Yet another MEH Synergy project

The Eminence Kappa 12-A LF on the MEH looks like it has a 2nd order acoustic bandpass LP at 500Hz, with its 2-4kHz frequencies "beaming" through the ports.
Screen Shot 2024-03-07 at 5.13.00 PM.png


MF looks ~ 2nd order <320Hz HP (surprised it makes it that low) 4th order 1kHz acoustic bandpass LP, with upper frequencies leaking through the ports. Anywhere from ~320-500 Hz between the LF & MF should sound good.
Smooth response up to 1kHz, I'd suggest a electrical crossover in that range.
Screen Shot 2024-03-07 at 5.10.49 PM.png

HF has a 4th order acoustic HP ~700Hz, plenty of overlap for a higher crossover point around the HF driver's 1kHz dip.

What distance where your measurements taken from the horn mouth?
 
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2nd completed today.

IMG_20240308_162024~2.jpg


And lifted into place. Angled to the side wall, as per Danley wisdom

IMG_20240308_211557.jpg


Did my driver checks as usual.
Wiring all okay.
Ran some sweeps, all looked okay.

On this cab I decided not to do any dishing driver side of the MR taps.

Resulting sweep

LH MR No PEQ.PNG


No XO or PEQ. Forgot to click up to 1/6!
Plenty low!

I also made a full volume rear cab cover for this one.
This has double the volume of the first one I made.

Other thing I tried was omitting the sound deadening sheeting, just to see the difference.

LF was down on SPL!
Upped it on the amp.
The LF plot isn't quite as nice either

(Pretty sure the sound deadening isn't what giving me less SPL on bass only. Other thing is the bass drivers are fully run in of course).

After tweaking similarly to the RH side, time for a listen.

First thoughts are, its actually quite polite, very revealing direct, but refined (less distortion from reflections of the Le Cléac'h and Tractrix?

More thoughts later..
 
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A little update.

Having tuned the pair yesterday and got the SPLs of each driver about right, things started coming together more.

I tried time aligning each driver within the horns to my listening spot.. I know, but I wanted to see:)
As expected having all 3 at the same setting and letting the Synergy do it's thing sounded much better.
What's the point of Synergy otherwise?

With only one 12" LF woofer per SH50 side (6dB down on Art's calcs :) ) there's still PLENTY of LF SPL in my room!

I can't say I can hear the difference double the reflex volume is making on the left channel vs half on the right.
I'm surprised at that!
I took the short port extenders out of the righthand side.. Less bass SPL. I'll try with them in both sides next, but might have to PEQ to smooth things a bit.

There's no port sounds that I can detect.

I had too much overlap with the tapped horns in the beginning and at higher volumes things in the room started resonating - not the speakers themselves but just random items that needed moving a bit :) :)

I then listened with the tapped horns off and there is still quite a bit of bass going on. 4 string bass guitar is covered pretty well by the SH50s alone!
Not quite deep enough for my tastes, but that's why I have tapped horns:)
I will try just one tapped horn next and see if that works. Coming it at 50Hz or so, should be directionless.

I like the LF tone and presence I'm getting.

The MF and LF presentation is quite a bit different to what I'm used to.
It's drier and perhaps more accurate.

If you could some how put that Vitavox S2 / Le Cleach horn sound inside the SH50 it would tick many boxes!
The BMS4550 is still a bit screechy at times (running in still esp the new LH side), but getting better with time.
The MF is there but not really standing out - also still running it,

I'm getting familiar with the different presentation these give, so called point source.

Some material is excellent, others I miss what the old system did.

With a pair, the true omi-directional ability is coming to the fore.
You can sit pretty much anywhere in the room and it sounds good, uniform and true. If you demo a lot or play to a larger audience perfect, I can see why these are good in larger home cinemas.

So, to sum up;
  • like the bass. Love the compact length (compared with old 1.3m long straight, no so LF horns).
  • like the omnipresentation sound.
  • MF and HF needs more running in / more tuning - even even small SPL changes improve things or worsen them.

Things that spring to mind immediately - these may change with running in and tuning more.

