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Old 01-13-2017, 11:40 AM   #20
earlevel
Human being with feelings
 
Join Date: Dec 2015
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Quote:
Originally Posted by Nowhk View Post
1 - You wrote: "For this filter, we’ll have a filter type (lowpass, highpass, bandpass, notch, peak, lowshelf, and highshelf), frequency, Q, and peak gain (used only for the low and high shelf types)." I think a typo, because you will also use peak gain for the Peak filter. Or maybe it was too predicted
Good catch, fixed.

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2 - Does it automatically works if Sample Rate is 48000hz? Or I should change somethings in the filter itself? You are only take care about sample rate when passing the freq (dividing by the sample rate).
Everything is relative (to the sample rate) in filters. I always use "normalized frequency"—1.0 is the sample rate, 0.5 is the Nyquist frequency, 0.0 is DC, etc.

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3 - Q is between 0.1 and 10.0. With a Peak filter I can just write 0.1-10.0 on my plugin Knob label. But when using it with a BandWidth filter, would be more suitable write octave in hz as label. Which conversion do you suggest? Rane conversion would mirror the math of your filter?
I'd suggest looking at other EQ plugins. I think you'll generally find Q in audio, though. Think about it...you have a filter with bandwidth 100 Hz. Is that wide, or narrow? If the center is at 6 kHz, it's narrow. If it's at 120 Hz, it's huge. And if you're sliding the center frequency around, looking to take out a resonance, perhaps, you don't want the sharpness of that filter changing as you move it around.

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4 - If I separately process left/right signal to the filter, I need to create 2 filters instead of one. But since they are filtering with the same parameters, instead of calculate coefficients twice, does it makes sense(after calculating coefficients of pFilterLeft) just take its coefficients values and paste them to pFilterRight? Or I should also reset some buffer?...
I suggest a stereo (or multichannel) filter. The filter would have just one set of coefficients (and obviously one set of methods to set frequency/Q/type), but two (or more) sets of delay pairs. You could have a separate left/right process routine, or a single one that accepts an index (or both). Your interface would treat it as a single filter, but you audio processing routine would view it as two processes.

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5 - What if I also need an All Pass filter? RBJ give one in his DF1 implementation. Maybe you can complete it with this one too
Yes, I omitted partly because I was keeping things simple and not bringing phase into the picture (and plotting it, etc.). Some day.
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