Optimizing subwoofers and integration with mains: multi sub optimizer - Page 16 - AVS Forum | Home Theater Discussions And Reviews
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post #451 of 777 Old 02-17-2017, 11:45 PM
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@andyc56 Is there a way to exclude filters from the biquad text export? I want to exclude certain filters that I use only for simulation purposes.

Markus

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post #452 of 777 Old 02-18-2017, 07:47 AM
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Originally Posted by markus767 View Post
@andyc56 Is there a way to exclude filters from the biquad text export? I want to exclude certain filters that I use only for simulation purposes.
Not at present. I can do this for a later version.
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post #453 of 777 Old 02-18-2017, 08:57 AM
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Originally Posted by andyc56 View Post
Not at present. I can do this for a later version.
That would be great. Maybe simply an option in the filter properties?

Markus

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post #454 of 777 Old 02-18-2017, 09:29 AM
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Originally Posted by markus767 View Post
That would be great. Maybe simply an option in the filter properties?
I thought of a couple of options for this.

The first would be to have an "exclude from export" property in each filter. To do it this way would, I think, require a special icon for filters having this property, since one might unintentionally set the property and leave the filter out of export when it might actually be needed. One would, at any rate I think, want to get visual feedback as to which filters actually have the property.

The second would be to have an "export selected filters" command. This would cause a dialog box to come up, listing the filters to be exported, their reference designator (FL10, FL23 etc.), their type, and a checkbox next to each one (checked by default) indicating whether or not one wishes to export the filter in question.

I'm leaning toward the second approach, to minimize unintended (and possibly undetected) surprises that could lead to wrong data being exported.
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post #455 of 777 Old 02-18-2017, 01:59 PM
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I agree. Option 2 is probably more foolproof.

Markus

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post #456 of 777 Old 02-19-2017, 04:54 PM
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OK, so MSO seems to do some great stuff. I'm a little confused on some results.

I've got full-range towers that I'm integrating with three subs. Two subs are new, ported, very powerful. The third is pretty old and doesn't extend down as far but is large/has lots of volume. I'm using a MiniDSP 2x4 HD to control the subs and I have a Yamaha AVR plus some external power amps.

First I did a sub-only configuration. I plugged in the delays and gains from the filter report into the MiniDSP and populated the PEQ's via cut and paste. It seemed to work well and the actual measurements look pretty good.

Should I be using the timing delta reported from REW when I do a measurement with acoustic timing reference to adjust the distance in the AVR between the Mains and Subs? When I do that, I get a sub distance that is several times greater than the physical measurement?

Second, I did an MSO with the mains and three subs. The gain, delay, and filter results indicated that Sub #2 isn't being used much.

Final gain and delay/distance settings:
Increase AVR sub out trim gain by 9 dB
Sub 1 gain: -0.27 dB
Sub 2 gain: -17.78 dB
Sub 3 gain: -2.26 dB
Decrease AVR sub out distance by 11.2 feet
Delay settings:
Sub 1 delay: 0.11 msec
Sub 2 delay: 20.05 msec
Sub 3 delay: 6.12 msec

So I ran it again with just Subs 1 & 2 (the new ones). Pretty much the same thing. So, does this indicate that Sub 2 is not in a good physical location to help out with the mains?

Thanks,
Scott
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post #457 of 777 Old 02-20-2017, 08:49 AM
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There is a new version up, v1.20, which adds the relative plotting mode requested by Markus. I still haven't updated the metrics calculations to show the effect of the optimization on the plot-only measurement groups (the ones excluded from optimization), but that will come in the next version.

When switching to relative mode in plots, I've tried to make it so that if you have manual scaling of the y axis, you usually won't have to change the scaling. The MLP is not displayed at 0 dB, but at a level that's convenient for retaining the y-axis scaling. This is done as follows:
  • The target curve, if any, is subtracted from the MLP to obtain a response that's nominally flat
  • The average of the result, in dB, over the optimization frequency range is computed, then rounded to the nearest 1 dB
  • The MLP is displayed as a flat line at this level

Here's an example of an absolute mode plot:



Switching to relative mode, the following plot is obtained without any other adjustments.