  • Could try a hybrid using the SH50 LF and my usual tractrix MF and Le Cleach HF - but that kinda defeats the whole object!
  • Plan the next project - a pair of SH60F alikes.. What coaxials to use, purchase 4 x 8" drivers etc etc.

I'm really enjoying the journey with these.
 
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If you could some how put that Vitavox S2 / Le Cleach horn sound inside the SH50 it would tick many boxes!
If you have a polar sonogram of the Le Cleac'h horn vs. the straight-sided MEH, I think you will see the difference in coverage response that likely gives rise to the subjective listening sound quality difference.

Since I can cursorily find no polar sonogram plots of the Le Cleac'h horns, I can't help you very much on that subject. (JMLC himself told me before his sudden passing that this was the biggest single difference in acoustic performance of his horn designs vs. "North American" horn designs having consistent off-axis response.)

But you can take the off-axis measurements of the two horns/compression drivers to produce a polar sonogram using something like OmniMic or similar app to produce the polar a polar sonogram of the Le Cleac'h horn/S2 combination from multiple off-axis measurements of increasing angles (e.g., 10 degree increments). Unfortunately, this is best done outside in order to avoid in-room reflections.

My biggest gripe with the SH-50 is that its polar coverage angle is far too narrow, leading to a subjective "hole" in the phantom stereo image when you move your head left or right from the symmetrical stereo axis pair in-room. I find that I would need two SH-50s side-by-side in each loudspeaker position to get the 90-100 degrees of horizontal polar coverage in-room that, to me, sounds correct. YMMV. That's why I used the K-402 horn for the K-402-MEH design. The following polar sonogram was made from data measured in my back yard using a spare K-69-A driver I had on-hand at that time (2015). The -6 dB polar response is in light blue color:

K-402-MEH Horizontal Normalized.jpg


Here is a SH-50 polar plot (horizontal beamwidth) pulled from an EASE data file provided by Danley Sound Labs, for reference:

SH50ver2.png.6b335140c68c50918c2a835e98c6c9e2[1].png


Chris
 
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@Speedysteve7 Could I have a request for you? I see you are pretty skilled in DSP adjustments. Could you please post impulse responses of the MEH three ways separately and one of the whole system? I am trying to understand how that should look like and what I am getting with my MEHs. That would really help me, thanks in advance!
 
@Speedysteve7 Could I have a request for you? I see you are pretty skilled in DSP adjustments. Could you please post impulse responses of the MEH three ways separately and one of the whole system? I am trying to understand how that should look like and what I am getting with my MEHs. That would really help me, thanks in advance
See above.

https://www.diyaudio.com/community/threads/yet-another-meh-synergy-project.409866/post-7622524

I think that covers your request.
I've since followed advice and raised the HP X/O on the HF.
 
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Took the back of the full rear cab.
Filled the driverless half with study IKEA zip bags I'd stuffed with needles felt.

Had a measure. That's better. Love it when a plan comes together.

LH RH LF equalized stuffed LH cab.PNG


This means that the tapped horns can do just 20-50Hz or so.
Can try going down to just one, for fun.

I've put chasing flatter phase aside for the time being.
EQ'd so it measures okay with each driver doing it's thing and no more, it's really starting to sing.

Daughters with younger ears listened to a lots of material they like.
Streaming is great for that. Not a song they wanted that Qobuz didn't have.

They noticed the missing top Freq 'snow'.
It doesn't bother me, but of course the Raal Lazy Ribbons I was using gave up to super tweeter frequencies, if the gear I use actually passed that on.
They described it as direct but tonally pleasing.
Compared it to the 5 way system favourably, commended the wider listening presentation than just the lightening spot.
The anywhere to wander around thing impressed.
As did playing it silly loud 😎
Thought it was a keeper.

They are very honest about these things.
Youngest would like the Raal ribbons playing over the top..

Note it's tme for a rest and settle down for a listen and resist fiddling too much
That is until I realise something can be improved upon😀
 
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After a while in the less than flat phase wilderness, and lots of time spent with fiddling with PEQs (both over and under done🙄🙂), 1st order X/Os on and off, and some time alignment to subtly shift phase too (is it right to do this Chris?,
I've read your posts and guides over and over!)...

