Horn vs. Waveguide

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Charles Hansen said:
Thank you for accepting my apology so quickly!



But in the nodal region, the response varies with time. It takes time for the standing waves to develop and create nodes and anti-nodes. And I don't think anyone knows for sure how the ear/brain interprets the initial energy versus the energy integrated over time.

If the ear/brain integrated *fully* over time, then we could just use pink noise and graphic equalizers to get perfect bass in any listening room with any speaker (at least at one listening position). But it's not that simple.


Dr. Geddes will surely shed some light into this, since he has studied in great depth the nodal region and psychoacoustics too.

It might be more but I hope that the cochlea entrance with its larger ''hairy'' sensors located there integrates enough, because Pink is practical and available. It works good enough for LF in known home or club installations until now. But if we could see a clean impulse in a real room down to DC it would be a big bonus, and we could be sure of possible hidden performance and realization of it. Science fiction? We never know if someone will eventually come up with an idea...
 
Charles Hansen said:


I am not talking about the averaged in-room response curves that Stereophile sometimes publishes. I am talking about the CSD (waterfall) plots that they publish. If you look at these, you will see the exact same problems that Keith Howard described in his article I previously linked.

The point here is that the assumptions that KH made when modeling his data were valid ones, that correlate well with actual measurements made of actual speakers when using a 6 msec time window.

And if you take the time to read the article, you will see that the data gathered with a 6 msec time window is almost useless below 1 kHz. And furthermore, it is extremely difficult to achieve better measurements without the use of facilities which are far beyond the reach of all but a handful of large research institutions (eg, large anechoic chambers).
All the CSD plots shown are of very different time frames. I cannot see how one can make a judgement with those. Have you ever seen a CSD of an amplifier with the window size set to full MLS data length?
 
iand said:


You're correct that if you know the amplitude *and phase* response at all points across the mouth you can calculate the far-field response anywhere both on and off axis. But it takes an awful lot of measurements (spacing should be a fraction of a wavelength to get reliable results, which means a grid smaller than 1cm at 20kHz) and a massive amount of number-crunching to do this, and most people don't have the skill, the software or the time to do this properly.

Gated measurements work fine so long as there are no relatively high-Q resonances, so the impulse response falls within the pre-reflection gating time. This isn't always the case either...

My vote goes for outdoors, facing upward, with the mic above the mouth by at least 4x the largest dimension of the mouth. The noise isn't enough to annoy the neighbours (unless you're doing maximum SPL tests rather than frequency response) and the results are very reliable and much less prone to gating setup errors.

Ian


Agree ,
Ground plane is the way to go , I have never had much luck with gated response indoor below 800 hz , 1 K being the most acceptable really . Gated response outdoor on a 5 ft high table speaker facing skyward will work , then you might as well do GP.

Gating is best left for testing down to 1 K, i have seen too many erroneous graphs below this , when compared to GP or anechoic measurements.

A.wayne
 
"If the ear/brain integrated *fully* over time, then we could just use pink noise and graphic equalizers to get perfect bass in any listening room with any speaker (at least at one listening position). But it's not that simple."


Yes you set-up with Pink Noise and an RTA and gate it for 1 second and it all looks quite good. Take the same RTA and run a slow sine sweep and find your room modes. Odd how that works out.

I wonder at times how long a sustained note has to last before it will excite one of the modes. They are very high Q so you certainly should not see them with a short MLS gate time.

Rob🙂
 
soongsc said:

Don't know, just seems like one of those that like to refer to other data that nobody knows enough specifics about.


Charles is Ayre. A very respected engineer. His references to $tereophile articles are a lot because he caters very successfully for the High-End boutique industry, and he probably follows the discussions of that sector. To me his contribution is always valuable.

Ayre

P.S. Did you choose ''Ayre'' from original ancient Greek for ''air'' Charles? Or is it something else?
 
I'd like to get this thread back on topic or lets just drop this topic and start the current conversation somewhere else.

I remain unconvinced that a 6 ms window is a problem below 1 kHz. I have done measurements here with a 6 ms. window and at the Thailand factory with a 15 ms. window and there is no difference. One can "create" a hypothetical situation where they are different , ala the Stereophile "suimulations", but that does not make this situation one that we would actually encounter in the real world. I haven't, and I have done more than my share of measurements.

As to the Baffle Step discussion, this is always overblown. I measure my speakers in the enclosure that they will be used in so the baffle step - more accurately phrased as the enclosure diffraction - is in the measurement and need not be considered twice. If you use infinite baffle speaker measurements and want to put that same speaker into an enclosure of finite size, then the measurements will not be accurate. The first order correction to this error has come to be called the "Baffle Step". But by simply measuring the speakers in the enclosure that they will be used in you are getting the full correction not just the first order correction, hence it is far more accurate and is the right way to do things.

