Thanks for explanation. But with delay after hitting "play" at home, wouldn't the sound be just fine if the delay was 3000 milliseconds (sync'ed on all amps, of course)?
How would you program a correction for "group delay"? Or are you hypothesizing an all-knowing DSP?
B.
Art answered that spot on (as usual). 
Here's the original thing that got me thinking about it a fair bit about group delay.Merlijn van Veen - A Track Named Impulse Response
FIR seems like a whole lot of unnecessary processing (a 1kHz sampling rate would be more sensible for subwoofer frequencies than a 48kHz sampling rate, but no one makes hardware like that). The other thought is to do some split band and delay stuff with IIR filters. See the attached screengrab from a Biamp Tesira setup. Something like this... not saying this would work as is. It'd probably need to be more complicated than this. Tesira will show you phase, but it'd be easier to tweak settings for something like this if you could see group delay instead. Note you could get a $50 Biamp Nexia DSP from eBay and do this same sort of stuff with that.
Slightly different topic, but this is also something I've considered doing, in an attempt to compensate for Fletcher-Munson / Robinson-Dadson curves. I don't have a BSS Soundweb, but I have a Biamp Tesira and a Rane Hal. I've done something similar to this with the Rane using AGC blocks with split bands and side chains.
http://www.bsstechsupport.com/public/appguide/creating-a-bass-lift-volume-control.pdf
Here's the original thing that got me thinking about it a fair bit about group delay.Merlijn van Veen - A Track Named Impulse Response
FIR seems like a whole lot of unnecessary processing (a 1kHz sampling rate would be more sensible for subwoofer frequencies than a 48kHz sampling rate, but no one makes hardware like that). The other thought is to do some split band and delay stuff with IIR filters. See the attached screengrab from a Biamp Tesira setup. Something like this... not saying this would work as is. It'd probably need to be more complicated than this. Tesira will show you phase, but it'd be easier to tweak settings for something like this if you could see group delay instead. Note you could get a $50 Biamp Nexia DSP from eBay and do this same sort of stuff with that.
Slightly different topic, but this is also something I've considered doing, in an attempt to compensate for Fletcher-Munson / Robinson-Dadson curves. I don't have a BSS Soundweb, but I have a Biamp Tesira and a Rane Hal. I've done something similar to this with the Rane using AGC blocks with split bands and side chains.
http://www.bsstechsupport.com/public/appguide/creating-a-bass-lift-volume-control.pdf
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It could be a problem if you're hitting "play" on a dvd player rather than a cd player ?
Sure, but not necessarily in all applications (such as processing post-recording). Anyway, I was being too cute saying 3000ms when a realistic figure is guessing 1/200 of that. Otherwise, processing delay - even with slowish CPUs - shouldn't intrude in SQ.
What about my other question about correcting group delay other than sealed box speaker design and motional feedback?
B.
Hey
What I showed in that Tesira screen grab attachment is a concept that should work. In my case, it's a typical ported sub. So there's a group delay peak of 20ish ms at tuning frequency of 25 Hz. I've added a subsonic filter as well that adds more delay (I'm debating if that filter is really necessary with recorded music). I can't speed up the output of the sub at 25 Hz, so the idea is you slow other frequencies down instead. With some DSP devices you can design your own signal path. What I showed there was taking the input signal into a 2-way crossover at say 30 Hz. Then I took the HF output through a delay block. 5ms is shown, but it could be more or less as needed. Then I summed the delayed HF with the (un-delayed) LF. This is over-simplistic for the point of showing the concept, but it effectively "speeds up" frequencies around 30 Hz. It would likely take more of these filters, delays, and mixer blocks cascaded. If I ever get this fully worked out and implemented, I'll share the details. I expect this would take a fair bit of trial and error and measurement.
What I showed in that Tesira screen grab attachment is a concept that should work. In my case, it's a typical ported sub. So there's a group delay peak of 20ish ms at tuning frequency of 25 Hz. I've added a subsonic filter as well that adds more delay (I'm debating if that filter is really necessary with recorded music). I can't speed up the output of the sub at 25 Hz, so the idea is you slow other frequencies down instead. With some DSP devices you can design your own signal path. What I showed there was taking the input signal into a 2-way crossover at say 30 Hz. Then I took the HF output through a delay block. 5ms is shown, but it could be more or less as needed. Then I summed the delayed HF with the (un-delayed) LF. This is over-simplistic for the point of showing the concept, but it effectively "speeds up" frequencies around 30 Hz. It would likely take more of these filters, delays, and mixer blocks cascaded. If I ever get this fully worked out and implemented, I'll share the details. I expect this would take a fair bit of trial and error and measurement.
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Thanks for Rephase link.RobWells;5200828[URL="https://www.minidsp.com/applications/advanced-tools/rephase-fir-tool" said:
Absolutely astonishing square waves! As a youth, I always thought sq-wave reproduction was the sine qua non of audio. Now I realize it is fairly meaningless* and that is because phase, as documented by a mic, is one of the great non-issues in music reproduction.
Looks like "group delay" has to be further separated between speaker mush (which the Rephase program addresses by fixing the crossovering) and driver group delay (a mechanical/acoustic issue) and "net" of at-your-seat sound (which packs driver, speaker, and room) and what I'm curious about fixing.
Interesting to see the dramatic effect of Rephase on pulses (see picture below). While the initial pulse (on left) is sure tightened up (by getting the bass and treble drivers to honk together, at least as far as a mic is concerned), the mush dribbling along after the initial pulse actually looks worse.
B.
*I don't say that just because I have never come anywhere near making square waves with my almost full-range electrostatic panels no matter how hard I try
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Freddi,
Harmonic distortion does make any sound seem louder and "richer", though odd order harmonics can detract from musicality. Musical instrument drivers are often designed to have specific distortion "signatures", and with very low Xmax may only require a few watts to distort.
Have you done an a/b comparison with the driver in a box with the same port area and volume to determine that it is the slots rather than driver causing the harmonic distortion?
Classical organ is one of my main musical diets. Interestingly many instruments vary, I prefer the Willis Organs with a richer low bass, many French classical instruments tend to be somewhat drier in the bass.
I have built systems in the past using smaller drivers (12 & 15") capable of 16Hz in room, the benefits of which, when well executed manifest themselves in greater musical reality. When there in the recording (alas all to rare) to feel real bass pressure loading the room can be revelatory when well done.
Some say that bandwidth to 5Hz is beneficial as well. I have built in my 20s a pair of equalised sealed subs with this kind of bandwidth as an experiment, this produced interesting results some of which not so desirable!!
Poor alignments with 30Hz capability let alone 16Hz result in musical disaster; there are all too many such commercial examples out there!
My next move is to have this extended bandwidth, but at a higher output / performance level.
DM
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