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.
Flags:
-P num = pitch transposition ratio (srate / r) [don't specify both P and r]
-Q num =quality factor (1, 2, 3, 4 or 5: default = 3)
-i filnam = auxiliary breakpoints file (no breakpoint by default. i.e. No ratio change)
-r num = output sample rate (must be specified if no P)
-o fnam = sound output filename (default: test.wav)
-A = create an AIFF format output soundfile
-J = create an IRCAM format output soundfile
-W = create a WAV format output soundfile (default after Csound version 6.15)
-h = no header on output soundfile
-c = 8-bit signed_char sound samples
-a = alaw sound samples
-8 = 8-bit unsigned_char sound samples
-u = ulaw sound samples
-s = short_int (16 bit) sound samples (default)
-3 = 24-bit sound samples
-l = long_int sound samples
-f = float sound samples
-r N = orchestra srate override
-K = Do not generate PEAK chunks
-R = continually rewrite header while writing soundfile (WAV/AIFF)
-H# = print a heartbeat style 1, 2 or 3 at each soundfile write
-N = notify (ring the bell) when done
This program performs arbitrary sample-rate conversion with high fidelity using the libsamplerate library.
The five levels of accuracy are: