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Custom FIR Filter Implementation, TOS #8 / trolling example
tkrieger
post Jul 20 2004, 20:51
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Is there a way to either alter the coefficients of an existing FIR filter or design a new FIR filter, which can be used to convolve with the raw CD data in real time?? I would be interested in trying a Lanczos filter, which works great with image files- I'm very curious to see if such implementation would also work with audio playback.

Responses are appreciated!!

Todd
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SebastianG
post Jul 20 2004, 22:35
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QUOTE (tkrieger @ Jul 20 2004, 11:51 AM)
Is there a way to either alter the coefficients of an existing FIR filter or design a new FIR filter, which can be used to convolve with the raw CD data in real time??  I would be interested in trying a Lanczos filter, which works great with image files- I'm very curious to see if such implementation would also work with audio playback.

Responses are appreciated!!

Todd
*


I'm pretty sure it can be done via Foobar2000 - get a CDDA reading plugin and the convolver plugin. The convolver plugin loads a certain impulse response (wave file) and uses it for convolution.

But...
What's your goal ?
Lanczos is AFAIK for interpolation / resampling.
It's probably less useful for audio.
Better do "sinc interpolation"

Funny thing: I'm currently working on a real-time-convolver in Java

bye,
Sebi

This post has been edited by SebastianG: Jul 20 2004, 22:38
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tkrieger
post Jul 21 2004, 05:38
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QUOTE (SebastianG @ Jul 20 2004, 02:35 PM)
I'm pretty sure it can be done via Foobar2000 - get a CDDA reading plugin and the convolver plugin. The convolver plugin loads a certain impulse response (wave file) and uses it for convolution.

But...
What's your goal ?
Lanczos is AFAIK for interpolation / resampling.
It's probably less useful for audio.
Better do "sinc interpolation"

Funny thing: I'm currently working on a real-time-convolver in Java

bye,
Sebi
*


I want to replace the classic "windowed sinc" function with the Lanczos3 function. It has characteristic like "sinc" except it has a faster build-up and decay. I think it would strike a happy medium between the classic "windowed sinc" (poor time response) and "filter less" implementations (excessive HF modulation). If Lanczos works with audio as it does with picture images, I think it would closest to restoring the digital signal to its "pre-digitized" state.

So I want to replace the "sinc interpolation" with the "Lanczos3 interpolation." And I'm pretty sure this is done by modifying the coefficients in the FIR filter. Either take a customizable "FIR filter" plug-in or change the coefficients in the existing FIR filter in source code with re-compile.
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Axon
post Jul 21 2004, 06:31
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QUOTE (tkrieger @ Jul 20 2004, 08:38 PM)
I want to replace the classic "windowed sinc" function with the Lanczos3 function.  It has characteristic like "sinc" except it has a faster build-up and decay.  I think it would strike a happy medium between the classic "windowed sinc" (poor time response) and "filter less" implementations (excessive HF modulation).  If Lanczos works with audio as it does with picture images, I think it would closest to restoring the digital signal to its "pre-digitized" state.

So I want to replace the "sinc interpolation" with the "Lanczos3 interpolation."  And I'm pretty sure this is done by modifying the coefficients in the FIR filter.  Either take a customizable "FIR filter" plug-in or change the coefficients in the existing FIR filter in source code with re-compile.
*


I don't think you can't do this in foobar without either writing your own plugin or modifying somebody else's. What you are proposing to do is a lowpass filter operation, which makes no sense from a playback point of view unless the sample rate is increased as well - but you cannot use the built-in resampler, because it will, in effect, do the lowpass operation for you through its own filtering.
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