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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- [MP3 ENCODER] default high pass filtering Gabriel Bouvigne
- Re: [MP3 ENCODER] default high pass filtering Greg Maxwell
- Re: [MP3 ENCODER] default high pass filtering Segher Boessenkool
- Re: [MP3 ENCODER] default high pass filtering Mark Taylor
- Re: [MP3 ENCODER] default high pass filter... Shawn Riley
- Re: [MP3 ENCODER] default high pass fi... David Balazic
- Re: [MP3 ENCODER] default high pass fi... Naoki Shibata
- Re: [MP3 ENCODER] default high pa... Matthias W�chter
- Re: [MP3 ENCODER] default hig... Naoki Shibata
- Re: [MP3 ENCODER] default high pass fi... Scott Manley
- Re: [MP3 ENCODER] default high pass fi... Segher Boessenkool
- Re: [MP3 ENCODER] default high pass fi... Christian Schepke
- Re: [MP3 ENCODER] default high pa... Segher Boessenkool
- Re: [MP3 ENCODER] default hig... Istvan Varga
- Re: [MP3 ENCODER] default... Segher Boessenkool
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