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[mattkhan]

That's correct, assuming that you're still doing your bass management downstream in convolution.  The settings you've chosen will result in 5.1 remaining 5.1, and everything else will get mixed to 5.1 or 5.0 (including stereo audio).  My understanding is that if there's no LFE in the source, JRSS will not create one with those settings, but if there is an LFE channel (e.g. 7.1 source) it will retain it in the mix.

If you don't want stereo audio to get mixed to 5.0 (or 5.1), make sure the box labelled "for stereo sources only mix to 2.1" is checked. That will, with your other settings, result in stereo sources remaining stereo (because you have subwoofer set to silent).

OK thanks. I find that 2.0 upmixed to 5.0 just sounds weird (quite thin & unnatural) so I will keep that "mix to 2.1" option checked.

FWIW, and going back a few posts to the discussion about whether to correct the near field response of the sub first, I got round to measuring this today. The measurements are of 2 sweeps with the correction filter applied, the red line is the end result of acourate correction with a linearised sub (i.e. in my case, the "inductance hump" removed) and the green line is without that and just correcting the room response alone. There are minor differences in the precise shape of the correction filter and the step responses are almost identical, you could argue the linearised sub is *marginally* "quicker" but whether that is audible is beyond me. I think this shows that, for me, it's not worth the effort involved in the extra measurements as it looks like the need to correct the room response wins (i.e. the final correction filter is barely any different as the room rings more than the sub does).

For completeness it would be interesting to embed that near field response change within the sub and then repeat the room correction and compare the outcome. However I don't get paid for this, it's just idle curiosity

[BradC]

Hi

I think what I said earlier was that the driver linearisation can correct problems (due to the driver) all around the room.

So to compare, you would want to see which approach gives a better result in multiple locations.

Of course, then deciding which is actually better is another thing.

[mwillems]

Hi

I think what I said earlier was that the driver linearisation can correct problems (due to the driver) all around the room.

So to compare, you would want to see which approach gives a better result in multiple locations.

Of course, then deciding which is actually better is another thing.

Exactly; that's the main reason to deal with them separately (that and it's easier to get reproducible results). 

But it's all relative; as the wavelengths get exceedingly large compared to the room, localization becomes less and less relevant.   A quarter wavelength at 20 Hz is 13 feet; unless you have a very large room or very widely scattered listening positions, correction in one spot is probably about as good as any other relatively nearby spot at 20 Hz.  Most domestic listening spaces aren't even large enough to have room modes at 20Hz.

That's much less true at 80Hz where the quarter wavelength is around 3.5 feet.  Then you're going to get much more variation.  And (IMO) above about 100Hz, correcting the speaker separately from the room and/or multi-seat correction becomes more important for good results if you have more than one listening position. 

But this is a long way to saying that I'm not necessarily surprised that it doesn't seem to make a huge difference which way you come at it with deep bass because positioning becomes less important at very low frequencies.

I don't have tons of hands on experience with Acourate and other similar suites; they may have some secret sauce that let's them distinguish between the speaker and the room from a single measurement (some folks have more or less said as much about audiolense).  But the physics of rooms dictate that what's good for one spot can't really be just as good for another, so that has to be addressed somehow (at least at frequencies with wavelengths short enough to have room modes in your room). 

[mattkhan]

Hi

I think what I said earlier was that the driver linearisation can correct problems (due to the driver) all around the room.

So to compare, you would want to see which approach gives a better result in multiple locations.

Of course, then deciding which is actually better is another thing.

I don't see how that applies in the modal region, for higher frequencies yes but I don't see how it could low down. Besides which there is essentially no measurable difference in the near field response anyway.

[mwillems]

I don't see how that applies in the modal region, for higher frequencies yes but I don't see how it could low down.

It definitely doesn't apply "below" the modal region (in the bass region below which your room has no modes), but it potentially has application in the modal region.  A modal resonance is like any other resonance it can be excited to varying degrees depending on SPL.  An anti-mode on the other hand, can often just be an unfillable suck out due to cancellation.  So you can still address the mode maxima by trimming speaker FR peaks to avoid over-excitation of resonant modes.  In those contexts, room positioning can be relevant.

An example: imagine three horizontally arranged seats in a large room.  Modes are periodic, so let's say the room has a 60Hz antimode in the center and resonant modes equidistant to the left and right.  So lets say the center seat is smack in the middle of an anti-mode at 60Hz, and the left and right seats are in the middle of the 60 Hz resonant modes.  Let's also stipulate that the subwoofer itself has a 4dB FR peak at 60 Hz.

