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ernestcarl

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I've had these two speakers for a very long time, but I've never really gone into a detailed comparison between these two side-by-side before.

This exercise is going to be divided into sections, so just skip ahead to the areas of interest to you.

Also, I'm going to be focusing on the Sceptres primarily as the Neumann are already discussed ad nauseam in ASR and elsewhere. :p

EQUALIZING THE SCEPTRES:

I bought a store demo pair of the S8 studio monitors at a ($200?) discount and found out that they're responses were not matching as well as one would have hoped. It's difficult to determine exactly the origin of this issue now, but my guess is that it is likely coming from driver factory variances where tolerances are simply not kept as high as with other more expensive models. Bare in mind, these drivers also seem to be a bit more complicated to build. Fortunately, the differences are not so severe that it cannot be remedied by a 'simple' A/B trace arithmetic function within REW.

The theory is this: you take two measurements from each transducer and equalize the differences between them. The traces need to be accurate enough and taken from nearfield data in order to do this. Ideally you should refer to to multiple directional curves or take an average of relevant curves from each channel. Confirm the effectiveness of the correction by measuring your speakers response in the MLP afterwards, as well as other seats/angles/distances. If you are short in time, you can also just use single on-axis traces, but this may not be as accurate.

1608638444197.png


1608638509597.png


1608638528057.gif


REW may have difficulty generating the exact PEQ corrections you want so feel free making manual adjustments as needed.

Once you've ascertained that your speakers are "pair-matched", you can then go about the more complicated task of speaker EQ "correction" or optimization.

This topic can be rather convoluted (and I'm certainly no expert) so this is only going to be a simplification. You would need to refer to the directional curves of your speakers or to the specific traces found in the spinorama data. The axial response, LW, PIR etc. If you don't have that, then you'll need to generate your own directivity curves for which you will refer to during equalization.

*Seeing how the EQ works on my 'pseudo-listening window' curve:
1608655181606.png


The Sceptres are coaxial in design so it's relatively easy to obtain horizontal and vertical traces required for this exercise. I don't have a proper measurement rig and so I just simply used a small angled ruler I had already readily on hand for this purpose. Of course, you will need a calibrated measurement mic, and a stand to hold this as well.

1608639221898.jpeg


The ff. are the directivity curves I measured with predictive corrections applied:

LEFT SPEAKER vs RIGHT SPEAKER (post pair-matching EQ and speaker 'optimizing' EQ)
Separated for clarity
1608639384097.gif


LEFT SPEAKER vs RIGHT SPEAKER (post pair-matching EQ and speaker 'optimizing' EQ)
1608639395567.gif


There's a bit of a suck-out around 2kHz and a small hump at 3kHz -- the latter adds to that in-your-face or forward vocal character of the monitors.

Now notice how many of the curves bunch-up closely between 5-10kHz, while I do like how the presence and brightness region of these speakers remains very well sustained off-axis, I'd just like to see that area little bit shelved down to tame the excesses I hear with sibilant and 'splashy' sounds in the HF which can get a little bit annoying over time.

Below is my most current (simplified) general speaker EQ correction to date:
1608639951949.png


Again, you have to confirm that the corrections work as intended. I used single-point as well as MMM technique at the MLP, including both the further left and right side of my listening couch.

1608641044302.png


It's just a left+right sum in the above graphic, but the per channel outcome yields very similar curve shapes just as well. Consistency at all seats!

1608639970833.png



So what the hell is the point with all this equalizing before the listening test even anyhow!? Well, I really wanted their responses to be both on par and as neutral as can be to my own ears so we can rule out any gross axial response differences as the cause of the variances in what I will be later hearing. Careful speaker EQ correction together with my pair-matching correction helps to 'optimize' the final response to better match the Neumann KH120s (applied -1dB HF shelving filter) which will prove very useful during my side-by-side mono listening session.

BTW, the correction we are using does not drastically alter the original curve. No, we don't want to apply any overly excessive correction.

1608641295452.png



SIDE-BY-SIDE LISTENING TEST

My listening space is quite small, but I did try to clear out the area so that the speakers are reasonably away from any wall boundaries.

