Optimizing subwoofers and integration with mains: multi sub optimizer - Page 18 - AVS Forum | Home Theater Discussions And Reviews
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post #511 of 797 Old 04-15-2017, 10:23 PM
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Quote:
Originally Posted by Svenibaer View Post
As already mentioned in my last post - the pretty good results I’ve archived with my Source Sink Sub /Bass Array of 3 subs in the last year have encouraged me to go for a fourth sub.
The reason for the 4th Sub is to balance out my bass array (membrane surface front vs. backside) in order to further improve its correcting effect for room modes and bass reverberation…
I believe your fundamental assumptions are completely flawed. The idea of a "source/sink" setup has its origins in the Double Bass Array (DBA) concept. In such a system, there is an array of subs on the front wall, usually four in quantity, but maybe more, arranged in such a way that the wavefront from the front subs approximates a plane wave. This setup is duplicated at the back wall, with the back wall subs having inverted polarity and a delay corresponding to the depth of the room, in such a way as to cancel reflections from the approximate plane wave produced by the front sub array.

The expectation that having two subs in the front of the room and two in the back can be made to behave as a DBA is unrealistic. Reversing the polarity of the back subs is a recipe for bad time domain performance, even if the frequency domain can be made to work. If you want a DBA, make a DBA. It cannot be wished into existence.

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post #512 of 797 Old 04-15-2017, 11:42 PM
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Quote:
Originally Posted by artur9 View Post
I am not getting any of the images you included.
I'm sorry. I've corrected the links.
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post #513 of 797 Old 04-16-2017, 09:48 AM
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Quote:
Originally Posted by andyc56 View Post
I believe your fundamental assumptions are completely flawed. The idea of a "source/sink" setup has its origins in the Double Bass Array (DBA) concept. In such a system, there is an array of subs on the front wall, usually four in quantity, but maybe more, arranged in such a way that the wavefront from the front subs approximates a plane wave. This setup is duplicated at the back wall, with the back wall subs having inverted polarity and a delay corresponding to the depth of the room, in such a way as to cancel reflections from the approximate plane wave produced by the front sub array.

The expectation that having two subs in the front of the room and two in the back can be made to behave as a DBA is unrealistic. Reversing the polarity of the back subs is a recipe for bad time domain performance, even if the frequency domain can be made to work. If you want a DBA, make a DBA. It cannot be wished into existence.
That's right, Andy!
Because of a rented flat I can't realize a real SBA/ DBA with Subs in the front and backwall at half room height... but the principle should work with tradeoffs. I already saw same (German http://www.nubert.de/downloads/dba.) reviews with regular subs standing on the floor just in front of the wall.
I know that my room isn't ideal for this - so I have to try what works and what doesn't - And maybe this is a kind of cherry picking. The help of MSO is highly appreciated

I see at least 5 alternatives or approaches
  1. Use the MSO correction 15 - 150 Hz as described as Run 1 and deselect some PEQs
  2. Use the MSO correction 35 - 150 Hz as described as Run 4 (and maybe deselect some PEQs)
  3. Do separate MSO optimizations for Sub1+2 (Front) and Sub 3+4 (Rear) and match/ blend them afterwards concerning gain and delay
  4. Maybe there's mix/ cascading of 2. and 3. possible?
  5. Skip the Source/ Sink principle and let MSO optimize all Subs an delays
  6. ?
Maybe you'll give my post a second look, since I've repaired the image links. Sorry for that.



Kind Regards,
Sven
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post #514 of 797 Old 04-17-2017, 09:50 AM
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Quote:
Originally Posted by Svenibaer View Post
...
But look at the time domain…
...
Your time domain is good, perfectly follows the frequency response.
I'm not getting what is the issue you are worrying about. Also your spectograms have different scales... Your first optimization run is good.

Also you are needlessly restricting the algorithm by limiting the degrees-of-freedom it can use. If your source/receiver positions are fixed, just run a 'regular' optimization - that is nr. 5, and go from there.
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post #515 of 797 Old 04-19-2017, 04:00 AM
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Multi Sub Optimizer - What a great tool

Dear Andy,

about 3 weeks ago I incidently came across the Multi Sub Optimizer, what a thrill. Ever since reading the Geddes Multisub concept and playing around with REW and its EQ functions I knew, this gotta be automized! You did a really good job in picking up that idea, technically implementing it in an optimization algorithm, your documentation is supreme, the software runs flawlessly and, last but not least, this thread proves the great patience you have with us motivated but sometimes under skilled users. Thank you so much.

To pay you back, here is a presentation of my own results followed by some comments on questions that were raised in this thread. I am also preparing a presentation of your software in my german “home” forum ********* (home of the DBA ;-) (edit: web site link german Translation of" *together.de" censored?)


My room has a very irregular shape and 1.5 foot concrete walls cause almost 100 % sound reflection at subwoofer bandwidth -> room mode horror!







Some pics (before Atmos installation) of my home theatre in a concrete wall basement. There are 4 18” closed box subwoofers on a Behringer NU6000DSP, so 2 separate DSP controlled subwoofer channels. Sub1 are the two boxes in the front and sub2 a stack of two boxes in the back corner.







Behind the screen are 3 Mackie HR824 active speakers. (the left subwoofer was moved to the left corner since this picture was taken)


Bad thing of the -3 db @ 37 Hz, they severely exited a 48 and 59 Hz mode, good thing: the have a built in 4th order Butterworth HPF at 80 Hz. Also there is a big hump between 120 and 250 Hz if bothe FL and FR both run a mono signal. Probably the worst thing is the cancellation near the 80 Hz crossover frequency. The subwoofer chassis have a very high inductivity, for sincere action above 80 Hz they need a very powerfill amp.
The following SPL curves are blue front left and rightspeaker with 80Hz highpass in a Denon AVR-4100 (subs muted), red and magenta the two subs with 80Hz low pass (mains muted), black alls speakers with AVR bass management.


so this is the starting situation.

Measurements were taken with REW, asio4all driver, HDMI5 as acoustic timing reference (surround left), a calibrated ECM8000 mic on a Tascam US144 interface. The 5 Positions are shown in the first pic, a drum cymbal stand does a great job to hold the microphone in a reproducible position.
It took me quite a while to learn and improve in programming filters, optimization conditions and transferring the MSO filterreport properly to the DSP systems.


This is what MSO came up with, unoptimzed measurement groups have a -20 db shift for clarity, the top curves are the optimized results with my target curve:


These are the combined filter channels, note the above mentioned HPF on the mains channel.


Considered the hardware limitations to handle the optimum filters, e.g. gain in .5 db steps only, the MSO prediction matches reality quite well. These are the results in measurement positon 1 of two optimization runs mlp from 15 to 120 Hz and from 20.5 to 120 Hz with real measurements of fl+fr+sub1/2. And guess what, it sounds great too


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post #516 of 797 Old 04-19-2017, 06:01 AM
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Some thoughts that came up while working with the program and its documentation and reading all posts in this thread.

