Some thoughts on ASR and the reviews


I’ve briefly taken a look at some online reviews for budget Tekton speakers from ASR and Youtube. Both are based on Klippel quasi-anechoic measurements to achieve "in-room" simulations.

As an amateur speaker designer, and lover of graphs and data I have some thoughts. I mostly hope this helps the entire A’gon community get a little more perspective into how a speaker builder would think about the data.

Of course, I’ve only skimmed the data I’ve seen, I’m no expert, and have no eyes or ears on actual Tekton speakers. Please take this as purely an academic exercise based on limited and incomplete knowledge.

1. Speaker pricing.

One ASR review spends an amazing amount of time and effort analyzing the ~$800 US Tekton M-Lore. That price compares very favorably with a full Seas A26 kit from Madisound, around $1,700. I mean, not sure these inexpensive speakers deserve quite the nit-picking done here.

2. Measuring mid-woofers is hard.

The standard practice for analyzing speakers is called "quasi-anechoic." That is, we pretend to do so in a room free of reflections or boundaries. You do this with very close measurements (within 1/2") of the components, blended together. There are a couple of ways this can be incomplete though.

a - Midwoofers measure much worse this way than in a truly anechoic room. The 7" Scanspeak Revelators are good examples of this. The close mic response is deceptively bad but the 1m in-room measurements smooth out a lot of problems. If you took the close-mic measurements (as seen in the spec sheet) as correct you’d make the wrong crossover.

b - Baffle step - As popularized and researched by the late, great Jeff Bagby, the effects of the baffle on the output need to be included in any whole speaker/room simulation, which of course also means the speaker should have this built in when it is not a near-wall speaker. I don’t know enough about the Klippel simulation, but if this is not included you’ll get a bass-lite expereinced compared to real life. The effects of baffle compensation is to have more bass, but an overall lower sensitivity rating.

For both of those reasons, an actual in-room measurement is critical to assessing actual speaker behavior. We may not all have the same room, but this is a great way to see the actual mid-woofer response as well as the effects of any baffle step compensation.

Looking at the quasi anechoic measurements done by ASR and Erin it _seems_ that these speakers are not compensated, which may be OK if close-wall placement is expected.

In either event, you really want to see the actual in-room response, not just the simulated response before passing judgement. If I had to critique based strictly on the measurements and simulations, I’d 100% wonder if a better design wouldn’t be to trade sensitivity for more bass, and the in-room response would tell me that.

3. Crossover point and dispersion

One of the most important choices a speaker designer has is picking the -3 or -6 dB point for the high and low pass filters. A lot of things have to be balanced and traded off, including cost of crossover parts.

Both of the reviews, above, seem to imply a crossover point that is too high for a smooth transition from the woofer to the tweeters. No speaker can avoid rolling off the treble as you go off-axis, but the best at this do so very evenly. This gives the best off-axis performance and offers up great imaging and wide sweet spots. You’d think this was a budget speaker problem, but it is not. Look at reviews for B&W’s D series speakers, and many Focal models as examples of expensive, well received speakers that don’t excel at this.

Speakers which DO typically excel here include Revel and Magico. This is by no means a story that you should buy Revel because B&W sucks, at all. Buy what you like. I’m just pointing out that this limited dispersion problem is not at all unique to Tekton. And in fact many other Tekton speakers don’t suffer this particular set of challenges.

In the case of the M-Lore, the tweeter has really amazingly good dynamic range. If I was the designer I’d definitely want to ask if I could lower the crossover 1 kHz, which would give up a little power handling but improve the off-axis response.  One big reason not to is crossover costs.  I may have to add more parts to flatten the tweeter response well enough to extend it's useful range.  In other words, a higher crossover point may hide tweeter deficiencies.  Again, Tekton is NOT alone if they did this calculus.

I’ve probably made a lot of omissions here, but I hope this helps readers think about speaker performance and costs in a more complete manner. The listening tests always matter more than the measurements, so finding reviewers with trustworthy ears is really more important than taste-makers who let the tools, which may not be properly used, judge the experience.

erik_squires

@mahgister Hey magister! I don't hate Amir, I just don't like his style and motives. You're not remotely qualified to assess what I say and mean, and for that matter, never have been.