1608641346150.jpeg



1608641863047.gif



Some very basic tech specifications:


PRESONUS SCEPTRE S8
Frequency Response (-3 dB) : 46 Hz - 20 kHz
Frequency Response (-10 dB) : 38 Hz - 23 kHz
Crossover Frequency : 2.4 kHz
LF Amplifier Power : 90W class D
HF Amplifier Power : 90W class D
Maximum Continuous SPL (with music, typical conditions?) : 105 dB (from archived 'brochure') but realistically you want to keep the ‘continuous SPL’ at 100 dB due to increasing harmonic distortion
Maximum Short-Term SPL : 110 dB (my guesstimate - but it should be frequency dependent too)
Peak SPL (transients) : 116 dB (from official manual)
LF Driver : 8" glass-reinforced paper
HF Driver : 1" horn-loaded titanium diaphragm compression driver
Horn Beamwidth Control Limits : 110° x 90° more sharply angled and no cont. transition to baffle (to bypass intermodulation distortion issue plaguing many other coax designs as it "keeps keeps HF energy off of the woofer cone.")


NEUMANN KH120
Frequency Response (-3 dB) : 52 Hz - 21 kHz
Frequency Response (-10 dB) : 46 Hz - 24 kHz
Crossover Frequency : 2.0 kHz
LF Amplifier Power : 80W peak class AB
HF Amplifier Power : 80W peak class AB
Maximum Continuous SPL (with music, typical conditions) : ~95 dB (my guesstimate OR approximation from their tech specs)
Bass capability: Max. SPL half space 3% THD 1m : 104.8 dB (official tech spec)
Peak SPL (transients) : 111.1 dB (very much frequency dependent, of course)
LF Driver : 5.25" sandwich cone composite material
HF Driver : 1" titanium fabric dome
Waveguide Dispersion : not so sure, but it should be wider due to very smooth waveguide seamlessly transitioning to the rest of baffle



I use the ff. manual channel mixing/routing settings to create a mono signal and switching between monitors with JRiver:

1608646286605.png


As for my own physical positioning: I was mainly seated just over a meter to a meter and a half away mid-point both monitors. I stood up as well for extended amount of time, and listened at extreme opposite angles too (both seated and standing/walking around a bunch).

I spent well over six to seven hours listening -- by the end of the long session, I was fatigued as hell and had to take a break!


The following are my notes and conclusion:

Presonus Sceptre S8
  • Speaker EQ which attenuates HF from original sound may not impress at first
  • Very clear & great clarity, although also less harsh & quite restrained in character
  • Can listen to it much longer
  • Can listen to it at a higher SPL
  • Vocals still sound forward after EQ, BUT also somewhat "rounder" or "warmer"
  • Highs, mids, and lows sound more cohesive which is very obvious when bobbing & swaying head and body up, down, left, right -- change in sound or any kind of vocal displacement is far less obvious/disturbing
  • FASTER, clearer transients, very obvious with extreme flamenco guitar track tested (also has a brightening effect)
  • Quick sliding hand movements across guitar strings sound more real -- to me this is significant! LOL
  • Cymbals, bells, and tamborine-like and other 'ringing' brass, string & percussion acoustic instruments sound nicer and are more pleasing
  • HF transient peaks do not sound squashed, compressed, or cut-off short
  • The excellent transient performance makes music sound "alive"
  • Overall quality sounds "rounder" / "softer" / less aggressive -- only partially explained by HF shelving and dips in response
  • Much preferred for everyday casual listening & for "finished" tracks
  • 'Some' deliberately added electronic distortion effects are more obvious to hear with these monitors
  • Port 'chuffing'? or more like cabinet resonance/distortion is audible with extreme sub bass-y electronic tracks -- but only at extreme volume levels where the KH120's can't even reach

