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| /*
A simple sound library for CSE 20211 by Douglas Thain (dthain@nd.edu).
This work is made available under the Creative Commons Attribution license.
https://creativecommons.org/licenses/by/4.0/
For course assignments, you should not change this file.
For complete documentation, see:
http://www.nd.edu/~dthain/courses/cse20211/fall2013/wavfile
*/
#include <stdio.h>
#include <math.h>
#include <stdlib.h>
#include <time.h>
#include <string.h>
#include <errno.h>
#include <inttypes.h>
#define SAMPLES_PER_SECOND 44100
const int NUM_SAMPLES = (SAMPLES_PER_SECOND * 2);
struct wavfile_header {
char riff_tag[4];
int riff_length;
char wave_tag[4];
char fmt_tag[4];
int fmt_length;
short audio_format;
short num_channels;
int sample_rate;
int byte_rate;
short block_align;
short bits_per_sample;
char data_tag[4];
int data_length;
};
FILE * wavfile_open( const char *filename )
{
struct wavfile_header header;
int samples_per_second = SAMPLES_PER_SECOND;
// int bits_per_sample = 16;
int bits_per_sample = 8;
strncpy(header.riff_tag,"RIFF",4);
strncpy(header.wave_tag,"WAVE",4);
strncpy(header.fmt_tag,"fmt ",4);
strncpy(header.data_tag,"data",4);
header.riff_length = 0;
header.fmt_length = 16;
header.audio_format = 1;
header.num_channels = 1;
header.sample_rate = samples_per_second;
header.byte_rate = samples_per_second*(bits_per_sample/8);
header.block_align = bits_per_sample/8;
header.bits_per_sample = bits_per_sample;
header.data_length = 0;
FILE * file = fopen(filename,"wb+");
if(!file) return 0;
fwrite(&header,sizeof(header),1,file);
fflush(file);
return file;
}
void wavfile_write( FILE *file, short data[], int length )
{
fwrite(data,sizeof(short),length,file);
}
void wavfile_close( FILE *file )
{
int file_length = ftell(file);
int data_length = file_length - sizeof(struct wavfile_header);
fseek(file,sizeof(struct wavfile_header) - sizeof(int),SEEK_SET);
fwrite(&data_length,sizeof(data_length),1,file);
int riff_length = file_length - 8;
fseek(file,4,SEEK_SET);
fwrite(&riff_length,sizeof(riff_length),1,file);
fclose(file);
}
int main()
{
short waveform[NUM_SAMPLES];
double frequency = 440.0;
int volume = 32000;
int length = NUM_SAMPLES;
int i;
for(i=0;i<length;i++) {
double t = (double) i / SAMPLES_PER_SECOND;
//pour 16bits waveform[i] = volume*sin(frequency*t*2*M_PI);
waveform[i] = 127 + (volume*sin(frequency*t*2*M_PI));
}
FILE * f = wavfile_open("essai_onefile.wav");
if(!f) {
printf("couldn't open essai_onefile.wav for writing: %s",strerror(errno));
return 1;
}
wavfile_write(f,waveform,length);
wavfile_close(f);
return 0;
} |
Besoin d'un petit coup de main pour corriger une sinusoIde
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Envoyé par dalfab
) on voit des techniques d'optimisation (genre v <<=1; v <<=1; v <<=1; v <<=1; v <<=1;...) qui n'ont plus lieu d'être aujourd'hui.
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