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Linux 2.6 - man page for srconv (linux section 1)

SRCONV(1)			  The Canonical Csound Reference			SRCONV(1)

NAME
       srconv - Converts the sample rate of an audio file. .

DESCRIPTION
       Converts the sample rate of an audio file at sample rate Rin to a sample rate of Rout.
       Optionally the ratio (Rin / Rout) may be linearly time-varying according to a set of
       (time, ratio) pairs in an auxiliary file.

SYNTAX
	   srconv [flags] infile

INITIALIZATION
       Flags:

       o   -P num = pitch transposition ratio (srate / r) [don't specify both P and r]

       o   -P num = pitch transposition ratio (srate / r) [don't specify both P and r]

       o   -Q num =quality factor (1, 2, 3, or 4: default = 2)

       o   -i filnam = auxiliary breakpoints file (no breakpoint by default. i.e. No ratio
	   change)

       o   -r num = output sample rate (must be specified)

       o   -o fnam = sound output filename

       o   -A = create an AIFF format output soundfile

       o   -J = create an IRCAM format output soundfile

       o   -W = create a WAV format output soundfile

       o   -h = no header on output soundfile

       o   -c = 8-bit signed_char sound samples

       o   -a = alaw sound samples

       o   -8 = 8-bit unsigned_char sound samples

       o   -u = ulaw sound samples

       o   -s = short_int sound samples

       o   -l = long_int sound samples

       o   -f = float sound samples

       o   -r N = orchestra srate override

       o   -K = Do not generate PEAK chunks

       o   -R = continually rewrite header while writing soundfile (WAV/AIFF)

       o   -H# = print a heartbeat style 1, 2 or 3 at each soundfile write

       o   -N = notify (ring the bell) when score or miditrack is done

       o   -- fnam = log output to file

       This program performs arbitrary sample-rate conversion with high fidelity. The method is
       to step through the input at the desired sampling increment, and to compute the output
       points as appropriately weighted averages of the surrounding input points. There are two
       cases to consider:

	1. sample rates are in a small-integer ratio - weights are obtained from table.

	2. sample rates are in a large-integer ratio - weights are linearly interpolated from
	   table.

       Calculate increment: if decimating, then window is impulse response of low-pass filter
       with cutoff frequency at half of output sample rate; if interpolating, then window is
       impulse response of lowpass filter with cutoff frequency at half of input sample rate.

CREDITS
       Author: Mark Dolson

       August 26, 1989

       Author: John ffitch

       December 30, 2000

AUTHORS
       Barry Vercoe
       MIT Media Lab

	   Author.

       Dan Ellis
       MIT Media Lab,
		 Cambridge
		 Massachussetts

	   Author.

COPYRIGHT
5.07					    06/23/2009					SRCONV(1)


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