This option for relative mode plotting is in the Trace Properties dialog, and only applies to measurement group traces, not other types of traces. If you are using an optimization mode that does not require that an MLP be specified, you must go into the Optimization Options dialog for the configuration in question and specify a "display MLP". Otherwise, those traces will be plotted in absolute mode.

The trace offsets work slightly differently in relative mode. Trace offsets for non-MLP measurement group traces are disabled. When you change the trace offset of the MLP, all non-MLP measurement group traces for that configuration will shift automatically with the MLP offset.

There is also a "remove frequency-independent errors in relative mode" plotting option. This reflects how the optimizer works in "best match of MLP with other listening positions" mode. In this mode, the optimizer actually rejects the mean error over frequency of any non-MLP trace relative to the MLP and optimizes only the standard deviation. This was done after seeing that, in some situations, surprisingly high mean errors were encountered, perhaps because the listening positions were very far apart. These high mean errors were handicapping the optimizer, causing higher ripple in the non-MLP positions than was desirable, and slow convergence. This plotting option reflects how the optimizer "sees" the problem in this mode. Here's an example plot with the mean errors removed.

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post #458 of 777 Old 02-20-2017, 09:10 AM
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Quote:
Originally Posted by Onetrack97 View Post
Should I be using the timing delta reported from REW when I do a measurement with acoustic timing reference to adjust the distance in the AVR between the Mains and Subs? When I do that, I get a sub distance that is several times greater than the physical measurement?
Not necessarily. The MSO documentation talks about this issue, referring to a quote from the REW documentation about the computed delay for subwoofers being inaccurate. It says this:

Quote:
Originally Posted by REW documentation
Note that delay values are not accurate for subwoofer measurements due to the limited bandwidth of the subwoofer response, the delay estimate is based on the location of the peak of the impulse response and subwoofers have a broad peak and a delayed response.
Quote:
Originally Posted by Onetrack97 View Post
Second, I did an MSO with the mains and three subs. The gain, delay, and filter results indicated that Sub #2 isn't being used much.

Final gain and delay/distance settings:
Increase AVR sub out trim gain by 9 dB
Sub 1 gain: -0.27 dB
Sub 2 gain: -17.78 dB
Sub 3 gain: -2.26 dB
Decrease AVR sub out distance by 11.2 feet
Delay settings:
Sub 1 delay: 0.11 msec
Sub 2 delay: 20.05 msec
Sub 3 delay: 6.12 msec

So I ran it again with just Subs 1 & 2 (the new ones). Pretty much the same thing. So, does this indicate that Sub 2 is not in a good physical location to help out with the mains?
It's hard to say. Each problem is different. The practical considerations and common sense need to come from the user's knowledge. If MSO can reduce the aggregate error by a small amount by reducing a sub level by 20 dB, it will do that, even though common sense would dictate that's not a good idea. What I'd suggest is to clone the above configuration, then in the clone, constrain the minimum gain of the Sub 2 gain block to, say, -6 dB. This is done by selecting the gain block, then changing the lower limit of its gain in the Properties dialog. Re-run the optimization, then using Config, Metrics, compare the errors in the original and cloned configurations, as well as the plots. Experimentation is almost always necessary.
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post #459 of 777 Old 02-20-2017, 09:42 AM
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Quote:
Originally Posted by andyc56 View Post
Not necessarily. The MSO documentation talks about this issue, referring to a quote from the REW documentation about the computed delay for subwoofers being inaccurate. It says this:





It's hard to say. Each problem is different. The practical considerations and common sense need to come from the user's knowledge. If MSO can reduce the aggregate error by a small amount by reducing a sub level by 20 dB, it will do that, even though common sense would dictate that's not a good idea. What I'd suggest is to clone the above configuration, then in the clone, constrain the minimum gain of the Sub 2 gain block to, say, -6 dB. This is done by selecting the gain block, then changing the lower limit of its gain in the Properties dialog. Re-run the optimization, then using Config, Metrics, compare the errors in the original and cloned configurations, as well as the plots. Experimentation is almost always necessary.
OK, sounds like a good plan. Thanks

Last edited by Onetrack97; 02-20-2017 at 10:53 AM.
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post #460 of 777 Old 02-20-2017, 11:52 AM
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Originally Posted by andyc56 View Post
There is a new version up, v1.20, which adds the relative plotting mode requested by Markus.
Thanks a lot Andy! Running some optimizations right now.