If one takes a single correction measurement at the center seat, unsophisticated software would see a huge suckout at 60Hz and try to correct it by boosting (which is exactly wrong).  More sophisticated software would recognize (possibly based on phase/group delay info) that the suckout was a room effect and would leave the FR alone at 60Hz.  Meanwhile, even in the "no correction" hypothetical folks in the left and right seat are getting blasted out of their seats anytime a 60Hz note is played because it's not just 4dB louder from the speaker, it's probably between 7 and 10dB louder (depending on how serious the mode is).  

In our example, if one (instead) did speaker correction ironing out the speaker's 4dB peak (or took multiple measurements and corrected based on that), the left and right seats would still hear a too loud 60Hz, but it would be significantly less "too loud."  The center seat would likely see minimal differences because it would still have a gigantic 60Hz suck out (that may or may not actually be deeper given how room cancellations can be at step one).  Additional seats not in a 60Hz mode or anti-mode would experience better response too.

Besides which there is essentially no measurable difference in the near field response anyway.

And that's the real answer  .  Theory is nice, but you took measurements at different locations and found it didn't make a difference in the relevant frequency band in your room, which is all you really need to know.

[Above edited to hopefully make it clearer what my point was]

[mattkhan]

It definitely doesn't apply "below" the modal region (in the bass region below which your room has no modes), but it potentially has application in the modal region.  A modal resonance is like any other resonance it can be excited to varying degrees depending on SPL.  An anti-mode on the other hand, can often just be an unfillable suck out due to cancellation.  So you can still address the mode maxima by trimming speaker FR peaks to avoid over-excitation of resonant modes.  In those contexts, room positioning can be relevant.

An example: imagine three horizontally arranged seats in a medium large room.  Modes are periodic, so let's say the room has a 60Hz antimode in the center and resonant modes equidistant to the left and right.  So lets say the center seat is smack in the middle of an anti-mode at 60Hz, and the left and right seats are in the middle of the 60 Hz resonant modes.  Let's also stipulate that the subwoofer has a 4dB FR peak at 60 Hz.

If one takes a single correction measurement at the center seat, unsophisticated software would see a huge suckout at 60Hz and try to correct it by boosting (which is exactly wrong).  More sophisticated software would recognize (possibly based on phase/group delay info) that the suckout was a room effect and would leave the FR alone at 60Hz.  Meanwhile, even in the "no correction" hypothetical folks in the left and right seat are getting blasted out of their seats anytime a 60Hz note is played because it's not just 4dB louder from the speaker, it's probably between 7 and 10dB louder (depending on how serious the mode is). 

In our example, if one (instead) corrected the speaker's 4dB peak (or took multiple measurements and corrected based on that), the left and right seats would still hear a too loud 60Hz, but it would be significantly less "too loud."  The center seat would likely see minimal differences because it would still have a gigantic 60Hz suck out (that may or may not actually be deeper given how room cancellations can be at step one).  Additional seats not in a 60Hz mode or anti-mode would experience better response too.

I don't disagree but we seem to be talking at cross purposes. I read what you've written as a description of room correction not speaker correction whereas I'm saying I'm not seeing an obvious case for speaker correction in the modal region. I am not arguing for no correction at all in that region.

In the room correction case, we're specifically moving the speaker (sub) away from a nice theoretically on target response in order to not excite those modal frequencies so much so that we experience an on target response at the listening position.
In contrast in the speaker correction case, we are trying to get the speaker itself to produce the on target frequency response irrespective of what the room might do to it.

[mwillems]

I don't disagree but we seem to be talking at cross purposes. I read what you've written as a description of room correction not speaker correction whereas I'm saying I'm not seeing an obvious case for speaker correction in the modal region. I am not arguing for no correction at all in that region.

In the room correction case, we're specifically moving the speaker (sub) away from a nice theoretically on target response in order to not excite those modal frequencies so much so that we experience an on target response at the listening position.
In contrast in the speaker correction case, we are trying to get the speaker itself to produce the on target frequency response irrespective of what the room might do to it.

My example (I had hoped) was trying to make the case for speaker correction in the modal region (but I guess I didn't do a very good job  ). The point of my example was that an attempt to do room correction based on a single listening position measurement could result in objectively worse results than speaker correction would have (i.e. just ironing out the 4dB lump in the speakers response).  That is to say, if, in my example, one did room correction based solely on the center seat, the center would be no better off at 60Hz, and the two satellite seats would be much worse off.  If you did speaker correction instead, the center seat would be no worse off and the two satellite seats would be much better off.  So in that case, speaker correction would "win" (at least at 60Hz).