I've got something a lot closer to flat over most of the freq range.

It sounds much better too.

I'm still not sure I'm doing it right, and the little kink (bubble), around the X/O point between MF and HF is still there but within 40 degrees phase shift now.

The LF still climbs as frequence drops - seemingly nothing I can do about that.

The higher HF frequency still has phase drop a bit.

However, It's a balanced sound.

Across the range it definitely completes with the old 5 way system now!

Just that HF smoothness that the Vitavox S2s and Le Cléac'h combo had - esp on female vocal, that I can't get out of my brain!

I'm loving the fact that the freq coverage of various instruments isn't moving around across drivers in a big physical space any more!

Time alignment of the two channels helped imaging.

A bit closer to audio heaven😂
 
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...is it right to do this Chris?...
Yes, of course.

It sounds much better too.
That's a good clue...

I'm still not sure I'm doing it right, and the little kink (bubble), around the X/O point between MF and HF is still there but within 40 degrees phase shift now.
That's probably not very audible. I wouldn't sweat that.

The LF still climbs as frequence drops - seemingly nothing I can do about that.
That's what's called "minimum phase behavior". That can be corrected using FIR filters, but remember that the required delays for the woofer below 100 Hz may quickly run you out of DSP taps, so you might not be able to change much of that. Some of that is due to the high pass nature of the woofer on the horn.

However, It's a balanced sound.
Great!

Across the range it definitely completes with the old 5 way system now!
Again, good news...

Just that HF smoothness that the Vitavox S2s and Le Cléac'h combo had - esp on female vocal, that I can't get out of my brain!
That could be the compression driver itself. I noticed that the BMS 4550 (1" driver) in the SH-50 was a bit less smooth sounding than the BMS 4592 ND (2" driver) in the K-402-MEH prototype. This may be a function of the driver's SPL response smoothness. I also noticed the smoothness of the Celestion Axi2050 (2" dialed-in carefully on a K-402) and the TAD TD-4002 2" drivers relative to the BMS 4550 in the Danley BMS 4550.

Chris
 
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Thanks Chris, I'm there pretty much then?!

What's nice is I did the RH side and just transposed the delays, PEQs & X/Os to the left side and it was spot on - my build was repeatable 😀

Yes, I tried that Rob. Just messed it up😂

I have 15 PEQs per channel limit in Najda, I know because during one 'hanmer it flat' attempt I hit the limit!
Now I've got 2 - 3 at most!

I had an audiofools epiphany and put some tube damper rings on the EL84 SET amp that drives the BMS 4550. I'll swear it sounds more refined - confirmation bias is a great thing😀

Thanks for all the help everyone.
 
I've been working away on my flatter phase and am much closer.

I'll share some plots soon.

Meanwhile, I have the feeling there must be a smarter quicker way?

I'm not sure of the exact tuning order for IIRs implementation though.
Read here and on the Klipsch forum posts by Chris etc, but not sure if interpreted them right?

1. No X/O on the LF and MF. BW 6dB/oct on HF hi-pass.
I am using 1000Hz at the moment

2. Volume level the drivers

3. Smooth the frequency plots using PEQs, hammer down the LF to interface with the MF. Hammer down any high freq rubbish from LF and MF that otherwise spoils the HF sound (had this to start with)

4. Further adjust to give nice enough Freq plot with the 3 drivers in unison.

5. Move time alignment of MF in relationship to HF back and forth to get sweet spot regarding phase flatness.
Same with LF to find the sweet spot.

6. Adjust PEQs to improve phase flatness

7. Kick back and listen🎶

How does this sound?
 
Meanwhile, I have the feeling there must be a smarter quicker way?
Well, there really is, but it really boils down to having just a little experience doing it once or twice. Let me explain...

