Shawn> 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. 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. -- Naoki Shibata e-mail: [EMAIL PROTECTED] -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
- [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 Wächter
- 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
- Re: [MP3 ENCODER] default... Istvan Varga
- [MP3 ENCODER] Lame_Enc.dl... Zia Mazhar