On Wed, 10 May 2000, Naoki Shibata wrote:

> 
>> 2- Resample the input file to something ridiculously low like
>> 40Hz, & subtract this from the original WAV. (Although that would
>> probably require the above method anyway.)
> 
> 
>   Assume that the original sampling frequency is f.
>   Downsampling to sampling frequency f/n (n is integer) is easy, since
> this is done by picking up every nth sample.

Ooh, ouch - this leads to heavy aliasing. Don't do that! Before any
resampling to lower frequencies, you have to do filtering to that
frequency's half.

>   Upsampling this data to frequency f requires filter with large order,
> but since almost all input sample is zero, this process requires only
> small computational power.

Sehr Wus,
- Matthias

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