Markus

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post #461 of 777 Old 02-20-2017, 12:35 PM
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Originally Posted by Onetrack97 View Post
OK, sounds like a good plan. Thanks
So, I changed the minimum allowable gain in both Subs 1 & 2 to -6 dB.

This is the resultant report:
Final gain and delay/distance settings:
Increase AVR sub out trim gain by 10 dB
Sub 1 gain: -1.43 dB
Sub 2 gain: -13.20 dB
Sub 3 gain: -0.46 dB
Decrease AVR sub out distance by 26.2 feet
Delay settings:
Sub 1 delay: 0.18 msec
Sub 2 delay: 4.75 msec
Sub 3 delay: 2.94 msec
So I'm assuming that Sub 2 gain of -13.20 dB is per the parameters because the overall gain of all three subs is to be set at +10 dB? Am I interpreting this correctly?

Still looks like it doesn't want nearly as much contribution from Sub 2 vs. 1 & 3.
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post #462 of 777 Old 02-20-2017, 12:57 PM
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Quote:
Originally Posted by Onetrack97 View Post
So, I changed the minimum allowable gain in both Subs 1 & 2 to -6 dB.

This is the resultant report:
Final gain and delay/distance settings:
Increase AVR sub out trim gain by 10 dB
Sub 1 gain: -1.43 dB
Sub 2 gain: -13.20 dB
Sub 3 gain: -0.46 dB
Decrease AVR sub out distance by 26.2 feet
Delay settings:
Sub 1 delay: 0.18 msec
Sub 2 delay: 4.75 msec
Sub 3 delay: 2.94 msec
So I'm assuming that Sub 2 gain of -13.20 dB is per the parameters because the overall gain of all three subs is to be set at +10 dB? Am I interpreting this correctly?

Still looks like it doesn't want nearly as much contribution from Sub 2 vs. 1 & 3.
It sounds like you have three individual gain blocks, one for each sub? You can better control the relative gains by having one shared gain block and two individual ones. The shared gain block can use the default max of 15 dB and min of -15 dB. For the individual ones, try this:


Sub 1: Max gain 0 dB, min gain -6 dB
Sub 2: Max gain 0 dB, min gain -6 dB
Sub 3: No gain block
Shared gain block: Max gain 15 dB, min gain -15 dB (default)

In the filter report, the shared gain block will be rounded to the nearest 0.5 dB (or whatever step size you defined in the global options), and slight adjustments will be made for Sub 1, Sub 2 and Sub 3.

You might also try "no gain block" for Sub 1 instead of Sub 3 above (that is, 1 shared, 1 each individual on subs 2 and 3).

Note: If you have N subs, the maximum number of total gain blocks you can have on the subs is N. This could be:
1 individual gain block for each sub, or:
1 shared gain block and N-1 individual ones
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post #463 of 777 Old 02-20-2017, 04:23 PM
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Originally Posted by andyc56 View Post
It sounds like you have three individual gain blocks, one for each sub? You can better control the relative gains by having one shared gain block and two individual ones. The shared gain block can use the default max of 15 dB and min of -15 dB. For the individual ones, try this:


Sub 1: Max gain 0 dB, min gain -6 dB
Sub 2: Max gain 0 dB, min gain -6 dB
Sub 3: No gain block
Shared gain block: Max gain 15 dB, min gain -15 dB (default)

In the filter report, the shared gain block will be rounded to the nearest 0.5 dB (or whatever step size you defined in the global options), and slight adjustments will be made for Sub 1, Sub 2 and Sub 3.

You might also try "no gain block" for Sub 1 instead of Sub 3 above (that is, 1 shared, 1 each individual on subs 2 and 3).

Note: If you have N subs, the maximum number of total gain blocks you can have on the subs is N. This could be:
1 individual gain block for each sub, or:
1 shared gain block and N-1 individual ones
OK, I can try that.

I've gone back and rerun all of my measurements now that I know more what I'm doing. I wanted to make sure I had known distances, levels, etc. I also ran a set with subs 2 & 3 swapping locations.

In general, when trying to integrate mains with subs, should I have a shared PEQ, Gain, & Delay, plus your suggestions from above?

