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/******************************************/
/*
play_saw.c
by Gary P. Scavone, 2001
Play sawtooth waveforms of distinct frequency.
Takes number of channels and sample rate as
input arguments. Uses blocking functionality.
*/
/******************************************/
#include "RtAudio.h"
#include <iostream.h>
/*
typedef signed long MY_TYPE;
#define FORMAT RtAudio::RTAUDIO_SINT24
#define SCALE 2147483647.0
typedef char MY_TYPE;
#define FORMAT RtAudio::RTAUDIO_SINT8
#define SCALE 127.0
typedef signed short MY_TYPE;
#define FORMAT RtAudio::RTAUDIO_SINT16
#define SCALE 32767.0
typedef signed long MY_TYPE;
#define FORMAT RtAudio::RTAUDIO_SINT32
#define SCALE 2147483647.0
*/
typedef float MY_TYPE;
#define FORMAT RtAudio::RTAUDIO_FLOAT32
#define SCALE 1.0
/*
typedef double MY_TYPE;
#define FORMAT RtAudio::RTAUDIO_FLOAT64
#define SCALE 1.0
*/
#define BASE_RATE 0.005
#define TIME 1.0
void usage(void) {
/* Error function in case of incorrect command-line
argument specifications
*/
cout << "\nuseage: play_saw N fs\n";
cout << " where N = number of channels,\n";
cout << " and fs = the sample rate.\n\n";
exit(0);
}
int main(int argc, char *argv[])
{
int chans, fs, device, buffer_size, stream;
long frames, counter = 0, i, j;
MY_TYPE *buffer;
RtAudio *audio;
double *data;
// minimal command-line checking
if (argc != 3) usage();
chans = (int) atoi(argv[1]);
fs = (int) atoi(argv[2]);
// Open the realtime output device
buffer_size = 256;
device = 0; // default device
try {
audio = new RtAudio(&stream, device, chans, 0, 0,
FORMAT, fs, &buffer_size, 4);
}
catch (RtAudioError &m) {
m.printMessage();
exit(EXIT_FAILURE);
}
frames = (long) (fs * TIME);
data = (double *) calloc(chans, sizeof(double));
try {
buffer = (MY_TYPE *) audio->getStreamBuffer(stream);
audio->startStream(stream);
}
catch (RtAudioError &m) {
m.printMessage();
goto cleanup;
}
cout << "\nPlaying for " << TIME << " seconds." << endl;
while (counter < frames) {
for (i=0; i<buffer_size; i++) {
for (j=0; j<chans; j++) {
buffer[i*chans+j] = (MY_TYPE) (data[j] * SCALE);
data[j] += BASE_RATE * (j+1+(j*0.1));
if (data[j] >= 1.0) data[j] -= 2.0;
}
}
try {
//cout << "frames until no block = " << audio->streamWillBlock(stream) << endl;
audio->tickStream(stream);
}
catch (RtAudioError &m) {
m.printMessage();
goto cleanup;
}
counter += buffer_size;
}
try {
audio->stopStream(stream);
}
catch (RtAudioError &m) {
m.printMessage();
}
cleanup:
audio->closeStream(stream);
delete audio;
if (data) free(data);
return 0;
}
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