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Full Discussion: A Function To Create A 1 Second Sinewave WAVE Beep File In Python.
Top Forums Programming A Function To Create A 1 Second Sinewave WAVE Beep File In Python. Post 302930838 by DGPickett on Wednesday 7th of January 2015 04:29:09 PM
Old 01-07-2015
Reminds me of an early C project I wrote myself 21 yrs. ago, which does a pure 440 hz sine wave with one byte samples and low sample rate:

Code:
 
a440.c
======
 
#include <stdio.h>
#ifdef BORLAND
#include <stdlib.h>
#endif
#include <math.h>
char hdr[]={'R', 'I', 'F', 'F', 0, 0, 0, 0, 'W', 'A', 'V', 'E', 'f', 'm', 't', 32, 16, 0, 0, 0, 1, 0, 1, 0, 17, '+', 0, 0, 17, '+', 0, 0, 1, 0, 8, 0, 'd', 'a', 't', 'a', 0, 0, 0, 0};
FILE * f ;
main(argc,argv)
int argc;
char**argv;
{
  #define SAMPLES 600L
  #define FREQ 440
  #define SPS 11025
  #define PI 3.1415927
  long i ;
  float w, x ;
  if ( argc < 2 )
  {
    fprintf( stderr, "Usage is '%s filename'\n", argv[0] ) ;
    exit( -1 ) ;
  }
  if ( !( f = fopen( argv[1], "wb" ) ) )
  {
    fprintf( stderr, "Can't open '%s' to write\n", argv[1] ) ;
    exit( -2 ) ;
  }
  w = 2 * PI * FREQ / SPS ;
  i = SAMPLES + 36 ;
  hdr[4] = i % 256 ;
  hdr[5] = ( i / 256 ) % 256 ;
  hdr[6] = ( i / 65536L ) % 256 ;
  hdr[7] = i / 16777216L ;
  i = SAMPLES ;
  hdr[40] = i % 256 ;
  hdr[41] = ( i / 256 ) % 256 ;
  hdr[42] = ( i / 65536L ) % 256 ;
  hdr[43] = i / 16777216L ;
  for( i = 0L ; i < sizeof(hdr) ; i++ )
  {
    putc( hdr[i], f ) ;
  }
  i = 0 ;
  while ( ++i <= SAMPLES )
  {
    putc( (char)( 128.0 + 127.999 * sin( w * i ) ), f ) ;
  }
  fclose( f ) ;
  exit( 0 ) ;
}

Last edited by DGPickett; 01-07-2015 at 05:46 PM..
This User Gave Thanks to DGPickett For This Post:
wisecracker
 

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LEARN ABOUT CENTOS

sfconvert

SFCONVERT(1)															      SFCONVERT(1)

NAME

       sfconvert - convert audio files

SYNOPSIS

       sfconvert infile outfile [output options]

DESCRIPTION

       The sfconvert command converts an audio file to another file format or data format.

OPTIONS

       The following keywords specify the format of the output sound file:

       format f
	   Output file format.	f must be one of the following:

	   aiff    Audio Interchange File Format
	   aifc    AIFF-C
	   next    NeXT/Sun .snd/.au Format
	   wave    MS RIFF WAVE
	   bicsf   Berkeley/IRCAM/CARL Sound File Format
	   smp	   Sample Vision Format
	   voc	   Creative Voice File Format
	   nist    NIST SPHERE Format
	   caf	   Core Audio Format
	   flac    FLAC

       compression c

	   ulaw      G.711 mu-law
	   alaw      G.711 A-law
	   ima	     IMA ADPCM
	   msadpcm   Microsoft ADPCM
	   flac      FLAC
	   alac      Apple Lossless Audio Codec

       byteorder e
	   Output byte order.  e may be either big or little.

       channels n
	   Number of output channels.  n is 1 for mono, and 2 for stereo.

       integer n s
	   Convert to integer audio data.  n indicates the output sample width in bits.  s specifies the sample format and may be either 2scomp
	   for 2's complement signed data or unsigned for unsigned data.

       float m
	   Convert to floating-point audio data with a maximum amplitude of m (usually 1.0). The integer and float options are mutually exclusive.

       The following options take no additional arguments:

       --help, -h
	   Print help message.

       --version, -v
	   Print version information.

SEE ALSO

       sfinfo(1)

AUTHOR

       Michael Pruett <michael@68k.org>

Audio File Library 0.3.6					    03/06/2013							      SFCONVERT(1)
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