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

other very important consideration about the 20KHZ limits of audibility :

«One of the major problems is that it is fundamentally impossible to determine the
requirements for sound reproduction systems by sound reproduction systems: when something is “inaudible” is this because of the limitation of human hearing or because of the limitation of the sound reproduction system (including the microphone(s), sound recording and storage system)?
By designing a sound reproduction system, you have to start somewhere and I have been told numerous times that the 20 kHz limit is based on the Fletcher�Munson curves. Apart from that, although I have deep respect for what people achieved 60 years ago, I seriously doubt that the equipment they had available in those days is superior to human hearing and any conclusions drawn from their work should be critically
examined with our current knowledge,
which, however, still leads to conflicting results. So far, I have never heard a sound reproduction system which comes even close to the live performance of a symphony orchestra. So there is still a lot of work to be done and we need deeper understanding of the workings of human hearing. In that perspective, I find the
historic background of the 20 kHz limit less interesting; more interesting is the question whether we need an extended frequency response in order to bridge the gap with the symphony orchestra as this 20 kHz number has penetrated the whole audio business. Just look at the specifications of the different components from microphones to recording equipment to tweeters»

 

Hans Van Maanen Linear audio vol.5

 
 

 

 

Now to go further read this:

 

«The discussion on the perceived quality of audio systems often lacks
objective criteria. This is partly due to the subjective experience of the
ill-defined property "quality", covering many aspects, partly to the lack
of understanding of all the properties that influence the perceived
quality. The latter is not synonymous with the technical quality of a
system to begin with.
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. The basic idea behind this is
the Fourier analysis of sounds, in which any sound wave, no matter how
complicated, can be decomposed into an infinite series of sine and cosine
waves of different frequencies, starting at zero and "ending" at infinity.
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.»
 
 
 

 

 

Then Amir is a seller of his limited set of tools , his stance on tube amplifier made no sense in acoustics, and his interpretation of the results of his Fourier tools are acoustically meaningless because human hearing dont work as Amir want it to do to sell his marketing measuring  site ...

Van Maanen is a scientist known worldwide in audio .

Amir is not...By far.... Even with 2 million visitors...

Science is not made in a marketing site of audio reviews...

 

Now read this attentively and you will learn why Hans Van Maanen is not in the ASR team but in science :

 

«The theory of Fourier analysis yields that the inverse Fourier transform of

the complex valued transfer function of any filter, and thus also of our
idealised audio system, equals the Dirac delta function response of the
system in time domain. Note that the impulse response thus tells us more
than the amplitude response of a system, because it contains information
about the amplitude response at ALL frequencies (not only those between 20
Hz and 20 kHz) and about its phase response, albeit in an implicit way.
 
.........................................
 
In other words, any audio system has the tendency to "smear out" the signal
both in amplitude and in time. These effects could reduce subjective
experiences like the "definition" and "transparency" of the perceived
sound. This smearing will always be a degradation of the original sound and
we will try to study its influence on the perceived sound.
........................................................
 
The temporal decay of high-end analog audio systems is higher than the

decay of digital systems in their present version and consequently the

temporal "smearing" of the formers is less.
.............................................
 
The superior sound quality of moving coil cartridges over moving magnet
ones is at least partly due to the extended frequency response and higher
temporal decay. Moving magnet cartridges with extended frequency responses
approach the perceived quality of the moving coil cartridges, especially
those which produce a higher output signal (and thus generally speaking
have a lower mechanical resonance frequency). Compensation of the
mechanisms that create the low temporal decay of moving magnet elements
leads to significant improvement of their perceived quality (ref. 1, 2).
One of the better ways to compare analog and digital systems is by lis-
tening to a good copy of an analog recording on disc and the CD made of the
same master tape. If the digital re-processing would not audibly effect the
signal, no difference would be perceivable. Yet, on a high-end audio
system, using e.g. electrostatic loudspeakers for the midrange and high
frequencies, the transparency and clarity of the analog version (half-speed
master copies) invariably showed to be better.
Comparing loudspeaker systems is one of the most difficult and tricky
aspects of audio. Yet, generally speaking, the loudspeakers sounding best
are those with the highest temporal decay. To mention some examples:
electrostatics, ribbon tweeters and last-but-not-least ionophones. Also,
loudspeakers that show a high temporal decay in Wigner distributions
generally sound best.
..........................................................................
 
The temporal decay seems to be a useful "handle" to get grip on the
temporal behaviour of audio systems and to make a semi-quantitative
comparison. It is an excerpt of the impulse response of a system, which
tells more about a system than its frequency response between 20 Hz and 20
kHz.
High-end audio systems often sound better with analog recordings than with
digital ones. This is at first surprising because of the very high quality
specifications of digital systems. But the temporal decay is one of the few
points at which analog systems beat their digital counterparts and it is
thus a clear hint of its importance.
The behaviour of the amplitude and phase characteristic of an audio system
above 20 kHz. is of importance to its temporal decay and can thus be of
influence on its perceived quality.»
 
 
 
 

 

 

 It is useless to argue with Amir about one piece of gear and contradict him  about his opinion of this piece of gear...

This is a dialiogue between deafs with NO fundamental  ARGUMENTS in acoustics...

Van Maanen speak about audio and acoustics...

That is my argument and it contradict all Amir mantra....Which one is serious?

"golden ears" Amir spotting digital artefacts to sell his measuring ideology out of any hearing real knowledge  or physicist and acoustician Van Maanen ?😊