You rarely find a single handed made software that after only 2 years of developments is as user friendly as yours. Every function you can imagine is well implemented tooltips everywhere, no bugs at all. You put great effort in providing help and guidance to newbies (easy walk through start) and also describe basic knowledge like the mathematics behind parametric EQ characterization. I really like that.
1. subwoofer only vs sub and mains integration
At first you had a hard time to understand the wish for sub only configurations. In my view the main argument for that hasn’t been mentioned yet: in a multi channel world the pure LFE channel sound needs to be optimized just as the low frequency content of highpass filtered main channels. A pure stereo guy doesn’t bother but with a dedicated LFE channel a compromise needs to be accepted. A crossover above Schroeder frequency wouldn’t be acceptable because of localization of bass sounds. If room modes bellow 80 Hz exited by the main speakers are as dominant as in my case they can be cancelled by the subwoofers. However, a LFE signal played through the subwoofers only would have significant dips at those frequencies. Probably the best thing is to EQ the combined subwoofers and mains each to a perfect roll of of the room response according to the bass management crossover design. I am lucky to have optional 4th order highpass filters built into my studio monitors.
2. know your signals
A system set up to measure room response is complicated and full of bad traps. Am I the only one sitting there so often and wondering why the graph doesn’t match my expectations? Most of the time I later found mistakes like, wrong calibration files in REW, Audyssey still activated in AVR, poor dynamics due to poor microphone gain settings, someone touched a pot on a poweramp and and and…
It really can’t hurt to make loopback measurements of your AVR preouts. What is the filter quality of the bass management of your AVR? Also, do the summation of measurements in REW and compare with a measurement of simultaneously exited spekers. If it doesn’t match your timing reference is wrong. Save a set of reference standard measurements with all DSPs in neutral mode. Everytime you want to start a new project, first reproduce this reference taken months earlier.
3. tricks to avoid too loud measurement sweeps and keep up resolution
I found it helpful do a full range measurement of the subwoofers and add a calculated low pass filter later on. The subwoofers have a strong roll off anyway and adding the AVR low pass results in a) very very loud measurements or b) no usable signal level > 100Hz in Of course one has to be sure of the AVR bass management filter characteristics.
Also I adjusted output signal level individually so that the microphone headroom always was 3-5 db without readjusting the mic preamp gain. Within the REW measurement control panel I then shifted each measurement so that all 5 x 3 measurements had the same -40 db output signal level.
4.Behringer filter compatibility
That is really the only suggestion I can come up with to improve the documentation. It took me quite some head scratching to learn hor to transfer the filter report into my Inuke DSP amplifier. Maybe you want to add this to the compatibility description page:
For parametric EQs use the Q(RBJ) from the filter report, not the Q as shown in the main window or at the start of the filter description in the report.
For LS6 and LS12 filters use the low end corner frequency as shown in the main window or at the start of the filter description in the report. Don’t use LS6 (alt) or LS12 (alt) filters.
For HS filters use the high corner frequency at the end of the filter description in the report, also indicated with “for behringer”

Thanks again and cheers
rumpeli
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post #517 of 797 Old 04-21-2017, 01:07 AM
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There is a presentation and tutorial growing in German. You have to be registered member to see more than thumbnails.
Sven , all, please check it out and let me have your comments forcorrections or improvements.


Thanks

Edit: This is too bad. Neither the site can be mentioned or a link posted, seems to be on some kind of avsforum index. I can't imagine why.

If you google for "der Multi-Sub Optimizer (MSO) ist eine windowsbasierte Freeware" you'll get exactly one hit. If anyone is interested I can post it here or send a Wordfile.

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post #518 of 797 Old 04-21-2017, 10:14 AM
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I discovered a minor bug.


set up a config with one main and two Subs
rename mains and Sub channel 1 but not Sub channel 2
add measuerments for 5 positions to each channel
create measurement Groups for 5 positions and assingn three measurements to each Group
rename Sub channel 2 to "sub2"
create a graph with SPL of thw measurement groups


the measurements in Groups 1 to 5, that are assigne to filter channel sub2 (formrly known as "Sub channel 2") are corrupt, lost. In a graph the measurement Groups can be selected but don't show as a curve.
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post #519 of 797 Old 04-21-2017, 12:22 PM
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Hi rumpeli,

Thanks for your posts and your kind words. That's a nice theater you have!

I didn't realize that the Behringer iNuke used the RBJ definition of Q. John Mulcahy has done some measurements of the Behringer DCX2496, and his results indicate that it uses the "MSO Q", rather than the RBJ definition. I had hoped that their definition of Q across products would be consistent, but it looks like that's not true. I'll fix the documentation to tell the user the correct Q value to use. Thanks for finding this information. It's very valuable and solves yet another mystery. I'll also clarify the usage of the shelving filter.

