Geddes on Waveguides

How significant is the throat length?

I occasionally found this:
http://www.faitalpro.com/img/products/schede/CD/HF10TX/HF10TX_datasheet.pdf

It seems a zero "throat length" on this driver. Supposedly this should be a "good start" for the following WG.

However there're still a step when the sound wave escape the passages -- especially the central region where all 'spokes' join together. Should all the spokes be razor-sharp to let sound wave leave cleanly?

Any comments?
 
I agree. The human hearing system is extremely inaccurate and insensitive above about 8 kHz, and then only if your young. If your old, like me, then above 8 kHz is nonexistant. And the audible content of frequencies this high in nature, including live music, is virtually nonexistant too. These frequencies only exist in recordings where the air absorption has not had the chance to dissipate everything up this high.
 
gedlee said:
... These frequencies only exist in recordings where the air absorption has not had the chance to dissipate everything up this high.

which is nearly every modern studio recording where sometimes the mics barely clear the player's knuckles

I believe I've seen it mentioned that Redbook CD pre/de-emphasis can't be used with current close miced recordings due to the high frequency content

but I certainly won't argue that the top 1/2 - 1 octave of Redbook CD does me a lot of good anymore
 
pjpoes said:



I'd be surprised if you could reliably detect it in blind testing, let alone find it worth fixing. I'm willing to be open minded, but it would surprise me.


gedlee said:
I agree. The human hearing system is extremely inaccurate and insensitive above about 8 kHz, and then only if your young. If your old, like me, then above 8 kHz is nonexistant. And the audible content of frequencies this high in nature, including live music, is virtually nonexistant too. These frequencies only exist in recordings where the air absorption has not had the chance to dissipate everything up this high.
I probably have mentioned this before, but I will share again.

I really did not know cone resonance at this kind of frequency was a problem, but while listening to Placido Domingo sing, I always detected a range of notes where suddenly the sound loses focus and very annoy sound makes me hate to listen to his singing. After reviewing the CSD, I decided to take care of a breakup mode around 20KHz; the better this was handled, the better Placido Domingo's singing was. As you can see, my efforts evolve from insatisfaction with existing sound, try to figure out what is wrong, then improve it based on measred data.

To be able to understand what I say, it is really necessary to first aurally detect what is wrong in the playback. If one is already quite satisfied with existing designs, then one will never reliably detect a correlation between technical data and listening experience.
 
SoongSC

This kind of anecdotal evidence is not to be given too much credence as there are too many other factors involved. I could give lots of evidence that anything above 10 kHz is meaningless as no one can hear it. In a concert hall, beyond the first row, no one hears anything above 10 kHz, because the air absorbs it all.
 
Gedlee,

I'm sure you have your own experience that dominates how you design systems. My purpose is just to share this particular experience in why I choose to take greater care in frequencies this high. There are lots of things that I choose not to do, but they have no significance in audio quality improvement, so I just don't mention them. Also, trying to find 1 people that agree with me is not my goal, it's the people that can find fault in my designs and can point them out in very specific music passages that are credible. In the threads that I have visited, I have only seen one person that expressed his listening experiences in a way that can do this.

The thing about concert hall recordings is where you put the mics. That is where the original recording is made. Why would I want to give a playback impression otherwise?
 
Originally posted by gedlee Doesn't LEDE use the absorption behind the listener?

No, from "THE ROLE OF THE INITIALTIME DELAY GAP IN THE ACOUSTIC DESIGN OF CONTROL ROOMS FOR RECORDING OR REINFORCEMENT SYSTEMS BY DON DAVIS" (AES paper 1547):

"Tho acoustic goal in the control room is to insure that its ITD is made
longer than the ITD of the studio. This allows the ITD of the studio
to be reproduced acoustically in the control room, unmasked by early
control room reflections. Thus a series of options opens. [...]
The most obvious option is to make the entire front half of the control
room non-reflective at the geometric acoustic frequencies. Then, by
proper spacing of the listener relative to the rear half of the room,
which is made reflective and diffuse, control room ITDs of from 5 or
less msec to in some cases over 40 msec are viable alternatives."

Best, Markus
 
soongsc said:




I probably have mentioned this before, but I will share again.

I really did not know cone resonance at this kind of frequency was a problem, but while listening to Placido Domingo sing, I always detected a range of notes where suddenly the sound loses focus and very annoy sound makes me hate to listen to his singing. After reviewing the CSD, I decided to take care of a breakup mode around 20KHz; the better this was handled, the better Placido Domingo's singing was. As you can see, my efforts evolve from insatisfaction with existing sound, try to figure out what is wrong, then improve it based on measred data.

To be able to understand what I say, it is really necessary to first aurally detect what is wrong in the playback. If one is already quite satisfied with existing designs, then one will never reliably detect a correlation between technical data and listening experience.
Our sense of hearing is very easy to influence. After fixing the breakup mode, did you blind ABX it? Would be a little hard to do (I imagine a switch that randomly chooses either the existing setup or the corrected setup), but that would be the best way to check if it actually makes a difference.
 
454Casull said:

Our sense of hearing is very easy to influence. After fixing the breakup mode, did you blind ABX it? Would be a little hard to do (I imagine a switch that randomly chooses either the existing setup or the corrected setup), but that would be the best way to check if it actually makes a difference.
Well, the process was not that simple. Many methods were tried to reduce cone breakup, then listening tests were conducted when data showed improvements that looked promising. Many times the improvement was not significant enough even if the difference was audible.
 
I used to spend a lot of time trying to fix the breakup in hard tweeters like the focals which takes place well past 10khz. I would play with crossover tweaks, lenses, phase guide modifications, and in the end, I generally found that response anomalies that high were typically not audible, and often the corrections would cause other problems in the range that is audible. I can measure a smoother response in the top end by removing the phase guide on the Focal tweeter, however the phase and response problems down lower are audible, and crossover tweaks in that range seem to be a bad idea. Instead I leave it in place, Focal knew what they were doing when they designed that tweeter (Which makes me question what Wilson is doing).
 
My horn crossover ignores inductance in third order mode Why it is that? No matter which value, SPL curve remains constant.
 

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