But the point of speaker measurement and correction is not to produce an on target frequency response and then stop completely; it's to provide a stable first step in total correction (speaker and room) by disentangling which phenomena can be corrected and which can't.  You can theoretically achieve a similar effect in the modal region by measuring in multiple listening positions, but it can be hard to sort out what's causing the issues.  For example, just measuring out in the room how do you know what's a dip in speaker FR (correctable) and what's a dip from a web of anti-modes (uncorrectable)?  Sophisticated software might be able to tell the difference, it's true, but hard to know in any given case without doing the extra measurements to find out.

Think of speaker measurement and correction as a diagnostic or methodological step, not an end product.  It may be that it's irrelevant in a specific frequency range, with a specific speaker, in a specific room (as in your case), but my example was intended to show that in the right room (or the wrong room depending on how you think of it), speaker measurement and correction could help resolve an issue that might be hard to diagnose just doing room measurements and correction. That's why speaker correction is valuable in the modal region, because without doing it before or alongside room correction, it can be very challenging to get optimal correction for multiple seats.

If you're only trying to get single seat correction, I agree that high quality room correction will get you just about everything you want in the modal region.

I hope I haven't confused the issue further  

[mattkhan]

It seems to be a question of terminology.

I don't know why you refer to that as speaker correction when you are correcting the effects of the room. Whether that correction is based on a single seat or many seems by the by to me (in terminology terms that is). At least that is my experience anyway, perhaps your experience is different.

I do agree that having the info from a near field measurement is valuable info though even if you don't do anything to it, both for the reason you give and for understanding the magnitude of room gain.

I agree with what you say though and I am not arguing for a mutually exclusive approach to any of these things. To be precise, when I said the modal region I mean the lower end of that where modes are more widely spaced. I think as you approach the transition frequency then things get less clear cut so coming at it from multiple angles may give better results. It is not something I have managed to do though as I have not found a good way to take a near field measurement of my mains (they have multiple woofers and multiple tweeters in close proximity).

[mwillems]

It seems to be a question of terminology.

I don't know why you refer to that as speaker correction when you are correcting the effects of the room. Whether that correction is based on a single seat or many seems by the by to me (in terminology terms that is). I agree that having the info from a near field measurement is valuable info though even if you don't do anything to it, both for the reason you give and for understanding the magnitude of room gain.

I was hypothesizing that the speaker itself had a 4dB peak at 60 Hz, not the room.  The point was that a room measurement taken in an anti-mode would not detect the speaker's 4dB peak. A 4dB response peak in the speaker is undesirable by itself (anywhere but an anti-mode), and would have the additional bad effect of making the 60 Hz resonant modes at neighboring seats go extra crazy.  The point was that just correcting the 4dB non-linearity in the speaker itself would be an improvement at 60 Hz over what the single seat room correction would have generated.  Does that make sense?

I agree that the room's reaction to the speaker's non-linearity is part of the issue, but even just correcting the speaker so it was flat at 60Hz with no attempt at room correction would have produced superior results at 60 Hz in the hypothetical.  

You may be right, we may just have definitional issues; but I think we're understanding each other's gist at this point  

I agree with everything you say though and I am not arguing for a mutually exclusive approach to any of these things. To be precise, when I said the modal region I mean the lower end of that where modes are more widely spaced.

If that's what you mean I think we're on the same page. I was referring to the range from 30 Hz or so to 200 Hz where most domestic rooms have strong modes as the "modal region."  If you just mean the range where a room is approaching the "no modes at all" region, I think I'm with you.

I think as you approach the transition frequency then things get less clear cut so coming at it from multiple angles may give better results. It is not something I have managed to do though as I have not found a good way to take a near field measurement of my mains (they have multiple woofers and multiple tweeters in close proximity).

We may have talked about this over in the other thread, but what kind of arrangement of drivers do you have?  Does measuring at the acoustic center not produce good results?  I have a 2x2 array of woofers and got good results measuring at the acoustic center (even measuring very close). It may be tougher with more widely or irregularly spaced drivers though.  