For the bug with renaming channels, do you have a project that demonstrates the bug you could upload in zip format? If the file is too big, maybe you could upload it to google drive or some other sharing service? Otherwise, I can try to reproduce the error with an existing project.

As a workaround, I think the bug can be avoided by not adding any measurement groups until you're done renaming channels. The program has a messy way of handling renaming of various types of objects (measurements, filter channels, configurations, etc.) These changes often spread out to many other objects that have dependencies on the name in question.
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post #520 of 797 Old 04-21-2017, 02:12 PM
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Quote:
Originally Posted by andyc56 View Post
I didn't realize that the Behringer iNuke used the RBJ definition of Q. John Mulcahy has done some measurements of the Behringer DCX2496, and his results indicate that it uses the "MSO Q", rather than the RBJ definition. I had hoped that their definition of Q across products would be consistent, but it looks like that's not true. I'll fix the documentation to tell the user the correct Q value to use. Thanks for finding this information. It's very valuable and solves yet another mystery. I'll also clarify the usage of the shelving filter.

For the bug with renaming channels, do you have a project that demonstrates the bug you could upload in zip format? If the file is too big, maybe you could upload it to google drive or some other sharing service? Otherwise, I can try to reproduce the error with an existing project.

Let me prepare and post a testfile to check the filter Quality handling. I wouldn't have mentioned the renaming Thing if I wasnt so amazed about how unbuggy and user friendly your program behaves. Let me see if I can post a demo file for that one too.
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post #521 of 797 Old 04-22-2017, 08:33 AM
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I did loopback measurements from the NU6000DSP Output with several filters activated. The measurements were taken with a
  • Tascam US144 Sound interface, analog Output to
  • Behringer NU6000DSP input , Behringer Input attenuation pot at 50% ,
  • poweramp output over a "FH-103 High to Low Impedance Converter" (device to attenuate the voltage) to the
  • analog Input of the Tascam Sound Interface.
In REW I made a soundcard calibration that includes the impedance converter, which has a significant low frequency roll off.

The neat measurement with no DSP filter has a noticable low frequency roll off, I am not exactly sure if any of this results from the measurement chain but I did my best on the calibration file.

In the Pictures you see which filters were activated in the Behringer DSP and how they measured in REW. The attached zip is a MSO config to simulate those filters on the neat measurement.
So please, can someone with more knowledge check which filter Quality definitions Behringer uses in their Inuke DSPs?

Thanks
rumpeli
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post #522 of 797 Old 04-22-2017, 05:24 PM
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Thanks for that.

I just noticed that MSO has no first-order HPF or LPF. I'm going to add those filter types so I can try to match the low-end roll-off of the iNuke amp as closely as possible with a first-order HPF. Once that's done, it should make figuring out the Q a little more accurate. It will take me a couple of days to get everything set up, as there's cool weather here and I'm using that to catch up on yard work.

Did you record what the Q1 and Q2 settings were in the iNuke? I don't need to know beforehand, but will eventually.
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post #523 of 797 Old 04-23-2017, 12:33 AM
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Quote:
Originally Posted by andyc56 View Post
Did you record what the Q1 and Q2 settings were in the iNuke? .

Q1 : Q = 1
Q2: Q = 2

Have a look at the screenshot of the Behringer DSP contol software. It shows all filters on Channel A. Filter #3 is active (botton highlighted inn orange) all others are bypassed.The graph shows the frequency response of the amp output signal with the activated filters. The bottom line has drop down selection fields for the filter type. One line above you can select the quality of PEQs.
You can see all 6 filter settings that I recorded the frequency response for. individually.

It would be good if you could check and confirm, if the Behringer Q definition indeed is the Q (RBJ) as listed in the MSO filter report.

