RATE/bpm;
//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(l=0; l<nsmp; l++)
for(currentSample=0; currentSample<numberOfSamples; currentSample++)
{
int s1, s2;
synthSample(&s1, &s2);
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.
buf[bp]=s1&0XFF; // Low byte first
bp++;
buf[bp]=s1>>8; //High byte second
bp++;
outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; // Low byte first
outputBufferPosition++;
outputBuffer[outputBufferPosition]=outputSampleOne>>8; //High byte second
outputBufferPosition++;
buf[bp]=s2&0XFF; // Low byte first
bp++;
buf[bp]=s2>>8; //High byte second
bp++;
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(bp>=2000)
if(outputBufferPosition>=2000)
{
rb->write(fd, buf, 2000);
bp=0;
rb->write(fd, outputBuffer, 2000);
outputBufferPosition=0;
}
}
}