[mattkhan]

I was hypothesizing that the speaker itself had a 4dB peak at 60 Hz, not the room.  The point was that a room measurement taken in an anti-mode would not detect the speaker's 4dB peak. A 4dB response peak in the speaker is undesirable by itself (anywhere but an anti-mode), and would have the additional bad effect of making the 60 Hz resonant modes at neighboring seats go extra crazy.  The point was that just correcting the 4dB non-linearity in the speaker itself would be an improvement at 60 Hz over what the single seat room correction would have generated.  Does that make sense?

ah ok, I didn't get that was what you were saying. Interestingly that is roughly (I forget the absolute magnitude) my situation in that the sub has a hump centred on about 60Hz. Reducing this makes next to no difference to the in room response at the listening position, the modal response just swamps it completely. In another room, perhaps especially one that has significant treatment to deal with such modal ringing, then i imagine the result could be quite different. I have no experience of that though.

If that's what you mean I think we're on the same page. I was referring to the range from 30 Hz or so to 200 Hz where most domestic rooms have strong modes as the "modal region."  If you just mean the range where a room is approaching the "no modes at all" region, I think I'm with you.

In a normal sized room, at least a normal sized room for the UK, I am pretty much talking about less than 80-90Hz. This covers the 1st 2 axial modes but not much more hence the modes are easy to spot.

We may have talked about this over in the other thread, but what kind of arrangement of drivers do you have?  Does measuring at the acoustic center not produce good results?  I have a 2x2 array of woofers and got good results measuring at the acoustic center (even measuring very close). It may be tougher with more widely or irregularly spaced drivers though.  

my front 3 are MK MP150 so 2 woofers next to 3 tweeters. I've tried measuring at 20-30cm and at 1m, either way seemed a bit messy (each time aiming at the centre of the speaker) which I put down to being off axis from all of them, not sure really.

[mwillems]

my front 3 are MK MP150 so 2 woofers next to 3 tweeters. I've tried measuring at 20-30cm and at 1m, either way seemed a bit messy (each time aiming at the centre of the speaker) which I put down to being off axis from all of them, not sure really.

If you're game to try again, I'd advise measuring the woofers and tweeters separately in the near field, by aiming at the acoustic center of whichever stage you're measuring rather than the acoustic center of the speaker as a whole. 

So when near-field measuring the mid-bass section, I'd suggest positioning the mic exactly in between the two woofers.  Similarly, I'd advise measuring the tweeters by positioning the mic directly centered on the center tweeter.  Then stitch the two measurements together at the crossover.  That way you can create a composite near-field measurement that's on axis for both stages, if that makes sense.

At 1 meter, you should probably be on axis with the center tweeter rather than pointed at the center of the speaker.   Bass frequencies will find their way to the mic, Treble won't always.

[mattkhan]

If you're game to try again, I'd advise measuring the woofers and tweeters separately in the near field, by aiming at the acoustic center of whichever stage you're measuring rather than the acoustic center of the speaker as a whole. 

So when near-field measuring the mid-bass section, I'd suggest positioning the mic exactly in between the two woofers.  Similarly, I'd advise measuring the tweeters by positioning the mic directly centered on the center tweeter.  Then stitch the two measurements together at the crossover.  That way you can create a composite near-field measurement that's on axis for both stages, if that makes sense.

At 1 meter, you should probably be on axis with the center tweeter rather than pointed at the center of the speaker.   Bass frequencies will find their way to the mic, Treble won't always.

I'll give that a try next time I get a chance to spend some time on it as I'm in the middle of getting a 3D LUT going atm, audio seemed sufficiently done to move on

What distance would you advise measuring at?

[mwillems]

I'll give that a try next time I get a chance to spend some time on it as I'm in the middle of getting a 3D LUT going atm, audio seemed sufficiently done to move on

That sounds exciting, good luck!

What distance would you advise measuring at?

In the guide I suggest taking 3 inch measurements, 1 foot measurements, and a single 1 meter measurement on axis with the tweeter, and then averaging them, and I still think that's good advice based on my own tests. 

If you're only going to take measurements at one distance: I think the three inch measurements are the most important, especially if you're going to go on to do room correction later on.  If you get anomalous results, try moving out a little bit until things start to make sense.

[mattkhan]

That sounds exciting, good luck!

it was surprisingly easy; mash a few buttons, leave it going for a few hours, mash a few more buttons... and out pops a nicely calibrated projector

In the guide I suggest taking 3 inch measurements, 1 foot measurements, and a single 1 meter measurement on axis with the tweeter, and then averaging them, and I still think that's good advice based on my own tests. 

If you're only going to take measurements at one distance: I think the three inch measurements are the most important, especially if you're going to go on to do room correction later on.  If you get anomalous results, try moving out a little bit until things start to make sense.

OK. I've looked into the equivalent functions in acourate and all the tools seem to be available so I think I'm ready to give that a go when I get a chance to measure.