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post #524 of 797 Old 04-23-2017, 08:32 AM
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Quote:
Originally Posted by rumpeli View Post
[SIZE=3][FONT=Arial][COLOR=black]I did loopback measurements from the NU6000DSP Output with several filters activated... The neat measurement with no DSP filter has a noticable low frequency roll off, I am not exactly sure if any of this results from the measurement chain but I did my best on the calibration file.
If you are interested a member documented a method to extend the low frequency response on the iNukes by adding several capacitors. I did it on the 6000 that I use for my Crowsons and it cost me $10 in parts and took about 30 mins. You can replace two caps and extend it somewhere around 10Hz or add 6 caps and get it down to 2-3Hz or so. I have photos of my modifications in the last 1-2 pages of my amp build thread, along with a link to the original thread that has a long how-to showing his signal measurements.

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post #525 of 797 Old 04-23-2017, 10:14 AM
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Originally Posted by FriscoDTM View Post
If you are interested a member documented a method to extend the low frequency response on the iNukes by adding several capacitors. I did it on the 6000 that I use for my Crowsons and it cost me $10 in parts and took about 30 mins. You can replace two caps and extend it somewhere around 10Hz or add 6 caps and get it down to 2-3Hz or so. I have photos of my modifications in the last 1-2 pages of my amp build thread, along with a link to the original thread that has a long how-to showing his signal measurements.
yeah, I found that too as I was searching for options to add pre outs to the amp. However, I see no reason to do this modification as the rolloff can be compensated with the dsp filters.
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post #526 of 797 Old 04-23-2017, 12:27 PM
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Quote:
Originally Posted by rumpeli View Post
It would be good if you could check and confirm, if the Behringer Q definition indeed is the Q (RBJ) as listed in the MSO filter report.
Okay, so I did some tests without a first-order HPF to simulate the roll-off of the iNuke. These were the steps:
  1. Made four sub channels, each with one PEQ
  2. Each PEQ has a 100 Hz center frequency. Two have a 5 dB peak and the other two have a 5 dB cut
  3. Make four plots. Each plot contains the iNuke measurement and the response of the corresponding sub channel
  4. Adjust the trace offset of the PEQ response so its peak has the exact same value as the peak of the iNuke measurement
  5. Adjust the Q of the PEQ using the MSO tuning feature to match the traces as closely as possible
  6. Observe the MSO Q (Qmso) from the Properties window, compare it to the RBJ Q (Qrbj) from the filter report and the Q specified in the iNuke user interface

There are four plots. I'll just show one sample.



This is why I wanted to use a first-order HPF, in order to get a good match at low frequencies. Since this is just a "one or the other" determination, I thought I'd try it this way. I matched the data from a little below the PEQ peak frequency to the top end of the frequency band, to avoid the low-frequency errors. Here are the end results:

Q1, 5 dB boost
Qmso = 1.300 (from Properties window)
Qrbj = 0.974862 (from Filter Report window)

Q1, 5 dB cut
Qmso = 1.390
Qrbj = 1.04235

Q2, 5 dB boost
Qmso = 2.620
Qrbj = 1.96472

Q2, 5 dB cut
Qmso = 2.700
Qrbj = 2.02471

So it looks like the iNukes use the RBJ convention for Q, just as you said. I've attached the project below. BTW, the .msow file tells MSO which tabbed windows should be opened up when opening the project.
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post #527 of 797 Old 04-23-2017, 07:05 PM
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For LS6 and LS12 filters use the low end corner frequency as shown in the main window or at the start of the filter description in the report. Don’t use LS6 (alt) or LS12 (alt) filters.
I just thought I'd mention a bit about the LF Shelf (Alt) filters. These were inspired by the Harman study on target curves. In that study, participants kept the "half-boost frequency" constant at 105 Hz and varied the LF boost to find the most preferred setting. If you had an LF shelf with a boost of, say, 6 dB, the "half-boost frequency" is the frequency at which the boost is 3 dB. The "Alt" variants of the shelving filters specify a center frequency, which is the same as the half-boost frequency. This allows you to vary the LF boost while keeping the half-boost frequency constant, which is useful for specifying target curves. For the normal LF Shelf filter, you specify the LF corner and boost. But if you keep the LF corner constant and vary the boost, the half-boost frequency varies, making them a PITA for specifying target curves.

