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

Your thinking is wrong. All testing is done in time domain. The graphs are shown in frequency domain since it is hard for a human to tease out the noise and distortion from a waveform display in time domain. Keep in mind again that based on Fourier Theorem, time and frequency domain are interchangeable.

The input signal may be in time domain, but the analysis is done in frequency domain and the graph you show only shows steady state response. No transient information. But it’s in the transient that truely show the performance with different loads. You also need to test with different loads as well. May be at 2ohm, 4ohm to show the current capability of the amp.

 If the P12 does what PS audio said, I think you will see it in the transient response.

 

That is still "steady state" by your definition since there is no discontinuity in the signal.

No the square wave will test for transient condition. It is like a step response to test how the amp can deliver the current. You can make the square wave period long enough so wait out any ringing or steady state settling time.

That aside, you can’t have square wave as a valid audio signal since it has infinite bandwidth (Fourier Theorem).

You don’t have to worry about that. A real world square wave will have finite rise and fall time. Music is not a sine wave or multiple of frequency sweep either but you use it for your testing.

 

As for measuring output impedance, what you did might not be adequate and I think you might have taken the short cut. In order to really test for it you have to:

1. Measure the PS12 output with no load.

2. Use a spectrum analyzer to measure the amplitude at several frequency since the output is not a pure sine wave.

3. Now connect the output to different load such as 2ohm, 4ohm, .... or more.

4. Then measure again the output spectrum again with the loads connected. The output amplitude now will be slightly lower due to the finite output impedance of the unit.

5. With those information, you can use a simple equation to calculate for the output impedance.

That’s the only accurate way to measure. It’s a little bit more involvement and I am not sure you’re prepared or have the setup to do that.

 

@amir_asr  

at 30% efficiencies almost 20kw input. At 220v thats almost 90 amps. And the "power meter" is the most useless meter I have ever seen, its a joke. Hardware for people with way too much money and no brains.

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 This is typically professionalism at your site that goes unchallenged by you or your moderators.

"We all focus on performance."
Wrong. Practically every reviewer looks at and comments on the build quality and the components used.

" I don’t know why you all don’t let this concept sink in by spending just a few minutes learning about the topic instead of relying on your lay intuition."Why should anyone listen to you? You have nothing important to add.

"If you act this unprofessional" I am unprofessional? What a joke. You are arrogant with nothing to be arrogant about, you are rude and overbearing. It is no wonder you are so disliked and have been thrown off so many fora. Go back to your cult an your minions.

 

1. Measure the PS12 output with no load.

2. Use a spectrum analyzer to measure the amplitude at several frequency since the output is not a pure sine wave.

What on earth are talking about?  Spectrum analyzer shows the spectrum, not time domain amplitude.  Regardless, that was done in the P12 review:

As show, the P12 actually increases AC noise, not decrease it.  It does so because it is not a true regenerator.

3. Now connect the output to different load such as 2ohm, 4ohm, .... or more.

Looks like you have forgotten about ohm's law.  P12 generates 120 volt RMS AC.  At 2 ohm, you would be asking it to spit out whopping 60 amps!  The could cause it to be damaged at worst, or shut down at best.  That is on top of needing a high voltage dummy load that could dissipate over 7 kilowatts of power!  You seem to be confusing how you measure the output impedance of audio amplifier rather than a high voltage AC generator.

5. With those information, you can use a simple equation to calculate for the output impedance.

Which is what I do when measuring output impedance of headphone amplifiers.  But per above, you are dealing with a completely different beast here with a high voltage AC source.

None of this is necessary anyway as output impedance of AC source only has a loose relationship to what comes out of your audio device as the latter has its own power supply and capacitor bank to provide power for transients.  Measuring an amplifier power is a much better and more clear verdict of whether the P12 is able to do its job transparently or cause performance to be lost.  The latter was clearly shown in my measurements.

I did use an impedance meter to measure  anyway so not sure why you are continuing on.  Here is my AC source impedance:

 

And here it is going through P12:

 

As noted, P12's output impedance is nearly 10 times worse despite company claims of it being better!

No the square wave will test for transient condition. It is like a step response to test how the amp can deliver the current. 

Square wave testing can generate highly misleading information as the signal itself may never be representable with music signals.  To wit, digital audio at 44.1 kHz won't have any components above 22.05 kHz.  Feed it a 10 kHz square wave and what comes out is a pure sine wave!  I have a video on that: