🚀 go-pugleaf

ketil@edb.tih.no (Ketil Albertsen,TIH) writes:

>In article <1993Jan23.151043.23034@impmh.uucp>, cm@impmh.uucp (Colin Manning) 
>writes:
>
>>According the the ms documentation, they accept data in 44.1Khz samples,
>>20.05Kh and 10.025Khz. ADPCM is sampled at different rate.
>>
>>Anyone any idea how easy/difficult it is to convert audio data sampled at
>>one rate to another ?
>
>Now that you ask: Yes, it is difficult... :-)
>
>If the only difference was the sampling rate, plain linear interpolation
>would probably be satisfactory - and anyone who can do simple multiplication
>and division in c could do the job.
>
>[and goes on to explain how decoding ADPCM is difficult]

Two remarks.  First, while linear interpolation works for rate
conversion, the result is rather poor: it can adds lots of digital
noise.  Unfortunately the techniques required to remove this noise are
difficult -- digital filters are hard to design, and hard to implement
in software.  (Read comp.dsp if you're interested in such things.)

Second, a free implementation of ADPCM exist.  This ADPCM (en/de)coder
is available by anonymous ftp from ftp.cwi.nl [192.16.184.180],
directory pub/audio, file adpcm.shar.  This is a library written in C
-- you'll have to write a program around it to use it.

Happy hacking,

--Guido van Rossum, CWI, Amsterdam <Guido.van.Rossum@cwi.nl>