Also, when the report says to decrease the AVR sub distance by X feet and that number would be less than zero in the AVR, what do I do?

Last edited by Onetrack97; 02-20-2017 at 04:32 PM.
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post #464 of 777 Old 02-20-2017, 05:01 PM
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Quote:
Originally Posted by Onetrack97 View Post
In general, when trying to integrate mains with subs, should I have a shared PEQ, Gain, & Delay, plus your suggestions from above?
Let's back up a little bit.

Your data showed the need for boosting the shared gain (AVR trim) by about 10 dB. This indicates the sub level was about 10 dB low during measurement. This means that when using all the gain blocks on a per-sub basis without a shared gain block, you need a fairly wide adjustment range for each per-sub gain block. This in turn can lead to wide differences between individual sub levels. If instead you use a shared gain block to do the "heavy lifting" of a large gain adjustment, you can then reduce the adjustment range of the individual sub gain blocks and more easily constrain the relative levels of the subs with respect to one another. The downside is that you're stuck with figuring out which sub should have no gain block in its path (there must be one, as you have a maximum of 3 gain blocks total allowed when you have 3 subs).

There's an analogous case with delays that applies to the non-HD miniDSP 2x4 units that's worth mentioning, even though you're not using one. These earlier units have a maximum delay of 7.5 msec, which is pretty limited. Often, a much larger delay adjustment range is needed. In that case, one would use a shared delay block in MSO with a wide range, including negative delay, implemented physically by the AVR in the guise of its sub distance setting. The AVR would then do the "heavy lifting" of a large overall delay adjustment if needed, while the non-HD 2x4 would make small individual delay adjustments. This setup also has the analogous problem of requiring one to figure out which sub should have no delay block (because there is a maximum of 3 delay blocks when 3 subs are used).

In your case though, the 2x4 HD has boatloads of delay range, so this problem doesn't come up. You can use an individual delay for each sub without a shared delay. I'd recommend allowing each delay to be negative on the minimum side of its range. If any negative delays do show up, they will be lumped into the AVR distance adjustment. A negative shared delay means "increase the sub distance relative to the as-measured condition".

In summary:
1) Use a shared gain block to do the large sub gain adjustment as needed, with two individual sub gain blocks having a smaller adjustment range than the shared one.
2) For delays, use individual delay blocks for each sub, say -15 msec to +15 msec. The filter report will lump all negative delay into a sub distance increase if needed, while making the final per-sub delays positive or zero.

Shared PEQ are usually used when one wishes to get the MLP by itself as flat as possible as a final cleanup step per figure 42 of the tutorial. If you do use shared PEQ, it should be as a final step after using individual PEQ to optimize multiple listening positions. Shared PEQ can't reduce the seat-to-seat variation of frequency response, but individual per-sub PEQ can.
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post #465 of 777 Old 02-20-2017, 05:28 PM
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Quote:
Originally Posted by andyc56 View Post
Let's back up a little bit.

Your data showed the need for boosting the shared gain (AVR trim) by about 10 dB. This indicates the sub level was about 10 dB low during measurement. This means that when using all the gain blocks on a per-sub basis without a shared gain block, you need a fairly wide adjustment range for each per-sub gain block. This in turn can lead to wide differences between individual sub levels. If instead you use a shared gain block to do the "heavy lifting" of a large gain adjustment, you can then reduce the adjustment range of the individual sub gain blocks and more easily constrain the relative levels of the subs with respect to one another. The downside is that you're stuck with figuring out which sub should have no gain block in its path (there must be one, as you have a maximum of 3 gain blocks total allowed when you have 3 subs).

There's an analogous case with delays that applies to the non-HD miniDSP 2x4 units that's worth mentioning, even though you're not using one. These earlier units have a maximum delay of 7.5 msec, which is pretty limited. Often, a much larger delay adjustment range is needed. In that case, one would use a shared delay block in MSO with a wide range, including negative delay, implemented physically by the AVR in the guise of its sub distance setting. The AVR would then do the "heavy lifting" of a large overall delay adjustment if needed, while the non-HD 2x4 would make small individual delay adjustments. This setup also has the analogous problem of requiring one to figure out which sub should have no delay block (because there is a maximum of 3 delay blocks when 3 subs are used).