Neumann KH120
  • Tonality for voices is preferred with these
  • Clearer it seems, or at least more deliberately obvious accentuation of details
  • Esp. high treble frequencies which is also much more extended
  • Easier to separate elements of a mix
  • HF extension and treble detail will very much highlight harshness (EXCEPT for harshness that comes from really fast transients which this monitor isn't able to resolve nearly as well)
  • More aggressive sounding (i.e. less "soft" / "restrained") in many of the tracks listened to
  • My ears tire out quicker/sooner with these monitors
  • YET, the monitors sometimes glosses over/misses deliberately added distortion effects in electronic tracks
  • Nevertheless, I probably will choose this first for critical listening / or when doing detailed dissection of sound tracks -- IF I'm not particularly fatigued/tired as it sounds perceptually clearer to me to more of the tracks being listened to anyway
  • Sitting exaggeratedly straight up or slouching down one can notice sound displacement between HF & mid drivers.
  • Bobbing & swaying head and body up, down, left, right -- more apparent change in sound, vocal displacement or some type of shadowing effect can be heard -- actually I've had this problem with them for a very long time as I use a sit-stand desk and I change positions heights a lot. For example, wearing shoes changes the the sound from what I have been used to listening from the several past days, and I find this disturbing enough to finally take-off what I'm wearing or make re-adjustments to the height of my speaker stands. BTW, minute changes to the horizontal angle placement makes bigger apparent changes to the sound (again, my past experience). This is magnified with my sometimes very nearfield desk setup use-case scenario (I can pull both monitors and keyboard closer or nearer as necessary).
  • Sounds somewhat "smaller", maybe even just a little less coherent (again, shifting of location source or sound character from driver displacement is magnified with any gross physical movement, esp. the closer you are from the speakers)
  • SLOWER transients, very obvious with extreme flamenco guitar track tested (which also has a darkening/dulling/de-brightening effect)
  • HF transient peaks sound as though they're being cut-off short or muted -- sadly, this effect is not subtle when switching monitors
  • And so it glosses over excessive harshness from fast transients in certain tracks -- BUT, of which can really only be heard clearly on the S8s (believe me, I wish the that observation was not so obviously the case since the frequency response of the KH120 is so incredibly flat and smooth.
  • Quite impressive and punchy bass for its size! I think many people will be happy and plenty satisfied even without the presence of a sub.
  • Port 'chuffing'? or distortion audible with extreme sub bass-y electronic tracks -- I think it relies too much in it's natural acoustic roll-off which can make it sound more boomy/distorted -- if not sometimes artificially extended than the S8 with its very steep 10th-order roll-off by comparison. I do feel like this may hype the bass quite a bit more than the S8s

CONCLUSION AFTER SIDE-BY-SIDE LISTENING SESSION:

It's ironic, but I actually feel a little bit sad because of how "good"/"realistic" and... surprisingly bright and extended HF transients sound on the (BTW, cheaper) S8 monitors. Having just a very flat and smooth frequency response and impressive spinorama graphs unfortunately is not enough reason for me to choose another speaker when it's finally time to upgrade or replace these. As I, and many others have frequently remarked, there are always, ALWAYS other considerations... that is, if you're the meticulous type when it comes to these esoteric speaker design matters. There are a number of times where I may have unfairly judged the Sceptres due to their obvious deficiency in frequency response evenness here and there. Not to mention the driver to driver consistency (frequency response matching) seems nowhere near as good it should be -- even cheaper speakers like the JBL LSRs or Mackie MR series (the few samples of speakers I've measured) seem to perform better in that regard, at least.

However, don't be so alarmed at the lobing behaviour... it's actually much more benign than you might think. This is greatly offset by the superior performance for off-axis listening vs a conventional speaker even with completely zero horizontal lobing -- though most of these conventional designs are also not too great in their vertical coherency anyway. The unevenness in the frequency response and (possible) pair-matching inconsistency can also be improved upon greatly if one is willing to make the effort -- that is, if you're comfortable working with a measurement microphone, REW, and with DSP correction. Overall, despite the listed disadvantages, I think it may still well be worth effort to try/audition these out. People like to move -- C'mon! We even shift our bodies all of the time when sitting on the same chair or couch! Don't you ever pause for a moment to stretch your bodies, arms, and legs? The advantages of this point source behaviour in a speaker is just so obvious, esp. in nearfield listening where you might still find yourself needing to move around a lot, despite one's supposedly fixed listening position.