Much of what I wrote on the K-forum about how to dial-in any loudspeakers using a DSP crossover was aimed at those that I eventually expected to help dial-in remotely. That is, they send me REW measurement files via email, and I send back updated DSP configuration files for them to import and take a new measurement in round robin fashion until the measurements both looked good and the owner said that it sounded good, i.e., not too bright for their tastes, etc.

So what's different about that way of doing it? Well, I've found that too many owners have such short attention spans that I had to amend my typical approach to dialing in. Instead of doing some initial basic measurements and settings to basically throw away later, I found that I had to use every measurement and added DSP parameters (i.e., PEQs, delays, etc.) and throw nothing away (thus resulting in more PEQs used in the end). Otherwise the owner gets antsy and start to go off on their own before the initial dial-in was complete and they heard the difference.

When I dial-in a new loudspeaker for myself in my listening room, here's what I actually do:

1) I zero out the DSP settings to nominal, then run individual measurements of each driver (temporarily muting the other drivers) so I can isolate one way at a time. I always use 1m measurements with the measurement microphone, except when measuring subwoofer response well below the Schroeder frequency of the listening room (where the listening position is, for this once time, the place to put the measurement microphone).

a. From this I can decide how to flatten the amplitude response of each driver within its passband (which I then do and then run individual measurements on each driver to verify and capture their individual flattened response)--i.e., how "deep" I flatten each driver's response in order to get flat SPL response and also (perhaps) increase the passband width of the driver by choosing to attenuate the peak of its response deeper using PEQs. Sometimes, one or both the the ends of the driver's SPL response may need to be carefully boosted in order to achieve an overall balance of channel gains.

This is especially true of the woofers, where they typically are the least "sensitive" (efficient) by several dB relative to the higher frequency drivers, and sometimes as much as 20 dB of boos is needed (like Klipsch uses in its Heritage Jubilee to get half-space flattened response). In the K-402-MEH, this boost is typically limited in order to simply attenuate the woofer's response to flat, but gain boosts are limited to ensure that the DSP crossover unit doesn't begin to run out of PEQ boost headroom. Instead, the drivers are attenuated, then the overall channel gain is boosted to match the higher frequency drivers. This way, it's ensured that the DSP's biquads (PEQs and slopes) are well within their headroom limits.

b. Then after the SPL response is flattened (nominally using the DSP output channel PEQs), I can see how to set the relative channel gains for each driver or "way" (the same type drivers in parallel/series) to get all the drivers adjusted in amplitude response (which I then do, and then run a measurement on each individual driver to verify).

c. I can also see the resulting phase response of each driver to see where a good place is to cross the drivers. You don't want to pick frequency where the slope of each of the drivers' phase responses isn't fairly flat--unless you're setting the low pass response of the woofers or midranges in an MEH to be at the "first notch frequency", where the phase generally is changing.

d. Then I look at a natural place to cross the drivers (and in an MEH, one of the drivers will be at its first notch (acoustic) frequency to take advantage of that resulting downslope of SPL as an acoustic low pass filter (using the MEH's inherent first notch cancellation to increase the downslope of the crossover). To this end, I look at how much I have to trim off each end of the passband of each driver/way using PEQs. For this, I use REW's EQ facility to quickly show me what attenuating PEQs to use on each driver output channel to achieve my desired -3 dB point for each driver. REW is basically indispensable for this duty, since accuracy predicts the resulting response of the driver after PEQs are applied. More on how this works later (perhaps).

e. Probably the most important part of the dial-in is matching the phase response of the drivers. At this point, only rough estimates of the actual output channel delays can be estimated by looking at identical measurement conditions of the microphone (i.e., the microphone is not moved between measurements of all drivers of a loudspeaker). Some guesstimating of driver's output channel delays can be done and implemented at this point.

For an MEH, it's pretty simple: just pick a delay value equal to 90 degrees of phase lag for each of the lower frequency drivers--and it will be spot on.

__________________________________________

(FYI: all of the above usually takes as many individual measurements as I allot myself for remote (email) dial-ins with impatient new DSP crossover users.)
__________________________________________

[Now that each driver or way of the loudspeaker is flattened in amplitude response and coarsely match in terms of relative gains (and there is actually a process I use to split the difference in gain structure on boosts and cuts of all the output channels--but I think you can envision how this works at this point), then I go about stitching together the "ways"...]