I didn't really follow the subjective assesment questions about LF sound fields, but here is what I have concluded on this. Below about 150-200 we tend to perceive the steady state sound field as it takes so much processing time in our hearing that "transients" are not really perceptable. This is why I do steady state measurements of the room below 200 Hz, but I do time gating above this point. Steady state measurements of the HF is very incorrect because they aren't what we hear. We hear the direct field - transient aspects - and the steady state, the reverb as seperate issues, so they should be dealt with seperatly. AT LF they cannot be seperate issues, there is only the steady state.
 
gedlee said:
I didn't really follow the subjective assesment questions about LF sound fields, but here is what I have concluded on this. Below about 150-200 we tend to perceive the steady state sound field as it takes so much processing time in our hearing that "transients" are not really perceptable. This is why I do steady state measurements of the room below 200 Hz, but I do time gating above this point. Steady state measurements of the HF is very incorrect because they aren't what we hear. We hear the direct field - transient aspects - and the steady state, the reverb as seperate issues, so they should be dealt with seperatly. AT LF they cannot be seperate issues, there is only the steady state.

Excellent. You put things into the right practical perspective. No direct vs rebound perception issues in the nodal region.
Measurement discussion has its own thread, we may talk more there.
 
Hi Earl,

Perhaps there is just one use for the cheap and cheerful method (mic at hornmouth) and it was this I was interested in - just to check the basic bandwidth of the horn. In particular the bottom end.

The question of what the lower limit of a compression driver is, if one makes a large enough horn is interesting (to some). You once kindly answered a question of mine on this. What it sounds like (and measures like in the far field) is another question.

Anyway I will now go and stand in the corner 🙂

Thanks and best rgds,

martin
 
Hello Martin,

Despite the critics about the measurement method you used, there is much more to find studying your curve than the simple checking of the cut-off frequency (at -3dB).

[BTW every Le CLéac'h horns with T between 0,7 and 0,8 we could measure gave strictly the same measured cut-off as the cut-off entered in my spreadsheet (or in Hornresp) during the designing process]

The main feature of the response curve you measured is its smoothness and this is a direct proof of the efficiency of the Le Cléac'h flare to avoid reflected waves from mouth to diaphragm (and the subsequent interferences). No parasitic dips and peaks = no reflected waves. No response ever published for a waveguide or a conical horn is so smooth, because their builders want to make them small and they terminate the mouth too quickly.

I can understand why some jealous people cannot admit that this was obtained without equalization (as the response curve of the TAD TD4001 mounted on a Le Cléac'h horn Fc = 320Hz)... ;-)

To close this message I have to thank you for your courageous work, specially on low-mid horns. Low-mid - because it is so important for the correct reproducing of human voices- is a domain for which experimented audiophiles feel strongly the need for horns + compression drivers (conventional direct radiating loudspeakers even mounted on horns may give good results, but they simply are not in the same league than compression drivers + horns after you have been exposed to the last).

Too bad that we cannot compare low-mid horns with low-mid waveguide since Earl's affirmed that "he failed to see how a waveguide below 800 Hz offers anything".

Best regards from Paris, France

Jean-Michel Le Cléac'h






Truetone said:
Hi Earl,

Perhaps there is just one use for the cheap and cheerful method (mic at hornmouth) and it was this I was interested in - just to check the basic bandwidth of the horn. In particular the bottom end.

The question of what the lower limit of a compression driver is, if one makes a large enough horn is interesting (to some). You once kindly answered a question of mine on this. What it sounds like (and measures like in the far field) is another question.

Anyway I will now go and stand in the corner 🙂

Thanks and best rgds,

martin


😉
 
Hello,

Yesterday evening, back from the office, I performed few measurements in my small living room on one of my TAD TD2001 compression drivers mounted on Marco Henry' J321 horn

(see Musique Concrete's webpage: http://www.musique-concrete.com/Etarifs.htm ).

The J321 uses a Le Cléac'h flare. It's cut-off frequency is 320Hz.

As my living room is full of pieces of furniture and loudspeakers enclosures ...

(I am currently preparing an enclosure using a full-range that I'll present at the contest organized by the French assocation Melaudia:
http://www.melaudia.net/2008concoursLB.php )

... there is several reflecting surfaces. I tried to reduce some of them using blankets, quilts... (but the whole enclosure being heavy I did'nt change it's position in order to use a better position for that measurement).

The reference level was first set to 94dB at 1 meter on a 1kHz sine signal. Then the measurement was performed using the logsweep + convolution method with a 30 seconds logsweep of sine from 200Hz to 23000Hz at 48kHz (Angelo farina's module "sweepgen" and "convolve" were used inside CoolEdit Pro for that). This allows to recover a fine impulse reponse.

The TAD TD2001 was fed by my transconductance amplifier (Zout = 111ohms).

NO equalization and NO denoizing were used or performed on the signal (well yesterday evening ther was a lot of birds singing and my neighbour's daughters were viewing an ado movie on their TV set...

You'll find the obtained pulse response (Fs = 48kHz, 16bits) in attached file (ASCII format).

Best regards from Paris,

Jean-Michel Le Cléac'h
 

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