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sfkaiser − Apply a "Kaiser" filter to a soundfile. |
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sfkaiser inputSound.wav outputSound.wav |
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sfkaiser reads in inputSound.wav then writes out a filtered version to outputSound.wav . |
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There are no options to the sfkaiser program, since it prompts the user for all the information it needs. |
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sfkaiser snd1.wav snd2.aiff |
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Apply the filter to snd1.wav and put the results in snd2.aiff . |
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The program will prompt the user for |
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the type of filter desired (lp [lowpass], hp [highpass], bp [bandpass], bs [bandstop]); the desired stopband attenuation in dB; the passband edge frequency in Hertz; the stopband edge frequency in Hertz. |
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The program will then calculate the coefficients according to the Kaiser filter formulas, filter the file, and save the result in snd1.aiff . |
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This program will only work with WAVE, AIFF, AIFC, and the old NeXT/Sun "snd" format soundfiles, that are 16-bit PCM encoded, with 1 or 2 channels. |
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Arun Chandra <arunc@evergreen.edu> sfkaiser is free software. |