In your case though, the 2x4 HD has boatloads of delay range, so this problem doesn't come up. You can use an individual delay for each sub without a shared delay. I'd recommend allowing each delay to be negative on the minimum side of its range. If any negative delays do show up, they will be lumped into the AVR distance adjustment. A negative shared delay means "increase the sub distance relative to the as-measured condition".

In summary:
1) Use a shared gain block to do the large sub gain adjustment as needed, with two individual sub gain blocks having a smaller adjustment range than the shared one.
2) For delays, use individual delay blocks for each sub, say -15 msec to +15 msec. The filter report will lump all negative delay into a sub distance increase if needed, while making the final per-sub delays positive or zero.

Shared PEQ are usually used when one wishes to get the MLP by itself as flat as possible as a final cleanup step per figure 42 of the tutorial. If you do use shared PEQ, it should be as a final step after using individual PEQ to optimize multiple listening positions. Shared PEQ can't reduce the seat-to-seat variation of frequency response, but individual per-sub PEQ can.
Thanks for all of the detail. I think I get it now.

I ran MSO on my configuration with subs 2 & 3 in reverse positions and the gains came out very close on subs 2 & 3.

You've got a great program. I can't see doing this without MSO. You make testing variations very easy.
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post #466 of 777 Old 02-27-2017, 03:23 PM
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So, being pretty much done with integrating the three subwoofers with the full-range towers, I still had one input and one output open on the 2x4 HD. I was interested in lowering the crossover point on my center channel without affecting the surround speakers.

So I hooked it up, changed the center to Large from Small and used REW to smooth out the 80 - 150 Hz range and added a HPF at 60 Hz. Looked good.

But, when I went back and retested the Mains + Subs, the results had degraded noticeably. My assumption is that the additional overhead in the DSP caused the main to sub timing to be off. I was not able to resolve with just adjusting the AVR sub distance. I did test a couple of config in the DSP. Just having a crossover and a couple of PEQ's loaded caused the problem, even without any signal on the center channel. So, I pulled it all out and went back to where I started.

Anyone else come across this?

Thanks,
Scott
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post #467 of 777 Old 02-27-2017, 06:46 PM
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Originally Posted by Onetrack97 View Post
So, being pretty much done with integrating the three subwoofers with the full-range towers, I still had one input and one output open on the 2x4 HD. I was interested in lowering the crossover point on my center channel without affecting the surround speakers.

So I hooked it up, changed the center to Large from Small and used REW to smooth out the 80 - 150 Hz range and added a HPF at 60 Hz. Looked good.

But, when I went back and retested the Mains + Subs, the results had degraded noticeably. My assumption is that the additional overhead in the DSP caused the main to sub timing to be off. I was not able to resolve with just adjusting the AVR sub distance. I did test a couple of config in the DSP. Just having a crossover and a couple of PEQ's loaded caused the problem, even without any signal on the center channel. So, I pulled it all out and went back to where I started.

Anyone else come across this?
I'm not completely sure I understand what you've done, but...

If you set your center to large, the bass from the center will not be redirected to the subs. Then, if you high-pass filter the center, you will have lost even more bass. This setup doesn't seem quite right. I have no idea why that arrangement would affect mains + subs though.
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post #468 of 777 Old 02-28-2017, 05:46 AM
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I'm not completely sure I understand what you've done, but...

If you set your center to large, the bass from the center will not be redirected to the subs. Then, if you high-pass filter the center, you will have lost even more bass. This setup doesn't seem quite right. I have no idea why that arrangement would affect mains + subs though.
My normal setup with the Yamaha AVR is L & R set to large, all others set to small. "Extra Bass" set to ON. Yamaha RX-A820 Subwoofer Out only goes to the subs, I don't have it connected to the Tower's LFE inputs. External power amps for Left, Center, Right, and one Passive Sub.