[mojave]

. . .I'm in the middle of getting a 3D LUT going atm, audio seemed sufficiently done to move on

Can you give me a link to how you did the 3D LUT?

[mojave]

Nevermind, I found it.

[mattkhan]

In case anyone else looks for it, I used this thread from AVS -
http://www.avsforum.com/t/1471169/madvr-argyllcms

Before and after measurements in this thread - http://www.avforums.com/threads/calibrating-an-htpc.1833478/#post-20728150

I was surprised how far my PJ had drifted in 18 months tbh but that is projectors for you I guess.

[mattkhan]

If you get anomalous results, try moving out a little bit until things start to make sense.

I took some measurements of the R speaker & am not sure what to make of them. I dumped them in my gdrive in case you were interested to look at the raw data.

I took 3" measurements for each individual driver, 12" for the woofers as a whole and tweeters as a whole and 1m for the speaker as a whole.

As far as I know, the crossover is at 1500Hz & the crossover is different for each tweeter so that the middle tweeter behaves differently to the top and bottom tweeters in order to control the dispersion pattern in some way (I've never seen explicit details of how this works so that's as much as I know.

All graphs are smoothed using acourate's psychoacoustic & FDW functions (basically what it provides as macro 1).

tweeters_3inch.jpg; red = top tweeter, green = middle, brown = bottom
woofers_3inch.jpg; red = top woofer, green = bottom
12_inch.jpg; red = woofer, green = tweeter
correction.jpg; turquoise = sub + r at listening position, blue = speaker at 1m, black = correction filter

FWIW I tried importing these as wav's into HolmImpulse but holm crashes every time I do that, not sure what format it is after (I tried the 3 different ways acourate can export it; 24bit pcm, 32bit, 64bit).

My thoughts;

Tweeters
- the individual tweeters do appear to behave quite differently, the 3inch measurements may well be useless as a result
- the 12in tweeter measurement shows rising frequency response above 10kHz which isn't in the 3in measurements
- the 12in measurement is about 3dB down from 2kHz to 10kHz, my target curve is the b&k curve (6dB down from 1kHz to 20kHz) which is also about 3dB down between 2-10kHz
- the 1m measurement does *not* show that same downward tilt >2kHz

My gut feel is that I should leave the tweeters alone as their natural response appears to be approximately the shape I'm after anyway. The mystery is the spike above 10kHz though this is not found in a measurement at the listening position.

Woofers
- the individual 3in measurements match quite closely; they both show elevated levels between ~130-300Hz, a dip centred on ~400Hz and then rolling off from 1kHz
- the 12in measurement accentuates the 400Hz dip but moderates the <300Hz levels though there is still a peak centred at ~150Hz
- the 1m measurement still shows that ~150Hz peak

The question here is whether I should do something about the 150Hz peak & also whether the 400Hz dip should be compensated for.

Looking at the correction filter I have I can see it is cutting <200Hz so this is being "fixed" without speaker correction.
The speaker is ~280mm wide which I think corresponds to baffle step issues hitting at ~410Hz. Is this coincidence? Again the room correction filter is seeing this as the 400Hz region is at 0dB in the filter so it is trimming around that region.

Overall it looks like acourate is handling this pretty well without needing me to correct the speaker. The 1m measurement (once smoothed) seems v similar to the listening position measurement >300Hz and you could argue that <300Hz, in an untreated room, is just going to result in correcting one way then the other for questionable benefit.

I would be really interested in other views though. All this is getting me more interested in building my own speakers

[mattkhan]

I then followed your guide for stitching together multiple measurements by doing the following

- splitnjoin at 1500Hz {sum of 3 3" tweeter measurements} with {sum of 2 3" woofer measurements} to produce composite 3" measurement
- splitnjoin at 1500Hz {12" tweeter} with {12" woofer} to produce composite 12" measurement
- average these 2 with the 1m measurement
- normalised gain of this average trace with my listening position measurement

this produces the attached graph (brown is my listening position measurement, green is the correction filter, blue is the average), I guess the question is whether the final correction is really going to change much if I were to correct the speaker in the 150-800Hz range.

[mwillems]

As far as I know, the crossover is at 1500Hz & the crossover is different for each tweeter so that the middle tweeter behaves differently to the top and bottom tweeters in order to control the dispersion pattern in some way (I've never seen explicit details of how this works so that's as much as I know.

Yes it definitely looks like they're using the tweeters differently to control directivity, I would base any correction of the tweeters on more distant measurements (1 foot or 1 meter).

