A Study of DMLs as a Full Range Speaker

De-essers are perhaps a bit off topic here...I guess the main question you are discussing is the importance of content above 8-10k?

But I would not say it is important for a de-esser to only compress a narrow band. Before there where dedicated machines and plugins you would simply send an bandpassed signal to the key input of a compressor applied to the whole signal. For vocals that will be perfectly fine. When singer sings a sibilant like "s" there is really no other important information in other bands, so reducing the whole signal is perfectly fine.
For some more creative uses, having the compressor only act on some band can be useful, but then I would say the unit is more intended as a multiband compressor with key input.

As for importance of contents above 10k, I must say that I find it very likely to be some kind of misunderstanding when a grown man claims to be able to tell nuances of what is going on at 18k. I done mixing since I was a teenager, and even when I could hear in that range I found that it has very little effect what is going on up there. On the typical software EQ that displays a graph over EQ curves, 10-20k is typically something like 1/10 of the graph, and to address say only 16-20k you would need an extremely narrow bandwidth (Q) for the EQ, which introduces a lot of phase shift and/or ringing and can sound bad.
For example, here is stock EQ in Bitwig, with an EQ with default 1.0 Q applied at 15K:

1712590245806.png

As you can see 10-20k is just a very little part of the display, and the 1.0 Q cuts the whole top octave and more.
The reason both the display and Q factor is scaled like it is is that making fine adjustments above 10k is typically pointless. The space between 10 and 20k is not only less important because it is beyond what hearing is focused on and only contains high order overtones from instruments, but also one octave is one octave. So 100-200Hz is the same interval as 10-20kHz, and you have the same problems to fit several EQ's and need a very narrow Q if you want to affect part of the octave only.

I appreciate some teenagers can hear the last octave and think it might add a little shimmer when it is there, or sound a bit shrill when it is too loud. But as soon as someone is describing specific qualities to specific bands in that octave, usually my BS meter goes off.
 
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Hi there folks :)
I have been reflecting on my experience building my DML panels ( which now sound great ) and how
the initial sound was 'horrible & honky'. It required EQ to get them sounding good.
I also recall a post by another member - calling the sound "honky".
It occurred to me that if both I and another person experienced this same 'Honky' sound, then there would
surely be other people that have also found this shortcoming with their panels. That honky sound.
This got me thinking to present a very simple solution to solve/improve this problem for those experiencing it.
The solution/modification is a simple passive filter requiring only 2 components of standard value. (easy to get)
One coil/inductor and one capacitor.
( this modification will also improve upper region sound by removing a 'masking effect' )
So this is a simplified bandstop filter??
Please share your calcs
Thanks
Eucy
 
JoskaNZ
For what it's worth, this is what I would do

1/ prepare the panel allowing for about a 10mm wide mounting strip around the edges.

2/ make a backing board from a material with poor acoustic properties (IE non resonant --(MDF or similar))

3/. Cut a circle approximately 200 mm dia in the backing board centred on the exciter location

4/. Make an edge strip for the backing board to achieve a final gap between the panel and the backing of no more than 20mm. You can experiment with different edge treatments. I used soft grade EPS strips directly glued to both backing and panel with some success. A more standard approach would be a rigid material (MDF or similar) topped with an absorbent foam seal to which the panel is glued.

5/ I would also line the inside of the backing panel with an absorbent material. I think even 8mm or so closed cell PU foam sheet stapled to the backing (as used for packing protection) would work... This will further reduce the air gap but I'd try it before making the gap bigger.

6/. Corner treatment... Tricky topic but probably try full mounting first before checking if any corner release is necessary. If so, you could release each corner and stuff the gap with fibre fill

7/ Supports to suit

If you do choose to try this I'd be pleased to get some feedback

Cheers
Eucy
 
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Epoxy has a lot of variations in formulation. 5 minute epoxy for example stays soft even when cured. Heat cured laminating resin has a sharp ring to it, very stiff. In a composite structure it holds the fiber in place, they carry the load. When using it as a paint layer, there is no guarantee it will be stiffer than shellac, or PVA or another material.

Determining Epoxy's Physical Properties - Epoxyworks

The flexural character is the material is the key. The thinnest amount of glass or carbon veiling would be much stiffer. The matrix that you hold it to a thin panel would probably not even matter. PVA, spray paint, any drying liquid that would wet it out and be able to be applied even and thin.