I've asked the question before in ASR if there is a point in applying further phase correction via something like rePhase (FIR designer, or the like) and convolution to an already time optimized speaker design -- maybe there are further refinements that can be made? Well, perhaps there's always something that can be improved on -- as the few time-domain obsessed guys here have so confidently asserted. But unlike basic modification of the frequency response, I'm somewhat more skeptical about altering what already came out of the box in what essentially are supposedly "finished systems" -- even more-so in the time-domain given I don't exactly have the tools and extensive expertise or know-how the original speaker and DSP designer(s) have. Okay, maybe I'm just a worry-wart; maybe as long as it does not deviate so much from the original response it'll be fine? Hmmmn... I dunno ... although, I would rather prefer not have any more added processing delay time if I can help it -- so there's that consideration.

It looks like the S8 is currently $100 cheaper than the KH120s in many online retail stores... but despite it what you get is a bigger 8" woofer with better dynamic-range and SPL peak capacity. Which means these monitors can handle very "LOUD" when called for -- no LEDs flashing alarmingly bright red -- Warning! Warning! Back Down the Volume Now! -- like with the smaller Neumann KH120 each time limiters are activated... I've said this before, and I'll say it again, it's a great "value" alternative to much more expensive model brands of speakers, esp. when utilized home for theater use (don't forget the off-axis advantage) -- IF! Again, IF, you're not specifically requiring "far-field" performant speakers. Hmmmn... I gather, in most use-case-scenarios, they're still going to be much more suited to small or medium-sized rooms anyhow.

@watchnerd mentioned a problem with translation... I think part of this can be explained by the fact that most speakers in the market are simply not phase-linear (mids to treble) -- and esp. at all directions! These do sound very different from most of the conventional speakers we often hear out there in the open, there's no doubt about it. But you have to hear it for yourself in person using test tracks that particularly highlight these monitors' design strengths. You also need to compare it directly side-by-side to a conventional state-of-the-art two-way or three-way design to find out what are the details it can resolve more clearly that more conventional designs simply cannot -- most obvious of which are very fast, very dense transients in the HF. It almost makes you wonder if there's something actually broken or wrong with the Neumann KH120s. It's a bit of an annoyance because I also own and use the KH120s almost everyday for the past, what? Almost ten years, now?

*I mentioned a specific test track sample with lots of dense transients and this is the one: Foc by Rodrigo y Gabriela

JRiver's audio analysis had showed the ff. report:
1608647026570.png


Confirmation with another VST analyzer:
1608647079030.png

This is before any DSP and volume level processing


1608647116782.png



More subsequent graphs, and measurements to follow.


In the meantime, the ff. are some informational details provided by Presonus for the Sceptres (which I've toned down quite a bit -- the language was full of marketing hype and filler words). :rolleyes:

  • The human ear hears most sounds as point sources.
  • Even when we trick our brains into perceiving a stereo soundstage, each ear most easily “assembles” the stereo effect from its own point source (think headphones).
  • Benefts of coaxials are related to radiation characteristics of a point source and a consistent acoustic center.
  • Response behavior is symmetric in both the horizontal and vertical axes.
  • Frequency response at a given angle in relation to the axis is “mirrored” at the same angle in the opposite direction.
  • Conventional two- and three-way speaker systems can’t achieve this. High and low frequencies originate from different physical points. Bass and treble arrive at the ear at different times — and from slightly different axes. This subtle “smearing” cannot be corrected with digital signal processing.
  • Multi-way monitors suffer from the changing relationship between the listener and the speaker elements and cannot possibly have an optimal impulse response over a wide area.
  • Crossover transition can theoretically be made inaudible.
  • The frequency point where highs and lows cross over can result in an audible dip (or peak) in the response of the monitor.
  • A coaxial multi-way speaker is a point source, it has the potential to completely hide the crossover point. Note that we said potential. It’s no slam dunk. And this brings us to why all studio monitors aren’t coaxial:
  • Both transducers nestled together on the same axis generate signifcant problems. A horn right in the middle of the low frequency speaker cone can cause diffraction and distortion. Rearward sound from the horn can bounce off the woofer cone and arrive at listener’s ears too late, smearing the stereo image.
  • The proximity of horn and woofer can cause the two outputs to experience audible intermodulation distortion.
  • Sceptre starts out with a transducer that
    • a) solves many basic coaxial problems through its physical confguation and
    • b) is also designed so that DSP can be used to correct other issues.
  • Resulting in smooth off-axis response.
  • The high-frequency horn contributes to frequency pattern control, and keeps HF energy off of the woofer cone.
  • The woofer’s larger radiating surface works with the HF horn to improve the bottom operating-range directional control.
  • TQ uses:
    • Infnite Impulse Response (IIR) highpass, lowpass, and parametric flters, plus delay
    • Finite Impulse Response (FIR) flters implement more detailed frequency response adjustments; and precise temporal (time domain) filters
  • DSP can eliminate physical horn reflections induced resonances when they happen consistently. Knowing in advance how the loudspeaker will respond to a particular signal, it is possible to calculate a new signal that not only avoids exciting natural resonances, but also actively kills these resonances before they become audible.

There's an interview with the designer, Dave Gunness in Nathan Lively's podcast. He doesn't talk about the Sceptres which are being produced by Presonus specifically, but I found it extremely interesting. It allowed me to better my own EQ methodology using some of his ideas -- but more importantly, his own remarks about the "brightening" effect after phase linearization (at least of his own coax speaker designs) was indeed very much confirmed in my own recent listening test above. It's also where I got the idea of adding Jennifer Warnes' song Somewhere, Somebody to my critical speaker listening tracks list. Not that I should know how a Fender bass guitar actually sounds like in real life, LOL!

Here's the YT media link in case someone is interested:

WARNING: It's about an hour-long, so if you'd rather prefer to skip that one, the ff. is a transcript of some things he had said which I copied on my phone (simplified, clarified and abbreviated).


----------------------------


Nathan asks him how it's possible to have "higher gain before feedback" about his speaker designs (which are mostly used for live sound applications).

Where Dave mentions a couple of important points:

  • The response is more consistent over the entire coverage pattern.
  • Having a really flat phase response helps with stability.
  • Wobbles in phase correspond to resonances.
  • Resonances encourage feedback.
  • If your’re phase is flat and not ripply, that means you are essentially resonance-free.
  • It’s a combination of TQ (proprietary FIR based EQ tech) which flattens phase and the coaxial approach which gives you a very consistent response and it’s noticeably more stable with regard to feedback

FF. discussion more about how they "voice" their speakers:

Primarily it allows us to make all the speakers sound the same. Killing all the resonances makes the voicing of speakers much easier. Typically you are trying to overcome resonances in one speaker that doesn’t exist in another, and it’s futile to try to make those two sound the same if one has a resonance the other one does not have.

You can’t really make them sound the same. You are just mitigating problems in each of them until they both sound good. But if you switch between them, you are still going to hear differences. TQ allows us to have everything in live sound to sound very, very similar. The smallest speaker and the biggest speaker we make have very similar low frequency extension — so they actually have the same frequency response. If you are far enough away from the bigger model, they sound about the same. It’s just that one gets 14dB(?) louder than the other. They have similar character so that it’s a very seamless transition between those speakers. Having an entire line-up of speakers that sound like they’re voiced by the same person saves a tremendous amount of time during commissioning... You, of course, mitigate bass problems caused by placement and the room. But what you are left with 500Hz and up doesn’t need to be mitigated with... in the end the system has a much more consistent character as you work around the room. Listeners are less conscious as they walk around the room. They’re not gonna look up and say, “I’m gonna walk up over there because I don’t like the sound of the speaker where I’m standing by.”

