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

not only that you distorted the matter saying your sinad tool is not a Fourier tool. This is an half truth. why ?

It is the full truth.  Fourier transform takes a time domain signal and converts to fundamental sine waves that created it.  This is a proven mathematical relationship.  Just like Pythagorean formula.  It is not subject to debate.  And  no experiment whatsoever has disproven it.  Again, it is a mathematical proof ("theorem").

it is useless arguing with you...

the context of interpretation of all designed gear and all tools is the Fourier context...

it is evident that your voltmeter or your sinad dont need Fourier transform as a tool  as such to be used  but interpretating the results will be in the Fourier context guess why ?

 hearing theory is done in the Fourier context...

you deliberately distorted my posts context : hearing theory and the Fourier context for the design of gear...

 

The research you put forward says that our hearing system due to its non-linearities, doesn’t follow this relationship. That when we trade off timing resolution vs frequency, they don’t follow a 1:1 relationship. But this has no bearing whatsoever on audio measurements!

Another distortion about Van Maanen and my posts :

It is evident for anybody that your audio measurements are aimed at the gear specs verification!

This is what i claimed  also and it is why i explained with 6 articles above that because the brain work in his own time domain and in a non linear way any designer must think about the conditions necessary to apply the Fourier theory BEFORE designing a piece of gear...And we dont have a complete and perfect  hearing theory , and what is revealed in the articles above is the ears/brain work in a way we do not understand yet to extract acoustic information...

This immediately imply that your gear measures cannot be translated in direct prediction about sound quality perception... As you falsely suggest to all ...

you are really a marketting dude not a scientist at all... you prove it to all here with your distortion of facts...

 

At 10 KHz, our hearing's frequency discrimination is as poor as 1000 Hz! 

All tools in audio directly or indirectly use Fourier mathematics as direct tool or  as the only context of interpretation.

No, no, no. Some of the measurements I perform have been around for nearly a century!  Way before we have had any audio analyzer had any computing ability to produce fourier transform.  You can go on ebay and buy analog THD+N analyzers such as this:

«The, never mentioned, assumption is that the frequency components above the
hearing limit, usually taken at 20 kHz, do not influence the perceived
sound in any way.

Although this seems a reasonable assumption at first, it is not as
straightforward as one would think. Two aspects play an important role: the
first is that Fourier analysis only holds for linear systems and if there
is one transducer which is non-linear, it is the human ear. In non-linear

systems frequencies not present in the original signal can be generated
and/or other frequencies can acquire more power than in the original sig-
nal.
This can easily be demonstrated using a 3 kHz sine wave with 5 periods
on and 5 periods off. Although Fourier analysis tells that 300 Hz is only a
weak component in this signal, it is the strongest one hears. As 300 Hz
corresponds to the envelope of the signal it is not surprising using the
non-linear properties of our ears. 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.»
 

you make a sophism here...

You use a temporary conclusion about our set of hearing measures as we know it now  and the gear design specs which you measure again and equate them  WHICH IS A FALSE EQUATION,  and  use this measures to PREDICT sound qualities..

Sound qualities suppose a listener...

A room....

Complementary piece of gear...

Then a tube amplifier cannot be a bad sound qualities  only because you decide that your measures set will replace hearing theory and even  a specific listener  biases...

You are not an audio designer  proposing a new amp or new speakers better designed to suit human hearing as a TOP  designer understand them, as Van Maanen for exemple, you are a marketer of a methodology to verify gear specs thats all ...

but you claim to be more ...

 

 

 

@mapman  +1

Speaking as a professional engineer now for a bit over 40 years, I agree with you.