FWIW I tried importing these as wav's into HolmImpulse but holm crashes every time I do that, not sure what format it is after (I tried the 3 different ways acourate can export it; 24bit pcm, 32bit, 64bit).

I think it's expecting 16 bit mono wav or a text file.

Tweeters
- the individual tweeters do appear to behave quite differently, the 3inch measurements may well be useless as a result

I think I agree.

My gut feel is that I should leave the tweeters alone as their natural response appears to be approximately the shape I'm after anyway. The mystery is the spike above 10kHz though this is not found in a measurement at the listening position.

All of your measurements (except at your listening position) show a differential rise above 10KHz, and all of them (including your listening position) show a 3dB-ish lump at 17 or 18 KHz.  

Sound becomes very highly directional above 10K, so even if your speaker has a rise at 10KHz at almost any real room position (we're talking a few inches off axis) you'll tend to get a roll off like you're seeing at your listening position, but if you look closely at your listening position measurement, there's still that same 17 or 18KHz lump.  That lump is probably a resonance in the tweeter (there's no other reason for a sudden high Q rise in FR up there), and even though sound that high up isn't super audible, it can cause intermod down below.  

It looks like the Acourate filter is trying to correct it, but if it still shows up in your "corrected" measurement, I would EQ it more aggressively.  I had a compression driver with a spike like that at 18.5KHz, and EQing it out was one of the best things I ever did for my highs.  I was skeptical that it would make a difference, but it improved my distortion measurements (in Holm) and (on a subjective note) it dramatically reduced listener fatigue.  I can't hear 18.5KHz unless it's very, very loud, but apparently it can still hurt my ears  

The question here is whether I should do something about the 150Hz peak & also whether the 400Hz dip should be compensated for.

It looks like Acourate is more or less doing exactly that.  I think your correction is probably doing a pretty good job with those already.  You could try generating correction based on the 1 meter measurement and see how it compares, but you're pretty close already.

The speaker is ~280mm wide which I think corresponds to baffle step issues hitting at ~410Hz. Is this coincidence? Again the room correction filter is seeing this as the 400Hz region is at 0dB in the filter so it is trimming around that region.

I think the baffle step is a coincidence, you'd expect to see a continued downward trajectory if it were the baffle step (rather than a dip and then a rise).  

It's either a non-linearity in the driver, or (more likely) its a room boundary cancellation effect (rear wall bounce is a likely culprit). How far are your speakers from the rear wall?  If the distance from the front of the cabinet to any wall is about .75ft (i.e. the speaker back is up against a wall or they're in corners), its very likely a boundary effect.  

I've also seen wide 400 Hz-ish modal dips in fairly small rooms (one I can remember that had the issue was about 8ft by 9ft).  It could also be a box resonance if your box is about .75ft deep (which you can test by placing something heavy on top of it and seeing if it changes)

How well does your acourate correction resolve that issue?  Is there still a dip there in your measurements with the correction on?  That will help you diagnose the source.

Overall it looks like acourate is handling this pretty well without needing me to correct the speaker. The 1m measurement (once smoothed) seems v similar to the listening position measurement >300Hz and you could argue that <300Hz, in an untreated room, is just going to result in correcting one way then the other for questionable benefit.

I agree that acourate is doing a very good job, and that may be (in part) because the differences between your 1 meter measurement and your listening position measurement are not very significant.  I've measured a few dozen speakers in a half dozen rooms and that kind of agreement between the 1 meter measurement and the listening position in an untreated room is a pearl of great price.  

You either have very a nice room (acoustically speaking), or acourate's filtering is even more incredibly sophisticated than I already thought it was

I would be really interested in other views though. All this is getting me more interested in building my own speakers

It's a lot of fun if you enjoy tinkering, and you can get sound quality that far exceeds anything you could buy for the same amount of money.  The time investment on the other hand... well, it's a hobby, right?  

Give a shout if you want to discuss any designs, I'm not an expert on speaker design, but I've built about four sets of speakers and a few subs, and assisted on several more.  So at the very least, I can tell you about some of the "pitfalls" we encountered and how we resolved them.

[Continued below]

[mwillems]

this produces the attached graph (brown is my listening position measurement, green is the correction filter, blue is the average), I guess the question is whether the final correction is really going to change much if I were to correct the speaker in the 150-800Hz range.

Given the relatively close resemblance of your 1 meter and your listening position measurements, I doubt it would make a huge change, although the listening position measurement does look quite a bit more jagged than the averaged response you attached. 