You, if anyone, are not polite. You explode with anger all the time and when confronted with your behavior, you apologize. You've done that many, many times. 

If anyone should imitate Amir, it is you, but you don't have that ability, as you've shown time over time all these years. The patience he's shown you is more than anyone I know would exhibit.

As for infestation of threads, that seems to be your forte. You've done it for years. Practically everything you've said in this thread, you've brought up before to the point of boring the heck out of members. You go off on your tangents demanding that others must respond and when one or two do respond, you claim vindication and insult other members when they complain of your tactic of highjacking a thread. 

You post multiple times in a row but no one answers and it spoils the thread and intention of those who want to  participate. Like others have already said, they (we) just pass over what you write hoping you lose interest (at least I do).

If you're of the mind, why don't you go over to ASR and start posting there and let us know how that goes.

By the way, if you really think members here are "gangstalking" you, reflect for a moment as to why and you'll discover it is because of you and your manner.

All the best,
Nonoise

I think mahgister is great. I mean the first 100 posts he made on this thread were a little boring but I think he's going to hit his stride on the next 100.

you miss completely the argument...

It is not about measurement here... He explained why it is very difficult to measure this without very serious research... You dismissed it without even getting the main point BECAUSE IT SUIT YOU..😊

It is about measurement.  This is from the summary right at the start:

"SUMMARY. In the discussion about the perceived quality of sound systems the temporal aspect is often neglected or its importance underestimated. In this paper we propose a semi-quantitative property of systems to compare these, taking the temporal behaviour into account. We have tried to find a simple, easily to find and to interpret parameter which by no means will be the final answer to the problems encountered in audio, but can help to improve the comparison of systems in a more objective way and could help to direct future developments."

It can't more clear that he is proposing an objective, measured parameter.  Yet, neither he, nor you apply this to any system to measure it.  Why advocate an objective measurement when you can't or haven't computed it?

The main point is here :

What you quoted is not in this paper.  Please stay on this paper instead of jumping to other ones.  It is a difficult enough discussion to have without doing that.

Then your pretense to predict sound quality with your narrow set of measures is preposterous... 

The paper introduces a dead simple measurement of its own, which is simply met with wide bandwidth.  It completely excludes distortions and noise, two of the most important impairments in audio.  Once again from the paper:

"Disregarding non-linear distortions, the frequency response between 20 Hz and 20 kHz of a system is very often taken as a major parameter determining the quality of a sound reproduction system."

A simple impulse response is not going to tell you anything remotely akin to fidelity of the system.  This measurement has been known for decades and decades yet it is not at all applied in this application.  You want to call a a measurement "narrow" and preposterous, there is no better example than what is in the paper you reference.'

It can be concluded that frequencies
above the hearing limit can indeed generate signals that are below the
hearing limit which could thus influence the perceived sound and the
quality experienced.»

Nothing as such can be concluded unless listening test results are shown to prove it.  Tests of high resolution music which by definition has higher bandwidth and less ringing in audio domain, have failed to provide clear audible evidence.  If doubling or quadrupling the system bandwidth and hence reduction in decay time can't be shown to have value, what he is saying is in dire need of proofs, not pleadings.

All this demonstrate the complete futility to PREDICT sound quality by measuring with Fourier linear tool some aspects of the gear piece ...

There is no such statement or position in the paper.  Per above, audio system non-linearities and noise are put aside and an argument is made for a single, trivial measurement that he hasn't perform to prove anything.

We must listen...

Which neither you, nor the author have done.  Given that, the paper should be dismissed then, right?

 

 

 

 

 

 

Adding on,

All this demonstrate the complete futility to PREDICT sound quality by measuring with Fourier linear tool some aspects of the gear piece ...

As I have repeatedly explained to you in the past, many measurements I perform are devoid of any use of Fourier transform.  SINAD for example is computed using simple signal subtraction (you take out the input tone and all that is left is noise+distortion which we call SINAD).  Signal to noise ratio is just a level differential.  THD+N vs frequency is the above but at different frequencies.

We perform fourier transforms so that we can then apply psychoacosutics to the measurements.  It is not by itself as you keep claiming, is the way measurements are performed.  So please stop calling my measurements Fourier based.