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A Broad Discussion of Speakers with Major Audio Luminaries

Measuring what? Frequency response? Radiation pattern? Distortion? Relative (discounting delay) phase of drivers?
i am not even remotely an audio luminary. but i think that's their (Kippel) NF FR measurement claim from what i read. others like here and elsewhere claim +/-1dB, which I doubt i could hear, personally and sorry if my ears are not to audiophile standards in 2025. don't want to waste anyone's time if my question is utterly stupid. Guess my question is about the transition from "what's good enough?" to "what's stellar enough for you to stop worry about this?"
 
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15 degrees inside of one 1/3 ERB, or 45 in an ERB. Gammatone filters are an ok sort of estimation of ERB's. I'd think 1/3 ERB envelope correlation with the ones above and below it ought to be simple enough, yes?

Thanks for replying. This is very helpful.

Looks like I need to familiarize myself with ERB's.
 
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15 degrees inside of one 1/3 ERB, or 45 in an ERB. Gammatone filters are an ok sort of estimation of ERB's. I'd think 1/3 ERB envelope correlation with the ones above and below it ought to be simple enough, yes?

For Duke: JJ said something similar in this older post from 2017 and went into a bit more detail. He did say "pi/6 or so" (30 deg) per ERB but I suppose knowledge may have advanced a bit since then :)
 
doesn't their website state their accuracy is within </+/- 1.5dB?
No, these are the specs:
Measurement accuracy:
In maximum SPL direction 1+/- 0.1 dB
In all directions +/- 1 dB


Maximum SPL occurs in on-axis response so accuracy can be incredibly good. There is a drop off in off-axis because the microphone is being used in other angles. There is no compensation for this but I don't use it.

Ultimately the accuracy is limited by the calibration of measurement microphone and number of measurement points uses to compute the response.
 
This is pure speculation, but I think that comb filtering that is the result of mixing "in the air" is far less audible than comb filtering that is "mixed electrically".
This is indeed the case and was tested by David Clark in his AES paper: Measuring Audible Effects of Time Delays in Listening Tests

Here are his conclusions:

"7 SUMMARY AND CONCLUSIONS
In this study a cascaded all-pass circuit and three full-band dual path systems were evaluated in a realistic listening environment. Two speaker mono was considered superior to the one speaker, one path mono. A reflection from a vertical surface was barely audible but a horizontal reflector were more audible. An electronic delay comb filter was highly audible and annoying. The all-pass filter was inaudible. With careful interpretation e TOS analyzer wee able to show probable cause for the audibility differences. It ia felt that elementary "two eared processing would give even closer correlation."
 
wow this is an awesome discussion and i have learned a lot from reading it.

my only question is - Kippel is the ultimate tool for utter accuracy when fighting about final measurements? doesn't their website state their accuracy is within </+/- 1.5dB? isn't that pushing us into much ado about nothing territory with many corner cases?
It is only the ultimate tool for when one does NOT have an anechoic chamber.
So one can do “pseudo anechoic measurements” in say a garage.

One gets close to the far field patter, but there can be some residual errors.
So as JJ points:
Measuring what? Frequency response? Radiation pattern? Distortion? Relative (discounting delay) phase of drivers?
 
For Duke: JJ said something similar in this older post from 2017 and went into a bit more detail. He did say "pi/6 or so" (30 deg) per ERB but I suppose knowledge may have advanced a bit since then :)

The exact number is not well known. I use 30/ERB but that's probably too conservative. But I have to be picky.
 
Thanks everyone, reading you makes me more and more convinced that both conceptually and practically, phase and time coherence are crucial to achieving high fidelity.
 
It is only the ultimate tool for when one does NOT have an anechoic chamber.
No. It blows the pants off of any reasonably sized anechoic chamber. NFS produces reflection free response down to 10 Hz. I don't know of any anechoic chamber that is so below 80 Hz. The advantage of anechoic chamber is speed. Not fidelity.

So one can do “pseudo anechoic measurements” in say a garage.
There is nothing "pseudo" about it. You are confusing gated, merged near-field measurements with what Klippel NFS performs. Klippel NFS uses power of computing to solve physics that would call for massive anechoic chambers. New solution to an old problem.

One gets close to the far field patter, but there can be some residual errors.
Anechoic chambers have that problem unless they are massive and you are measuring very far. And if they are doing that, then you have to worry about temperature gradients causing phase errors.

Learn the topic first before writing such incorrect claims.
 
Measuring what? Frequency response? Radiation pattern? Distortion? Relative (discounting delay) phase of drivers?
Indeed, it's always wise to specify both the units and the quantity being measured before discussing tolerances.