RC groups will have more information on veil in combination with foam and application methods. Carbon Fiber Veil cloth - RC Groups
So I have you used cf veil on styrene foam or other materials? if so how thick is the thin version you mentioned? I wonder would non woven polyester sewing interface material work do you think? I would nor expect it to be as strong as glass or cf but in situations where you would not need a lot of stiffness such as thin low density eps rigid foam (the kind used for packing and light insulation) this would be stiffer than a full strength layer of pva on each side and I would guess lighter if using the cf but I am not sure how heavy the polyester non woven interface material is.
If you have built panels out of this material how did they turn out and what materials did you use? Thanks for posting
 
Hi there folks :)
I have been reflecting on my experience building my DML panels ( which now sound great ) and how
the initial sound was 'horrible & honky'. It required EQ to get them sounding good.
I also recall a post by another member - calling the sound "honky".
It occurred to me that if both I and another person experienced this same 'Honky' sound, then there would
surely be other people that have also found this shortcoming with their panels. That honky sound.
This got me thinking to present a very simple solution to solve/improve this problem for those experiencing it.
The solution/modification is a simple passive filter requiring only 2 components of standard value. (easy to get)
One coil/inductor and one capacitor.
( this modification will also improve upper region sound by removing a 'masking effect' )
1712651872776.png
1712651895419.png

It's a simple series notch filter that places an impedance into the line at about 600hz. The frequency of the notch does not change with load impedance.
The inductor coil in this circuit has a DCR of 2 ohms. The lower the DCR, the deeper the notch and the better the lower frequency response.

Download Xsim. It's very simple. And it's free: https://libinst.com/Xsim/XSimSetup.exe (This link won't take you to a website. It will download the file directly into your default download folder. Look for it there.)
 
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I was relieved to find a solution, now to make semi closed backs...
I do this with my PA panels, but only because I use a sub on stage, and I need a reduction of SPLs behind the speakers.
I'm messing with various positions and shapes and holes on the rear absorber to reduce the amount of attenuation on the bass end. The panels bend in specific positions at specific frequencies, and the rear absorber can be used to flatten the response.
As it is, I use a wide bandwidth absorber (25mm honey-comb-shaped holes with the rear panel lined with carpet felt on the inside) and this does effect the bass end AND the over-all efficiency of the panel.
 
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Thanks Eucy, that sounds like a good method of backing the panels. Have you tried this yourself? I may give it a go if my solution isn't good enough.

For these ones, as I have the frames made out of Oak and I want a slim solution that will look nice in the lounge, I don't have a lot of depth to play with so am thinking of making a back frame with 20mm fibreglass insulation sandwiched between packing blanket, hidden with speaker grill cloth. I did a quick measure and seems to cut above 400hz - 5db to 10,000hz - 10db approx. Below 400hz not affected. I did just have the thought that a layer of mass loaded vinyl could be added, might get some to try.

I wanted to keep away from a solid back as I imagine then I will have the rear sound reflecting back onto the panel which may sound worse?

Andre, I haven't seen that honeycomb absorber I will look it up. Glad to hear it helps you.

I don't want to lower the efficiency too much, obviously it will have less output from the back of the panel but that was causing the issue in the room.
 
As it is, I use a wide bandwidth absorber (25mm honey-comb-shaped holes with the rear panel lined with carpet felt on the inside) and this does effect the bass end AND the over-all efficiency of the panel.

Carpet felt is great for this...I have used it a lot. I didn't mention it because it's quite hard to get in Aus now .. replaced by crappy faux rubber underlay which fails after a few years.

Andre, on the bass, from what you're saying I gather in your case it's reducing the bass??

The so called air spring is supposed to increase the bass response, but I guess if you open up the rear panel as much as you have that effect may well be lost

Eucy
 
from what you're saying I gather in your case it's reducing the bass??

The so called air spring is supposed to increase the bass response, but I guess if you open up the rear panel as much as you have that effect may well be lost
I was messing around with the panels without the rear absorber in place, and congratulating myself that the bass was coming through nicely. Then, when I put the absorber back in place, all of my congratulations disappeared... I didn't bother measuring.

I think the air spring is supposed to work on the first few modes where large-enough areas of the panel are out of phase with each other. The air spring will possibly reduce the out-of-phase component by a momentary high pressure/low pressure on a specific mode pattern. The air spring is not a totally enclosed back panel. It has large cut-outs in it that appear to correspond to specific modes. I have not spent enough time with it yet to come to any real conclusions.
 
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Good link.
Yes there are some middle-aged "golden eared" hi-fi pundits who think they can hear sibilance beyond 10kHz. But what they're most likely (mis)hearing is second-order distortion of a 4 or 5khz signal. This would most likely be caused by the improper setup of a stylus/head/arm or the tracking force on a turntable, or the break-up modes of a cone driver working beyond its pistonic mode. There are multiple possible sources of HF break-up in home "hi-fi." Sometimes ears themselves can be to blame since ear drums can also introduce distortions due to injuries, disease or even having had grommets. Sometimes a boost at 10kHz (found on low-quality vinyl pressings) will have an effect on the signal from 7khz up to 13khz, depending on the filter Q.
This high-frequency boost plays havoc with sibilance on vinyl pressings.
But a professional recording studio will not allow any kind of sibilance to be released to an album. It would be like a surgeon leaving tools inside after he's sewn you up. I'm not referring here to home studios or self-proclaimed 'producers" who wouldn't know the back end from the front end of a ribbon mic.

There seem to be basic misunderstandings abounding here regarding what "sibilance" actually is, and how a de-esser works to minimise it.

