single driver double bass array ( a.k.a. bow your heads ) - Page 2

Borat · 12:54 PM

Quote:

Originally Posted by Don Hills

Your technique is used very day. You're just looking at the problem in the time domain, whereas it is usually looked at in the frequency domain.

i have thought about that. see here:

Q about Signal Processing

my motivation was to provide individual pre-processing for subs in a multi-sub system. the final processing would be by an EQ realizing mirror image of measured combined response at the listening position.

the point was to come up with pre-processing that takes into account the interactions between the multiple subs.

for example if there are 2 subs at the opposite corners the time delay for two of the dimensions is simply reduced by a factor of 1/2 because the distance sound travels was effectively cut in half. with 2 subs all the odd order modes are canceled by the subs themselves and only the even ones remain to be canceled electronically. reducing the time delay by 1/2 automatically shifts the nulls to the new correct frequencies.

if the two subs are not in the opposite corners then odd modes are canceled out for only one dimension and only one dimension gets time delay reduction of 1/2.

now using EQ you could either EQ each sub individually or EQ the sum of all the subs or even first EQ each sub individually and then EQ the sum. but neither of those approaches would be optimal for getting the flattest response over a wide listening area.

since different subs couple to different modes the optimal approach is to EQ individual subs for their individual energization of standing waves and EQ the final sum for for the flat response at listening position.

but how do you EQ an individual sub for energization of standing waves ? measurement at listening position would miss some of the modes as you might be measuring in a null. the mode would then show up in the wider listening area. and of course this measurement wouldn't be taking into account the coupling of the individual sub to the one in the opposite corner !

you could spend your entire life running around with a microphone trying to determine the optimal processing for a system with lets say 5 subwoofers.

with my approach you dial each one in based on simple math and then EQ the final summed response once based on measurement at the listening position.

Don Hills · 9th February 2010, 08:11 PM

As was pointed out to you in the other thread, your approach is the same as applying filters in the frequency domain. It's a bit like working with optics - is light made of waves or particles?

Starting from first principles, let's take a simplest example: A room with a first mode at, say, 40 Hz, and perfect reflection. Now play a 40 Hz note through the speaker. On a piece of paper, draw a few cycles of the sine wave representing the signal being reproduced from the speaker.

Underneath that wave, and in the correct phase and timing, draw the reflected wave as it arrives back at the speaker. (Perfect mode and reflection, so it should be the original wave, delayed one cycle.)

Underneath that, draw the waveform required to cancel that reflected wave. (The same wave, inverted.)

Now add the first and last waves together. The result is the signal that will be applied to the speaker. Do you see the flaw in your argument?

In the real world, the reflected wave will not be the same waveshape or level. The reflection is less than perfect and is reduced in level. A lot of the original signal doesn't come straight back but "sloshes" around the room. A simple delayed and inverted version of the original will not be enough. As you point out, it's a losing game trying to cancel each of these secondary reflections.

If you're having trouble visualising that, some time spent with the ripple tank simulator may clarify it for you. Draw a rectangle, place a source in one corner, and adjust the rectangle size and the source frequency until you get a first mode resonance. (It's surprisingly hard to do, it does a good job of simulating a perfect room.)
http://www.falstad.com/ripple/

A DBA only works at all in a very restricted set of conditions: The room must provide a symmetrical overall acoustic environment, and the driver arrays must produce a sufficiently close to plane waveform at the frequencies of interest. Once you depart from that, such as drivers in corners and other odd positions, it all goes to pieces and you're back to Geddes or Welti's schemes. (The ripple tank simulator can produce plane waves, and multiple sources with adjustable phase between them, so with a bit of work you can simulate a DBA setup.)

Borat · 9th February 2010, 11:01 PM

You pay too much attention to words. Words are only there to get a response

The only thing my approach has in common with DBA is that i am using an inverted and delayed wave.

The wave i am canceling out is not the first reflection, but the SECOND reflection. Read my post above - the PURPOSE of all this is to provide pre-processing for individual subs in a Geddes/Welti type setup.

You seem to be replying to the name of my thread, not to what i have actually written INSIDE of it.

I have never claimed that it will produce, using a single sub the kind of performance that DBA offers - go ahead - try to find a place where i claim that. what i wrote is correct. you're just not reading it right

Lynn Olson · 10th February 2010, 02:31 AM

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