One thing that I haven't seen is how your system measures with the correction filter applied (and if you have more than one listening position, how the other positions look).  If you have one listening position and the corrected measurement looks nice and flat across the band, then it probably wouldn't help much to do the speaker correction first.  

But it may be the case that the correction filter is adding boost in places where it won't help at all, and you'll still have a dip there but with a goofy tonal balance due to all the boost, or it may be that the cut applied by the filter isn't enough to EQ out a particularly stubborn resonance, etc. Or it may be that the correction filter creates a really weird measurement at another listening position (if you have one).  There's just no way to know what's going to happen with any given correction without doing a corrected measurement.

If any of those are the case, those would be reasons to do the speaker correction separately, if you see what I mean.

[mattkhan]

Thanks for your comments.

Here's another graph, I've spread out the measurements so they are easy to see

green; listening position, uncorrected
blue; predicted result of convolution
red; listening position, corrected
brown; 1m measurement
turquoise; correction filter

It looks like the correction has spotted & fixed that 17-18kHz peak. It also looks like it might be overcorrecting a bit >1kHz, I might try tweaking the FDW parameters to see if that helps. There is also an anomalous behaviour around the XO (~1250-1750Hz) that could be looked at. The woofer seems to have been handled ok, the 250Hz peak is perhaps a room effect?

I probably need to go and check the L and C now. It's never done this thing 

[mwillems]

here's another graph, I've spread out the measurements so they are easy to see

green; listening position, uncorrected
blue; predicted result of convolution
red; listening position, corrected
brown; 1m measurement

It looks to me like the 400Hz piece is probably room related and the correction you've got is about as good as it's likely to get.  It looks like the correction did a good job of resolving that 17KHz resonance as well.

But the corrected measurement at 1.6 KHz and around 230Hz looks non-optimal and correctable. The 1.6KHz in particular looks particularly avoidable (that huge peak only exists in the corrected trace).  If I had to guess, since your crossover is right there, the application of boost and/or phase correction in that region is having unexpected consequences due to the interaction of the drivers at the crossover.  

Have you tried correcting the phase around the crossover (or is that a built in part of the correction)?  If not, the crossover region is one of the primary areas where phase manipulation can really pay dividends, and it's also an area where speaker correction can be helpful (because the phase relationship at the crossover can be hard to interpret/correct when off-axis).

Looking at your averaged trace of speaker measurements above, I'm not sure whether generating correction based on that would necessarily produce better results. It might, but it might not. It looks like it would result in less boost being applied at 1.6KHz at least, and if it has more "representative" phase measurements it may result in better correction around the crossover for that reason as well.

If you can manually dial down the boost at those two regions (or increase the cut),  that would probably get you 90% of the way there (i.e. the correction otherwise looks pretty good).  That said, since you already have the averaged speaker measurement, you could try correcting based on that and see what happens (since you've already done the hard part  )

In all seriousness, though, that's a nice looking corrected response overall. +/- 4dB around the target across the band is a big deal, even in a treated room.

[mattkhan]

It's either a non-linearity in the driver, or (more likely) its a room boundary cancellation effect (rear wall bounce is a likely culprit). How far are your speakers from the rear wall?  If the distance from the front of the cabinet to any wall is about .75ft (i.e. the speaker back is up against a wall or they're in corners), its very likely a boundary effect. 

I've also seen wide 400 Hz-ish modal dips in fairly small rooms (one I can remember that had the issue was about 8ft by 9ft).  It could also be a box resonance if your box is about .75ft deep (which you can test by placing something heavy on top of it and seeing if it changes)

they are on wall speakers, about 5-6" deep, and my room is 13.5' x 12' x 10'

Give a shout if you want to discuss any designs, I'm not an expert on speaker design, but I've built about four sets of speakers and a few subs, and assisted on several more.  So at the very least, I can tell you about some of the "pitfalls" we encountered and how we resolved them.

cool, thanks for the offer. It is just the germ of an idea atm but I'm quite tempted to just try and knock up some small monitors just for experimentation purposes really. I'm planning an extension atm which will contain a games room, it would nice to be able to build my own speakers & sub for that room. This gives me ~2yrs to experiment

It looks to me like the 400Hz piece is probably room related and the correction you've got is about as good as it's likely to get.  It looks like the correction did a good job of resolving that 17KHz resonance as well.

