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#371 |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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It's indeed what I would expect. Kudos for writing this software.
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There's a time for everything, and this is not it. |
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#372 |
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diyAudio Member
Join Date: May 2002
Location: Switzerland
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Hi Bruno
First of all I'd like to wish you a Happy New Year. Second: One transient-perfect opology that wouldn't be too hard to implement is the subtractive-delay one. That would be much easier for the users and the programmer compared to FIR or other pseudo-noncausal methods. Regards Charles |
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#373 |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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Hi Charles, vir jou dieselfde!
So far the subtractive topologies I've seen all trade in phase matching for improved transient behaviour on-axis. I wouldn't know what Psi are doing though, they seem to have figured it out quite well.
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There's a time for everything, and this is not it. |
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#374 |
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diyAudio Member
Join Date: May 2002
Location: Switzerland
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Dankie !
PSI is using conventional crossovers and a classic phase-equaliser (built from 2nd order allpass sections) in front of them. What I meant by subtractive-delay is something like that (second picture of 2nd post): Problem with subtractor block | EngineerZone If you use a higher-order Bessel for the lowpass you will end up with quite good phase-matching between the different ways. But the Highpass wouldn't be very steep. Regards Charles |
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#375 |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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Could the same responses not be approximated using the filter/delay functionality in the existing Hypex software?
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There's a time for everything, and this is not it. |
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#376 |
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diyAudio Member
Join Date: May 2002
Location: Switzerland
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There is just a tiny little bit that is missing: You would need to subtract an IIR lowpass function from a delayed (but otherwise unaltered) version of the input signal. The delay must be set to the same value as the group-delay of the lowpass at zero Hz.
Filtering and delaying alone wouldn't work. Regards Charles |
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#377 |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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That's why I said "approximate"...
__________________
There's a time for everything, and this is not it. |
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#378 |
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diyAudio Member
Join Date: May 2002
Location: Switzerland
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OK, sorry ...
One way to correct such things would be the use of a chain of allpasses but they wouldn't be easy to dimension. I for instance do this with an Excel sheet and some trial and error. But there are of course filter design programs that can do phase-linearisation but they are quite costly, especially if you don't need them too often. Regards Charles |
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#379 |
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diyAudio Member
Join Date: May 2002
Location: Switzerland
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This paper here was presented at an AES meeting and shows a method that is a good compromise regarding transient accuracy and frequency response. Unfortunately it also suffers from some lobing issues but less so than the classic constant voltage subtractive crossovers:
http://www.harschacoustic.ch/Crossov...20response.pdf Regards Charles |
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#380 |
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diyAudio Member
Join Date: Oct 2006
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I also think linear phase is overrated....
We have to see everything in it's perspective. Only a loudspeaker with 1 driver hase a linear phase, if you ad a driver it aint anymore. Phasing will be dependent of listening position and can't be corrected with a filter ( or you should put a sensor on your head attached with your dsp The attached picture is from the earlier mentioned linear phase xover, but demonstrates this phase "problem" clearly ( and does not solve it!). |
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