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Old 20th July 2007, 06:11 PM   #1
mazurek is offline mazurek  United States
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Default digital compensation to make loudspeaker transient perfect

I was wondering if anyone around these parts knows anything about how to make inverse allpass compensation filters on the computer.

I was thinking about the harsh driver requirements necessary to minimize group delay effects for high spl systems, my current loudspeakers are three ways using LR24 acoustic slopes. Lower slopes to minimize group delay will push the drivers out of their preferred bandwidth.

A phaseless DSP crossover solution is beyond my means, but my analog LR24 crossovers are cascaded 2nd order allpass. It seems to me that one should be able to take an ordinary well designed loudspeaker, and apply digital correction for the 2nd order allpasses, and result in a transient perfect loudspeaker.

I have a Emu 1616M, I'm not sure what it entails to program digital filters for the thing. I have a copy of Matlab that I can use to generate filter coefficients, but I'm not quite sure what I am doing.

Anyone have any information on the subject?

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Old 20th July 2007, 06:26 PM   #2
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Old 20th July 2007, 07:05 PM   #3
mazurek is offline mazurek  United States
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Thanks! Looks like exactly what I want.
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