I understand what you are saying about the center, will probably not mess with it.
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post #469 of 777 Old 02-28-2017, 03:03 PM
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Quote:
Originally Posted by Onetrack97 View Post
My normal setup with the Yamaha AVR is L & R set to large, all others set to small. "Extra Bass" set to ON.
Oh, with "extra bass" set to ON, my comments don't apply and it should be okay to put a high-pass on the center. I'm still puzzled as to why doing so would alter the behavior of the integration of L and R main speakers and subs though. If the change were only in the delay of the unit in the sub channel paths, one would think that changing the sub distance in the AVR would fix it. It's a very strange situation.
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post #470 of 777 Old 03-11-2017, 01:48 PM
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Hey Andy, SMWTMS is broadcasting another meeting from Earl Geddes' house, and it turns out that he's actually using your software to do his new multiple subwoofer setup. Assuming you're going to miss it, hopefully they will post a recording later... I'll post a link.

Evidently he didn't like it much and fell back to SpectraPLUS. He also comments he would have never included time delay as a variable factor for the optimization.

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post #471 of 777 Old 03-12-2017, 07:16 PM
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He also comments he would have never included time delay as a variable factor for the optimization.
He told me that in an email too. He said he performed an optimization in MSO that "ran amok" when delays were used, but worked okay when they were not. By design, the solutions MSO comes up with never have a higher error than the ones obtained with the filter parameters manually entered by the user. So the claim of the optimization "running amok" when using delays seems far-fetched. One need only omit delay blocks if they are not desired, so it need not be a concern. There's actual data demonstrating the falsehood of the claim that variable delays can't help.

He has also claimed that time-synchronization of measurements is not only unnecessary but in fact wrong. He is apparently using non-time-synchronized measurements for some of his own optimizations. I was rather taken aback when I found out about that. Floyd Toole's comments regarding the error of not properly taking phase into account when using the superposition theorem apply equally to non-time-synchronized measurements as well. Maybe Earl is using a power summation instead of the superposition theorem?

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post #472 of 777 Old 03-13-2017, 10:25 PM
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There's actual data demonstrating the falsehood of the claim that variable delays can't help.
I thought a bit about this, and I think the main reason behind this is the fact that he effectively doesn't believe in crossovers - he wants to run the same full-range signal to the mains and the subs and let them simply acoustically fall off, thereby gaining the maximum range of overlap which may give the most range in which you can make corrections. That was probably reasonable before when he was just running the speakers without EQ, though he now has EQ capabilities individually on each driver so that provides extra flexibility (and complication). Anyway the time alignment, I believe, is most important for gaining predictable response in the crossover region - and here's the key - it's not useful lower down. I think Earl is thinking of this mostly as a pure LF problem, therefore discounting time alignment to keep things simple, which may be a reasonable approach below the typical crossover region.

I haven't built my subs yet, but I have ported mains so I can't just run them full-range, I guess. Or maybe their crossover has an HP filter built in? Hmm, that might be something to explore. In any case, I've half a mind to use the software to optimize a setup without delay and with, and invite Earl over to explore the difference. In fact it might be instructive to see if he would care to optimize the system his way and compare it to the software's results.
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I thought a bit about this, and I think the main reason behind this is the fact that he effectively doesn't believe in crossovers - he wants to run the same full-range signal to the mains and the subs and let them simply acoustically fall off, thereby gaining the maximum range of overlap which may give the most range in which you can make corrections. That was probably reasonable before when he was just running the speakers without EQ, though he now has EQ capabilities individually on each driver so that provides extra flexibility (and complication). Anyway the time alignment, I believe, is most important for gaining predictable response in the crossover region - and here's the key - it's not useful lower down. I think Earl is thinking of this mostly as a pure LF problem, therefore discounting time alignment to keep things simple, which may be a reasonable approach below the typical crossover region.

I haven't built my subs yet, but I have ported mains so I can't just run them full-range, I guess. Or maybe their crossover has an HP filter built in? Hmm, that might be something to explore. In any case, I've half a mind to use the software to optimize a setup without delay and with, and invite Earl over to explore the difference. In fact it might be instructive to see if he would care to optimize the system his way and compare it to the software's results.
You can run satellites without a HP but this will also reduce their max. output and response consistency within the overlapping region to the subs.
You can run subs without a LP but this will affect imaging and can make them localizable.
The problem with multiple low frequency sources and non-monophonic content is that there's less room for optimization. I've found that I get better (smoother and more consistent) results when minimizing the sat/sub overlap.