You can use the "Alt" variants of the LF shelving filter just fine with the iNuke. Just go into the filter report, and it tells you the LF corner value. This is the number to enter into the iNuke software. To illustrate this, I've created a project having two LF shelving filters. One is an "Alt" LF shelf with a center frequency of 105 Hz and a boost of 6 dB. The other is a conventional LF shelf with a boost of 6 dB and an LF corner of 88.3465 Hz, obtained from the filter report of the "Alt" filter. They are the same filter, specified in two different ways, as can be seen in the graph.
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post #528 of 797 Old 04-25-2017, 05:35 AM
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Quote:
Originally Posted by andyc56 View Post

Q2, 5 dB boost
Qmso = 2.620
Qrbj = 1.96472

So it looks like the iNukes use the RBJ convention for Q, just as you said. I've attached the project below. .
First of all, it's easier to work with my measurements if you add -77.1 db gain to have a zero reference level. I should have done that in REW already.

I tried to mimic the Behringer high pass roll off in MSO by importing a txt file with all zero SPL and Phase as "flat measurement", creating a new sub channel with "flatt" added and applying second order highpassfilters to that one. Taking into account the min freqeuncy limit of 10 Hz I had the best result with a HPF variable Q 12 db/Oct at 11 Hz and Q = 0.73.

I also created a filter channel with the Q = 2, 5 db boost measurement and applied a PEQ at 100 Hz, 5 db cut , Q = 2.67 so that the filtered measurement matched the Behringer measurement with als DSP filters bypassed. The filter report states Q (RBJ) = 2.0022

?Unfortunately, at times I sill have trouble to transfer the filters as calculated by MSO into the Inuke DSP and I don't yet know why. Not always but sometimes, the small SPL graph in the DSP software user interface looks a little different from the filter channel gragh in MSO and more severely the check room response measurement really sucks. ?
Edit: maybe I found a relly stupid mistake I made, let me check...

cheers
rumpeli

Last edited by rumpeli; 04-25-2017 at 07:51 AM.
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post #529 of 797 Old 04-25-2017, 06:19 PM
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I believe your fundamental assumptions are completely flawed. The idea of a "source/sink" setup has its origins in the Double Bass Array (DBA) concept. In such a system, there is an array of subs on the front wall, usually four in quantity, but maybe more, arranged in such a way that the wavefront from the front subs approximates a plane wave. This setup is duplicated at the back wall, with the back wall subs having inverted polarity and a delay corresponding to the depth of the room, in such a way as to cancel reflections from the approximate plane wave produced by the front sub array.

The expectation that having two subs in the front of the room and two in the back can be made to behave as a DBA is unrealistic. Reversing the polarity of the back subs is a recipe for bad time domain performance, even if the frequency domain can be made to work. If you want a DBA, make a DBA. It cannot be wished into existence.
Hi Andy, curious as to what you mean by "bad time domain performance"?

The source / sink method can be achieved with as few as one sub front and one sub rear. I've done it many times. Of course may not be as good as having more subs available but in my experience is one of the best methods of achieving good bass in a small room if you only have two subs available.

You can scale from there to two subs front, two rear (at floor level), the next step up would be two subs front, two rear at mid-height, then four front, four rear.

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post #530 of 797 Old 04-25-2017, 06:47 PM
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Hi Andy, curious as to what you mean by "bad time domain performance"?
Non-minimum-phase behavior, for which flat magnitude response does not guarantee that time-domain performance will also be optimized.
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post #531 of 797 Old 04-25-2017, 07:21 PM
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Non-minimum-phase behavior, for which flat magnitude response does not guarantee that time-domain performance will also be optimized.
Audible consequences of non-minimum phase behavior? Minimum phase would normally be associated with a narrow and sharp dip in the response which we generally hear through. Narrow dips are surely more preferable to room mode resonances and a non-flat response? The simple fact of the matter is that the source/sink method, even if not involving four (or eight) subs, generally gives you extremely consistent seat-to-seat response in a multi-row, multi-seat home theater, and does not require any aggressive, high Q EQ (which generates it's own phase shifts) in the region in which the source/sink array is working except for a shelving or low pass filter.