2. I run a measurement across two of the "ways" in one loudspeaker, then look at the both the spectrogram response and the excess group group response of the two ways to see how much the output channel delays must be adjusted.

a. After adjusting the delays, I rerun measurements to verify the time delays result in a "vertical" spectrogram or pretty much 0 (zero) excess group delay growth from high frequency to low frequency (or as small an excess group delay as possible while still getting smooth amplitude response across the crossover interference band). This usually takes the most time and expertise to find a happy medium. Once that medium is found, I move on to any other "ways" that have to be stitched together and repeat the process here.

b. After stitching together the second (or subsequent) crossover bands, then I run a full sweep of all drivers to verify time alignment and relatively smooth amplitude response across all drivers in a loudspeaker (or loudspeaker + local subwoofer in the same corner of the room). If I've made any errors in compensating the higher frequency delays (made in my head) with the lower frequency delays, it will show up here.

c. As a finishing touch on the entire passband of the loudspeaker (+ subwoofer), I add input channel PEQs to smooth the amplitude response across each crossing set of drivers. Then I look to add any further PEQs (or adjust existing PEQs) to achieve something like ±1.5 dB flatness (using psychoacoustic smoothing) across the entire spectrum to put the finishing touch on it. At this point, I do not typically add any "room curves" (downward-sloping amplitude response) to the higher frequency spectrum above 1 kHz, but there are many people that I think are fairly lazy about looking at the music they play, looking for clipping that's been added to the mastering of the tracks, thus dramatically increasing the "harshness" of the overall sound. All it takes is a simple "Clip Fix" and then a "Normalize" on each track to eliminate these issues so that room curves are no longer needed. (Sometimes, the mastering guy puts in a huge boosting PEQ above 1 kHz to make the tracks sound more "alive". This also needs to be "undone" to balance the tracks for hi-fi listening pleasure without resorting to draconian room curves to "plaster over" the bad mastering practices of the music that they choose to play.)

While the above explanation is nowhere near an exhaustive list of actions that I take, it does illustrate the steps that I go through, pretty much smoothly and without starting-stopping the process. To me, it's a smooth process.

Chris
 
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That's a great insight Chris.
I'll have to read it a few times to digest it all.

Meanwhile I got to this far today;

all 3 15 Mar PEQ and time XO optimised.png


Rather like crinkle cut 🍟😀

This is with all the drivers in phase.
No nasty frequencies from lower drivers playing too high.

I then meshed in tapped horns as usual, phase aligning on suitable mirror'd matching X/O slope.
I do like a bit of overlap of the LF of the SH50s and the tapped horns - bass head 🙄🙂

I'll be evaluating the sound later this evening
 
Not sure what you mean by 'digital version' ?
If you mean do you get rights to the settings in the DSP processor to freely give or share....well, I don't care what the law says, but that is not anything I'd ever do.
I think you need to own the speaker to have rights to the XO and presets, and the rights stay with the speaker. (just my 2c on what feels ethical to me...)

The SH-50 version I want is the regular passive xover version....again to dissect for the purpose of learning.
But same thing would/will hold with however I can copy the passive digitally....if I can build a good set of replicating FIR files, I would not view that they are mine to give out. Be happy to discuss how the xovers work generally...but that's about the line I think.

Besides, I don't want the passive xo or a digital replica or it....I want to try my own hand on the speaker...THAT i could give out :D

Have folks noticed any particular issues with the stock SH50 passive crossover? I ask because I have found a local seller who has 12 SH50s, built without the crossover, intended to be crossed over externally using a DSLP48. Topically this seems amazing, I can have total control over the drivers, but I'm wondering if there are any sacrifices made with it in the first place. There are other benefits to using a tri-amped model with external crossover (resilience mostly, for a nightclub setting), but for a home setting, I'm wondering if it will end up just being a hassle, or if there are potential fidelity/phase benefits to using an external crossover as opposed to just sticking with a stock model and simpler system.