But the corrected measurement at 1.6 KHz and around 230Hz looks non-optimal and correctable. The 1.6KHz in particular looks particularly avoidable (that huge peak only exists in the corrected trace).  If I had to guess, since your crossover is right there, the application of boost and/or phase correction in that region is having unexpected consequences due to the interaction of the drivers at the crossover. 

Have you tried correcting the phase around the crossover (or is that a built in part of the correction)?  If not, the crossover region is one of the primary areas where phase manipulation can really pay dividends, and it's also an area where speaker correction can be helpful (because the phase relationship at the crossover can be hard to interpret/correct when off-axis).

acourate deals with phase correction "automatically", you can influence what it does by tweaking the frequency dependent window params & the strength of the pre ringing compensation but you can't control it directly (as far as I know). If you're interested in what acourate does then this site is really useful.

The basic method is that speaker correction is expressed through the XO files, this deals with linearising the speaker/driver & time alignment and results in the sweep you use to measure at the listening position. This measurement is used to create an amplitude correction for the room. Acourate then combines this with the XO to generate the final correction filters, this stage is generally an iterative process as you work through various combinations of FDW & PRC params until you get a stable outcome (things to consider include group delay discontinuities, pre ringing & IACC).

I think I'll take the Q around the effect on the XO to the acourate user group.

In all seriousness, though, that's a nice looking corrected response overall. +/- 4dB around the target across the band is a big deal, even in a treated room.

thanks, Acourate is a lot of work but the results really are great.

[mattkhan]

I took a few more measurements just to see if I could see more of what was going on.

Firstly I checked our 2nd seat just to see what the effect of the correction was on there. Acourate targets a single seat only but you can precisely control the frequency range over which correction is applied & you can apply arbitrary IIR filters on top. This means I could, for example, manually EQ the bass region based on a spatial average if I wanted to (I don't bother though atm).

seat2_L.jpg; red = uncorrected, green = corrected (left speaker)
seat2_R.jpg; brown = uncorrected, blue = corrected (right speaker)

I then look at the L speaker (previous graphs have been the R speaker) to compare the 1m against the LP, this is in 1m_LP_corrected_comparison_L.jpg

red; listening position, uncorrected
green; predicted result of convolution
brown; listening position, corrected
blue; 1m measurement
turquoise; correction filter

the same oscillation around the 1.5kHz XO is present here too though it's not really any worse then the actual measurement in the same range. The L speaker is +/-3 throughout apart from that XO region though so it's difficult to complain too much.

Finally I compared my LCR (all same model of speaker) at 1m, I thought this might say something about the room effects

red; left
green; centre
brown; right

I'm not sure what to make of this one.

[mattkhan]

It seems the anomalies must just have been a result of moving the mic between before & after measurements. I took the Q to the acourate mailing list and was asked to repeat the measurements more methodically. From this I can see I was doing acourate an injustice before as it is really producing +/- 1.5dB at the measurement position.

1-0 to acourate by the looks of it 

[mwillems]

Sorry about that, I just noticed this thread had some replies I hadn't seen/processed.  It looks like you sorted the issue out, though?  Regardless, that's a heck of a response curve.  I assume when you say "more methodically" you mean without moving the mic in between measurements?  

[mattkhan]

Sorry about that, I just noticed this thread had some replies I hadn't seen/processed.  It looks like you sorted the issue out, though?  Regardless, that's a heck of a response curve.  I assume when you say "more methodically" you mean without moving the mic in between measurements?  

yes basically. There were a few other tweaks around how I worked through the workflow & organised data to make it more obvious what was going on. I have to say the surround soundstage for films is quite amazing now.

[mwillems]

yes basically. There were a few other tweaks around how I worked through the workflow & organised data to make it more obvious what was going on. I have to say the surround soundstage for films is quite amazing now.

I bet! 

Surround is the next big frontier for me;  I pretty much wore out my welcome (and my wife's patience) building my washing-machine-sized mains, so I'm waiting a few years before introducing several additional speakers 

[Hendrik]

I still need to calibrate my system... something to do once I'm back home, just need to get a mic..
Calibrating a full surround system sounds like a lot of fine tuning.

[mattkhan]

I bet! 

Surround is the next big frontier for me;  I pretty much wore out my welcome (and my wife's patience) building my washing-machine-sized mains, so I'm waiting a few years before introducing several additional speakers 

I get a free pass to some extent as she uses the system more than I do. This didn't stop her rolling eyes when I was suggesting building a new sub with 2-3 18" drivers the other day mind you  ? 

I still need to calibrate my system... something to do once I'm back home, just need to get a mic..
Calibrating a full surround system sounds like a lot of fine tuning.

worth it though