Have you experimented much with the source / sink approach?

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post #532 of 797 Old 04-25-2017, 07:41 PM
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Audible consequences of non-minimum phase behavior?
The original question was about measured data, not audible consequences.

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Minimum phase would normally be associated with a narrow and sharp dip in the response which we generally hear through.
I think you meant "non-minumum phase" above. Mathematically speaking, non-minimum phase in an analog system means the transfer function has zeros in the right-half plane (e.g. an all-pass filter). This concept was originally introduced by Bode in his 1945 book Network Analysis and Feedback Amplifier Design.

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The simple fact of the matter is that the source/sink method, even if not involving four (or eight) subs, generally gives you extremely consistent seat-to-seat response in a multi-row, multi-seat home theater, and does not require any aggressive, high Q EQ (which generates it's own phase shifts) in the region in which the source/sink array is working except for a shelving or low pass filter.
I normally like to distinguish fact from claims of fact.
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Optimizing subwoofers and integration with mains: multi sub optimizer

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The original question was about measured data, not audible consequences.







I think you meant "non-minumum phase" above. Mathematically speaking, non-minimum phase in an analog system means the transfer function has zeros in the right-half plane (e.g. an all-pass filter). This concept was originally introduced by Bode in his 1945 book Network Analysis and Feedback Amplifier Design.







I normally like to distinguish fact from claims of fact.


Sorry yes I meant non-minimum phase.

I'm not sure what you mean "claims of fact", I have plenty of data on source / sink from BEM simulations plus actual in room measurements. For sure some of those rooms have areas of the response with sharp dips and some non-minimum phase behavior (but critically only for some, not all seats), but source / sink (or it's more developed forms CABS etc) gives the best response for the room in terms of flatness of response and seat-to-seat consistency. Better than any other documented approach to multi sub optimization in a rectangular space with consistency of wall/ceiling construction.

I think you dismiss source / sink too easily. You should try it out even in its two sub form. It actually works very well, but it's quite hard to dial in the delay and level of the rear sub without multi-position measurements, so isn't for everyone.


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The original question was about measured data, not audible consequences.







I think you meant "non-minumum phase" above. Mathematically speaking, non-minimum phase in an analog system means the transfer function has zeros in the right-half plane (e.g. an all-pass filter). This concept was originally introduced by Bode in his 1945 book Network Analysis and Feedback Amplifier Design.







I normally like to distinguish fact from claims of fact.


Have you read the various published papers on source / sink incl those outlining two sub approaches?


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post #535 of 797 Old 04-25-2017, 08:24 PM
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I'm not sure what you mean "claims of fact"
I define this as someone saying "X is true" / "X is a fact" without providing supporting evidence.

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I have plenty of data on source / sink from BEM simulations plus actual in room measurements.
I'd be interested in seeing the data. (Really, I'm not trying to be a wiseguy here.)

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I think you dismiss source / sink too easily. You should try it out even in its two sub form. It actually works very well, but it's quite hard to dial in the delay and level of the rear sub without multi-position measurements, so isn't for everyone.
Maybe so. I'll give it a try when I get the chance (which will be in a few weeks).
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Have you read the various published papers on source / sink incl those outlining two sub approaches?
I have read the DBA papers, but it's been at least a year, maybe more.

It's worth revisiting the classic case of a plane wave tube in acoustics. But the discussion gets long, so I'll wait until tomorrow evening to discuss it further.
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post #537 of 797 Old 04-25-2017, 08:53 PM
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I define this as someone saying "X is true" / "X is a fact" without providing supporting evidence.
Funny you say that, because that's what it came across as to me when you basically told the poster that source / sink was terrible and he should avoid like the plague (or that's how it came across to me anyway). Hence my interjection in your thread.