Delay is an important factor in optimizing the response of multiple subwoofers. They are virtually never equidistant from the main listening position and even if they are one might get better results by playing with delay. MSO automates that process and let's one define limits so there's no unwanted effects.
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Here's a system without an explicit crossover. Can any honest person say that the electroacoustic crossover frequency is not between 80 Hz and 90 Hz?



And if so, why would such a system be immune from potential improvements of automated calculation of sub distance/delay, as compared to a system with an explicitly implemented crossover?

MSO was originally conceived, in part, to provide functionality that's incompetently implemented in commercial "room correction" systems: the ability to properly integrate main speakers and subs. In fact, the original implementation of MSO didn't have "sub-only" configurations at all, and required the user to enter main speaker data. The problem with traditional room correction systems is their reliance on time domain data to choose the sub distance for integration. In the absence of sheer luck, this technique gives wrong results, and, despite a couple of decades of failure, is still in use today.

It seems peculiar to ask someone to defend their decision to replace incompetent integration procedures with competent ones.
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I've found that I get better (smoother and more consistent) results when minimizing the sat/sub overlap.

Delay is an important factor in optimizing the response of multiple subwoofers.
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And if so, why would such a system be immune from potential improvements of automated calculation of sub distance/delay, as compared to a system with an explicitly implemented crossover?

It seems peculiar to ask someone to defend their decision to replace incompetent integration procedures with competent ones.
Hey, I agree with you guys. Just trying to see the problem from his point of view, partly because I want to figure out whether what we have is a problem of semantics - and partly because I want to figure out how to convince him otherwise.

And I really shouldn't say "he doesn't believe in crossovers" but "he believes that with his speakers not using a crossover between subs/mains is a better approach". Personally I think that ultimately the best approach is highly contextual.
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You want to "convince" Earl? Not going to happen

Well the problem is pretty simple. The more low frequency sources you have, the smoother the response (after some optimization steps). BUT those sources have to be fed a monophonic signal. Overlapping satellites means non-monophonic content. Furthermore low frequency sources become localizable above a certain frequency. Studies (W. L. Martens, J. Braasch, and W. Woszczyk, 2005) have shown that frequencies above 60-80Hz can be localized and crossovers aren't brick walls...
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Optimizing subwoofers and integration with mains: multi sub optimizer

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You want to "convince" Earl? Not going to happen

I was really surprised by one of his comments. Namely, asserting that the MSO optimization "sounded bad." That's a very un-Geddes comment. He didn't mention taking any confirmatory measurements. He'd not tolerate that kind of assertion from another without data.

Also, I wonder how closely he read the MSO documentation. He claimed REW was required, but the MSO manual clearly mentions support for holmimpulse, which he mentions later as a program he uses. Also, I don't use REW but I expect it is dual channel, i.e. it can be used with loopback time correction.
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He claimed REW was required, but the MSO manual clearly mentions support for holmimpulse, which he mentions later as a program he uses. Also, I don't use REW but I expect it is dual channel, i.e. it can be used with loopback time correction.
If one is using a USB mic, REW is required, as its acoustic timing reference is the only way I know of to get time-synchronized measurements with that arrangement. If there is some other software that can do this using a USB mic, I"d be interested in finding out so I can add it to the documentation. If one is using an analog mic and e.g. an external USB sound device, any measurement software supporting loopback time correction (which includes REW, FuzzMeasure and others) can be used. HolmImpulse can do time-synchronized measurements without a physical hardware loopback via its time-locking feature. But according to Bill Waslo (OmniMic author), that only works if the output DAC and the input A/D clocks are locked to one another. This can't happen with a USB mic arrangement. All of this is explained in the MSO documentation.
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There's a site called the "AV Rant Podcast" where people email them questions and they answer them on the podcast. In this one, somebody asked the following question:

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And Nick B. found Dr. Earl Geddes’ approach to multiple subwoofer setup (additional LINK 1 and LINK 2) very interesting, and wonders if it is worth paying his $400 software-based setup fee.
Doing this with MSO would violate the license agreement, so it makes sense that he would do it his own way.
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If one is using a USB mic, REW is required, as its acoustic timing reference is the only way I know of to get time-synchronized measurements with that arrangement.


I didn't see anything where he mentioned using a USB microphone, though I may have missed it. He does mention Holm later.

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