an open-source digital signal processing and sound synthesis language
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resample - convert sampling rate


resample -r NEW_SRATE [options] inputfile [outputfile]


resample can do two kinds of sampling rate conversion: (1) Kaiser-windowed low-pass filter (better) (2) linear interpolation only, no filter (faster)

For (1), use either no option, in which case you get the default, decent-quality resampling filter; use the -a option for a better quality filter; or use a combination of the -f, -b and -l options to design your own filter. For (2), use the -i option.


Here are the options:
     -a       triple-A quality resampling filter                     
     -f NUM   rolloff Frequency (0 < freq <= 1)       [default: 0.9] 
     -b NUM   Beta ( >= 1)                            [default: 9.0] 
     -l NUM   filter Length (odd number <= 65)        [default: 65]  
     -n       No interpolation of filter coefficients (faster)       
     -i       resample by linear Interpolation, not with filter      
     -t       Terse (don't print out so much)                        
     -v       print Version of program and quit                      
  If no output file specified, writes to "inputfile.resamp".