PM me your email addy and I can send you over / put on dropbox some stuff to look at. Looking forward to your thoughts; always good to have a knowledgeable eye take a look at something.

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post #538 of 797 Old 04-25-2017, 10:53 PM
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PM me your email addy and I can send you over / put on dropbox some stuff to look at.
I don't do PMs. If you have data, please post it. That's the whole point of public discussion.
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post #539 of 797 Old 04-25-2017, 11:31 PM
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PM me your email addy and I can send you over / put on dropbox some stuff to look at. Looking forward to your thoughts; always good to have a knowledgeable eye take a look at something.
Nyal, please post your data here so we all can benefit. Maybe we could even run some tests what solution MSO would have found in comparison.

Markus

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post #540 of 797 Old 04-26-2017, 09:56 AM
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Markus and Andy, you should read this paper if you haven't already: Fazenda et al., "Subjective Preference of Modal Control Methods in Listening Rooms", J. of the Audio Engineering Society 2012, S.338 http://www.aes.org/e-lib/browse.cfm?elib=16324. They did listening tests to evaluate the preference of different subwoofer placement and integration strategies, and source/sink (with a two sub setup) came out top.

Unfortunately I can say now that it's unlikely I'll have enough time to give you all the information I'm sure you will ask for. I have been rarely posting on the forums or writing my blog due to work and life being so busy.

The key benefit of source / sink over other approaches in rectangular rooms of consistent construction is independence from length mode related frequency response variations. Essentially with source / sink you don't have any length modes in the frequency range the sub array is operating. In a multi-row home theater (the main thing I am interested in, since these comprise the majority of the projects we do) this is extremely valuable, as all the other methods for integrating the rear sub(s) will result in some length mode variation. For example with zero delay on the rear sub(s) per Welti you will cancel odd order modes (1,3) but accentuating even order (2,4) modes. If you add some time delay to the rear sub(s) such as 2, 4 or 8ms or some other number then you can shift where the peaks and nips are (nodes and antinodes), and hopefully arrange them in such a way that there is more consistency in the response across each row. In larger rooms, where you have flexibility with row placement, then these latter two approaches can work well. However I've rarely found in practice that there is any flexibility with where the rows go. They go where they go because of other design elements: row-to-row distance requirements for reclining theater seats; distance from back row to wall behind (to get distance from ears to rear speakers); distance from front row to screen; depth of screen/baffle wall.


What you do have with a two sub setup (one front / one rear) is height and width mode variation, as you would expect. So you have height modes and you have width modes. If you have only one listener at one height then the two sub setup can actually work very well, as the Fazenda paper shows. I have data too I can post. Of course you would expect the response at seats which are along the width axis of the room to have different response, particularly if they the location of those seats starts to approach the null points for the 2nd width mode, because there is no width mode control in a two sub front / rear setup with the subs at midwall. If the room is wide enough, and the other seats on the row do not get near the null points for the 2nd width, then you can actually have very good consistency across the row. With a two sub setup (one front / one rear) you can of course move the front and/or rear sub away from the centerline of the room. This will result in some width mode cancellation but will make the response variation between seats on the same row worse (as you would expect for any non-symmetrical sub setup).

With four subs (two front / two back) at floor level the width mode variation is reduced by cancellation of the odd order modes. You can either place them in the corners or at the 25/75% of room width locations, with the expected typical differences between those two setups.

If you then take those four subs and move them to half height in the room you reduce interaction with problematic 1st axial height mode (unless your room is very tall higher order modes are not generally an issue). This approach is normally called CABS in the literature, though of course there seems to be some differences in use of the terminology, with one of the key researchers calling an eight sub setup as per the below CABS as well.

If you have eight subs, four front / four rear, then you have a DBA, and you get height mode cancellation as well.
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