Here's a classic anecdote featuring two Nobel laureates in physics:

When Pyotr Kapitsa first approached Lord Rutherford at the Cavendish Laboratory, he asked to join the research team. Rutherford replied that the lab was already full - about thirty scientists - and there was no room for another.

Kapitsa then asked: “What is the accuracy of your measurements?”
Rutherford answered, “About two or three percent.”
Kapitsa responded: “Then adding one more person - me - would still be within your margin of error.”

Amused by the clever reply, Rutherford let him stay. Kapitsa went on to spend over a decade at Cavendish, becoming one of the most influential experimental physicists of his time.
 
No. It blows the pants off of any reasonably sized anechoic chamber. NFS produces reflection free response down to 10 Hz. I don't know of any anechoic chamber that is so below 80 Hz. The advantage of anechoic chamber is speed. Not fidelity.


There is nothing "pseudo" about it. You are confusing gated, merged near-field measurements with what Klippel NFS performs. Klippel NFS uses power of computing to solve physics that would call for massive anechoic chambers. New solution to an old problem.
The near field measurement can only approximate the far field. You need to get into the far field to have a true far field measurement.
When one is closer then it gets to be an approximation.
Albeit usually a close approximation

Anechoic chambers have that problem unless they are massive and you are measuring very far. And if they are doing that, then you have to worry about temperature gradients causing phase errors.
There is routinely something 10x (or maybe 20x or 100x) lambda as the near field limit.
It may be more of an RF thing, or both RF and audio.

Learn the topic first before writing such incorrect claims.
The main thing “incorrect” is that we are talking by each other.


Let’s rewind back to the reverberations in the room and the direct sound.
Obviously people and Klippel machines both can work out the direct sound from the reflections.
We may not disagree so much here?
 
The near field measurement can only approximate the far field. You need to get into the far field to have a true far field measurement.
Only for simple measurements, not with computational holography performed by Klippel NFS. You really don't understand the technology.

When one is closer then it gets to be an approximation.
Which is the case in anechoic chambers where you cannot get far enough from the speaker to be in true far field. CEA/CTA-2034 acknowledges this issue and still stipulates measurement at just 2 meter:

"Measurement Distance Ideally measurements should be made in the far field of the DUT and, for reasons of standardization the sensitivity should be referenced to a distance of 1 m. The far field for large diaphragm loudspeakers can be several meters away and listeners may sit in the near field of these loudspeakers. For typical loudspeakers the far field begins about 2 m from the DUT, and the typical listening distance is closer to 3 m from the DUT. By taking many measurements and displaying the results as spatial averages useful data can be gathered within the near field. Therefore measurements shall be made at 2 m from the DUT and the data shall be reported as the equivalent sound pressure level (SPL) at 1 m, which is a convenient 6 dB higher than the SPL at 2 m."

The main thing “incorrect” is that we are talking by each other.
Not at all. You have no concept of signal processing being applied to solve two problems that NFS masterfully handles:

1. Using near-field measurements to compute far field response. Huge benefit here is increased signal to noise ratio, obviating the need for an ultra quiet measurement space (which is independent of whether the room is anechoic or not).

2. Using field separation to remove room reflections and generating truly anechoic response. An anechoic chamber relies on filtering those reflections but the wedges are not nearly large enough to absorb the massive wavelength at frequencies at true bass frequencies.

Both of these require computation power which did not exist in decades past. But now that we have it, we are able to generate complete sound field of speaker down to 1 degree resolution and at any distance. Such a result is practically impossible with an anechoic chamber and a microphone array.

Don't go putting it down as "garage" measurements and such nonsense you post. This is a $100,000 instrument meant for serious measurement work.
 
Is there a way to measure large speakers (like Utopia Grande, Wilson Wamm) properly? Is there a way to measure large electrostatic panels properly?
 
Is there a way to measure large speakers (like Utopia Grande, Wilson Wamm) properly? Is there a way to measure large electrostatic panels properly?
Magico uses a crane along with Klippel.

1754372599288.jpeg



I'm really curious about Genelec's set-up though, can't see the Klippel logo at 8381A charts.
 
Who is receiving sound "that is phase and time aligned"

If you are talking about direct sound solely, it means you are talking inner phase relations or group delay (that is how I understand the comment). Phase and delay differences need at least two reference points. And if we look at what is audible under lab conditions regarding group delay and which phase shifts are caused by existing gear, my guess would be that very low, artificial bass events (like bassdrum sounds or techno beats) with a very distinct time relation to their harmonics (the ´click´ part in the sound) are the most likely to show audible differences.

humans have the cognitive ability to separate direct sound from summed-with-reflections sound, and will perceive the sound as erroneous if the direct sound from the speakers is not well-balanced, even if the summed sound is not badly balanced. Therefore, getting the direct (anechoic) speaker response to sound well-balanced is important.

