quad2samp.c

/******************

File: quad2samp.c
Purpose: take file of quadruples (freq, dur, decay, amplitude) as input,
produce file of sampled waveform as output
Author: Hakon Hallgrimur
Date: March 17, 2011

Example:

1. Create file “foo.quad” with contents

220 1.0 0.5 0.2
330 1.0 0.5 0.5
440 1.0 0.5 1.0

Here 220 is the frequency of the first sound in Hertz, 1.0
is the duration in seconds, 0.5 is the decay constant (larger
for more sustained sound, smaller for more percussive sound),
and 0.2 is the amplitude (larger = louder, smaller = quieter).

2. Complie this file:

% gcc quad2samp.c -o quad2samp

3. Create a text file of the sampled waveform:

% ./quad2samp foo.quad foo.samp

4. Create the corresponding .wav file:

% text2sf foo.samp foo.wav 44100 1 0.90

Here 44100 Hertz is the sample rate, 1 is the number
of channels, and 0.90 is the gain.

Note the output to the terminal window:

TEXT2SF: convert text audio data to soundfile
copying….
Done. 132300 sample frames copied to foo.wav
PEAK information:
CH 1:0.8989 at 2.0006 secs

The program text2sf (See: “The Audio Programming Book”,
Boulanger and Lazzarini, eds) has processed 132300 sample
frames, which checks: 3.0 seconds of sound x 44,100 sample
frames per second = 132,300 samples.

Note that foo.samp is a 1.4 MB file, while foo.wav is a
260 K file. So 0.47 MB/sec for foo.samp, 87 KB/sec for
foo.wav.

5. Play the file:

% play foo.wav

(Mac: ) This command brings up the QuickTime player loaded with foo.wav.
Click the play button to listen to foo.wav.

NOTE: Since I am lazy, I usually package all of the above as follows:

a. Create a file m.sh with contents

gcc quad2samp.c -o quad2samp
./quad2samp foo.quad foo.samp
text2sf foo.samp foo.wav 44100 1 .90
rm foo.samp
play foo.wav

b. Make an alias:

% alias m ‘sh m.sh’

c. Run the whole shebang:

% m

Thus when experimenting, you have only to type a single character to
test the each step.

*************/

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>

// Strip trailing newline
void stripnl(char *str) {
  while(strlen(str) && ( (str[strlen(str) - 1] == 13) || 
       ( str[strlen(str) - 1] == 10 ))) {
    str[strlen(str) - 1] = 0;
  }
}
 

#ifndef M_PI
#define M_PI (3.141592654)
#endif

#define SRATE 44100

int main(int argc, char **argv) {

    FILE *infile, *outfile;
    char line[100];
    char *tok;
    float freq, dur, decay, amplitude;

    int srate = SRATE;
  
	int sr, nsamps, phase;
	double samp,k,a,x;
	double twopi = 2.0 * M_PI;
	double angleincr;
	double maxsamp = 0.0;


     // Open infile.  If NULL is returned there was an error
    if((infile = fopen(argv[1], "r")) == NULL) {
      printf("Error Opening Input Fle.\n");
      exit(1);
    } 

    // Open outfile.  If NULL is returned there was an error 
    if((outfile = fopen(argv[2], "w")) == NULL) {
      printf("Error Opening Output File.\n");
      exit(1);
    } 
  
   float ATTACK = atof(argv[3]);
   float RELEASE = atof(argv[4]);
  
   phase = 0; // global phase -- runs from start to finish of waveform
   while( fgets(line, sizeof(line), infile) != NULL ) {
   
     // Get each line from the infile 
     
    
    // Parse the line to recover the elements
    // of tuple as floats
     tok = strtok(line, " "); freq = atof(tok);
     tok = strtok(NULL, " "); dur = atof(tok);
     tok = strtok(NULL, " "); decay = atof(tok);
     tok = strtok(NULL, " "); amplitude = atof(tok);
     
     // Set up parameters
     nsamps = (int)(dur * srate);
	 angleincr = twopi * freq / srate;
	 k = dur/nsamps;
	 float dampingFactor = exp(-k/decay);
	 float dampingAmplitude = 1.0;
	 
	 int i; // local phase - use for a single tuple
	 
	 // Parameters for basic waveform shaping
	 float endAttack = ATTACK*nsamps;
	 float releaseSamples = RELEASE*nsamps;
	 float beginRelease = nsamps - releaseSamples;
	 
	 for( i = 0; i < nsamps; i++ ){
	    phase++;
	    float A, attackAmplitude, releaseAmplitude;
	    
	    // Compute attack amplitude
	    if ( i  beginRelease ) {
	       float j = (nsamps - i)/(nsamps - beginRelease);
	       releaseAmplitude = pow(j, 2);
	     } else {
	       releaseAmplitude = 1.0;
	    }
	    
	    // Update dampingAmplitude
	    dampingAmplitude = dampingAmplitude*dampingFactor;
	    
	    // Final amplitude is a product of amplitudes
	    A = attackAmplitude*releaseAmplitude*dampingAmplitude*amplitude;
	    
	    // Sample the shaped sine wave
     	samp = A*sin(angleincr*phase);

		// Write the sample to file      
		fprintf(outfile,"%.8lf\n",samp);		
	 }
  }
  
  // Close the files
  fclose(infile); 
  fclose(outfile);
}

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