De-essing IS NOT a broad-band EQ or a shelving filter just tacked onto a vocal track. Such an EQ would simply make the vocal track dull and lifeless.
De-essing is also NOT tacked onto the final mix. Sometimes the mastering engineer cannot correct a final mix that has not had the vocal (or drum) tracks properly de-essed.
De-essing IS a high-Q, narrow-band filter that drives a dedicated narrow-band compressor. The de-esser is set to the correct target frequency which is found by manually sweeping a narrow-band peaking EQ over a looped section of that part of the vocal track that carries the objectionable sibilants. I have never, in 40 years in recording studios, ever seen any recording engineers find objectionable sibilants above 9kHz, except in one case where a lady vocalist had dental braces on the inside of her upper teeth.
The de-esser is triggered by the high-frequency "s" or "t" or "shh" signals, which fires the compressor to duck that part of the track without effecting any other frequencies on that track.
Sometimes a mix engineer might tack a de-esser onto a drum track to reduce the hi-hats or a snare attack without disturbing the rest of the drums EQ. It can also be used to reduce the plectrum sound on a very bright acoustic guitar mix. It's a powerful tool in the right hands.
Very well written post there.

I thought to complete this 'off topic' aspect of sound would be to post something about "the other side of the coin" >
being dullness, and solutions that are ALSO non broadband EQ based.
This is probably the most famous product to correct 'dullness'.
https://www.waves.com/plugins/aphex-vintage-aural-exciter

@Andre Bellwood might even elaborate & expand upon this :)
 
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Thanks Mister Audio.
Yes, an aural exciter of some description is part of my toolkit when mixing tracks of any kind. It adds second-harmonic harmonic distortion to signals between, say, 3k and 6khz or so (depending how you set it up) which means that sounds are added between 6k and 12khz that were not present in the original signal, and gives the effect of brightness and clarity. It is usually used on any single track that requires an added bit of 'life", but it can also be used in the final mix. Obviously, since this distortion is on the recorded track in the final mix, the end listener does not pick it up as distortion, but rather as a detailed, airy sound.
On the other end of the spectrum, the low end of the mix can also be boosted sans EQ by performing a similar trick with a bass booster (https://www.waves.com/plugins/maxxbass ) which is a different type of add-in, which either halves the lower frequencies (frequencies from 60 to, say 200hz), and adds the resultant signal back into the track, or it adds sub-harmonics to the track to produce a psycho-acoustic illusion of more bass.
 
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Andrew bellwood.
On the other hand maybe when I listen to the George Michael CD , I can hear the sssss and tttttt on the CD recording, and so can everyone else.
It is on the CD, and I'd say they wanted it there, to enhance the sound, it even has its own echo.
But my point was that these sounds can set off distortions in the panel and exciter, which is very painful if not delt with.
The cavity distortion can be anywhere around the 3k area, and the panel distortion somewhere around the 10k area.
This all depends on the panel and exciter used.
Ignore them at your peril .
Steve.
 
and the panel distortion somewhere around the 10k area.
There is a interesting situation regarding the High Frequency performance of DML panels.
The high frequencies are 'forced' through/out of a medium that is measured in Grams vs Milligrams.
The high frequency performance of a DML can sound 'sibilant', This is part of the physical function = A form of distortion.
This is why I chose to use a Tweeter in my design.
 
There is a interesting situation regarding the High Frequency performance of DML panels.
The high frequencies are 'forced' through/out of a medium that is measured in Grams vs Milligrams.
The high frequency performance of a DML can sound 'sibilant', This is part of the physical function = A form of distortion.
Can you explain what you mean?
 
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Can you explain what you mean?
Lets consider music content of 5Khz upwards ...
With a tweeter, the electo-magnetic-motor energy has to only move a diaphragm that has a weight of milligrams.
This allows for fast acceleration and also recovery. It is such characteristics that make tweeters work so well with high frequencies.
( this situation could be considered as > Mechanical Slew Rate )
When a DML panel is used, the high frequency energy of the exciter finds 'opposing mass' regarding very fast acceleration.
So, in practice the high frequency performance of a DML is very much 'a forced thing' > This creates distortion.

Of coarse you don't just add a Tweeter, you also take high frequencies away from the DML panel using a coil/inductor :)
 
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I cannot say I recognize that there is an issue with HF distortion with DML. In fact I experience the opposite, especially when playing on high levels, that DML does not distort in the same way as regular speakers. I had several people commenting how clear and non fatiguing the treble is on my system.

Of course like most people I cannot hear things like "singing at 18K", so I might be missing something. I do have a small distortion peak around 4k, but seen no evidence of distortion above that though. Perhaps it depends on material, coating or exciter used?
Do you have any recorded responses in for example REW where it shows?

Also, I would say there is no such thing as "musical content above 5k". While some instruments have overtones stretching in to the last couple of octaves, even the ones that does stretch quite high like piano and violin will produce fundamentals around 4k tops. Put a 48dB HPF at 5k on any track and you will hear some hihats and maybe occasional sibilance, but nothing resembling music. Most fundamentals will be below 1k and you have to go far below 5k to catch enough overtones to actually get the faintest idea about the musical content.
 
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