While I fully agree with Dr. Toole´s claim, I do not think it is sufficient to have only the direct sound balanced. Once our brain has a tonal reference (the direct sound) of the timbre, it is pretty likely to recognize spectral deviations in the early reflections as well as in the diffuse, later reverb. That's why we can easily differentiate binaural recordings from a glass hall vs. a wood-paneled concert hall. I have been in a diffuse field simulator creating such reverb artificially, and it is pretty astonishing what our brain can understanding about the room´s properties just by changing the initial delay and the tonality.

humans very strongly prefer the sound of a classical music performance in a hall with good acoustics more than outdoors.

Which non-amplified classical performances have you attended under free-field conditions? I have, quite a few, including operas and oratorio performances. Cannot say that people prefer halls necessarily, it is just very different, less loud, timbrally thin and close and distant at the same time. Instruments and voices heavy on middle frequencies, depending on early reflections, seem to have the maximum of problem with this, during general rehearsals I noticed first and foremost contraltos, tenors, horns and clarinets struggling. If is impossible to do a recording of this btw as it would not tell us anything about the room.

I want that acoustic energy to exactly match the electrical signal given to the speaker.

The transducer properties will render this impossible. Turning a signal into soundwaves, it not a mathematical process. You can only approximate the electrical signal and choose which alterations are acceptable.

we never had the ability to make such phase and time aligned technically excellent speakers before. Muti-way active DSP has truly changed what speakers are capable of.

Phase and time alignment at one single point does not necessarily help, it has to be even over a greater window both for direct sound and early reflections.

A few speakers can come pretty close to that, if they employ DSP x-over, FIR filters with a linear-phase mode, and sufficiently large drivers/low crossover points. They are around for something like 20 years. Encourage everyone to listen to such a unit, if possible switching to minimum phase mode. Differences are not huge, most obvious in the lower bass with artificial sounds like EDM. Interestingly these bass differences survive even horrendous room-induced alterations in a listening test, according to my experience.

a) the direct sound should be time-coherent north of about 500 Hz in order for the overtones to arrive simultaneously, resulting in these peaks;

In theory I agree, but in practice the necessary coherence above 500Hz it not as difficult to achieve. At 500Hz, we are not very sensitive to time issues, even if they pose an interaural difference for which our brain is more sensitive. Try to localize a male singer singing a long ´O´ or ´A´ without consonants! Pretty difficult, we are used to localize them by bright vowels or brilliance-heavy consonants.

(Elsewhere Griesinger mentions 700 Hz and 1000 Hz as being the lower end of the frequency region that matters most to the ears, giving me the impression that the octave between 500 Hz and 1 kHz is a somewhat fuzzy transition region.)

Absolutely correct regarding audibility of time differences. This band (400-800Hz or 500-1K) is, on the other hand, very important for tonality, as our brain seems to be pretty sensitive to level differences between overtones.

Group delay is even more likely to be inaudible below 1K, and above that, a loudspeaker designer needs to do pretty strange things to cause linear distortion in the range of milliseconds.
 
Both of these require computation power which did not exist in decades past. But now that we have it, we are able to generate complete sound field of speaker down to 1 degree resolution and at any distance. Such a result is practically impossible with an anechoic chamber and a microphone array.

Yes and no. Having had the possibility to compare a vast number of truly anechoic measurements with Klippel NFS ones, I have to say that NFS measurements of every speaker I have seen so far that employs drivers which are not foo far away from each other (within the measuring distance) and do not rely on broadband cancellations effects are surprisingly accurate. I see no reason to mistrust the Klippel algorithm within its range.

That said, there seem to be limitations in case you have several drivers very far from each other (such as sound reinforcement linesources) and units with a lot of cancellation effects (such as bending wave transducers, large planars, cardioids and dipoles). You can indirectly verify this by checking if the calculated directivity index is plausible. If this ones falls significantly below 0dB in a frequency band like lower bass which is radiated undoubtedly in omnidirectional manner with this particular model, well, you have evidence of limitations.

can't see the Klippel logo at 8381A charts.

Pretty sure they are using AudioPrecision, the graphs look familiar to me.
 
Pretty sure they are using AudioPrecision, the graphs look familiar to me.
For speaker measurements?
First time I hear that. How does the physical set up looks like?
 
For speaker measurements?
First time I hear that. How does the physical set up looks like?

Anechoic chamber + microphone +
AP2.jpg


Pretty much of a legacy system, don't know if they use this generation or a later one.
 
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