The voicing is just flat through the mids. With displaced systems where they are not coaxial, it’s usually beneficial to have a dip where they have the crossover because the HP on the HF and the LP on the LF are both resonant — and they’re both sitting right next to crossovers with all kinds of resonances. These resonances call attention to themselves. If you have a 15kHz xo and you dial it flat, you are going to hear a 15kHz character that does not belong there — you are hearing the resonant character. What typically happens is eventually there’s a dip at crossover because when people are EQing by ear they are coming back to that 15kHz by ear and pulling that back down — so evetually you have a dip at xo. With coaxial systems you don’t have that issue because you don’t have the change in directivity as you’re going through the crossover. We also use extremely low Q crossovers where we have sometimes up to two octaves of overlap between the low and high frequencies in order to stabilize the polar pattern between the entire range. As a result, even without the FIR filtering it is very hard to pick out where the xo is. When somebody asks “where’s the xo?” It’s spread over two octaves... With displaced speakers you have to try to get out of one driver and into the other one as quickly as possible because if they’re on at the same time they screw up the directionality.

That is the first thing that anyone would notice — just flat throughout the xo. It sounds very different than what we are used to listening — speakers with displaced drivers xo at 15kHz or so which sounds normal. But when the artifacts associated with that go away: “Wow! This very different!”

The other aspect of the housecurve is the low end and the top end. For the low end, we do a very gradual beginning to roll-off and don’t turn and go down hard until maybe 4 or 5 dB. So it looks a little bit like a butterworth cascaded with a 6dB. That’s the curve that we found sounding most natural and 'even' in terms of notes. If you have a speaker that starts to go away at 80Hz, and you use this particular shape, the notes on the bass guitar will all have about the same weight — the melody will still be there. Whereas if you put a butterworth filter on it where there is a sharp corner, all the notes are going to sound like that corner — all the notes will have too much 80Hz and they all sound kind of the same. When you go to the 70Hz note, it’s weak. When you go back to the 80Hz note it’s strong... so the bass-line wonders around. It’s the technical response, but the reason is aesthetic. This is what makes all bass notes sound like they have equal weight.

At the top end, it will generally roll-off very, very slightly from about 1kHz up — not quite a dB per octave. At 10kHz it may be 2-3dB less than it was at 2kHz. Partly it makes the speaker sound warm and powerful. One of the cardinal rules of EQ is you don’t want a peak to be louder than something below it in frequency. If you have a 102dB peak at 2kHz but it’s running a 100dB from 500Hz to 1kHz, that 2kHz is going to stand out. If the overall level were 2dB lower so that peak matches 500Hz to 1kHz, that 2kHz area will not call attention to itself as much. This gradual decline from 1kHz up is what we find more forgiving of reflections in the space. It keeps everything warm and full.

And also, when you have very coherent high frequencies, they sound louder. People are used to listening to speakers that are nominally flat, but with displaced drivers and without TQ. When you flatten the phase it gets brighter — supressing that just a little bit makes it sound a little bit more like what we are used to... the high frequencies are clearer and more articulate not because they’re louder.

When we are tuning a speaker, there is a very definite house sound we are going for and we know how to describe the departure from that sound.

We don’t EQ by only the axial response, we EQ by looking at the whole family of off-axis curves including the axial.

If there is a bit of a rise near the pattern edge of a given frequency, there needs to be a dip so that you de-emphasize that peak in the response away from the primary axis. It makes it go a lot faster to be able to to look at the response in every direction on the computer screen while you are developing the filters. And then when you turn it on and listen to it you start out a lot closer (to your end goal) than we’re used to in the old days.

We typically do about ten curves in various places of the pattern. The way our proprietary software works is when I tweak an EQ, all ten curves morph at the same time. If I’m going after an off-axis peak to try and lower it, I can see the effect it is having in the on-axis at the same time so you don’t (unwittingly) strike a compromise between those two curves.

For people developing they’re own FIR filters, I will definitely look into multiple directions. When you’re listening there is a tempetation to sit dead straight to a speaker and that is fine if what you are making is a home hi-fi speaker or a studio monitor. But if what you are making is something that is going to address an audience, all the audience members are equally important. You have to walk through the entire pattern and you may have to compromise a little bit with the axial response in order to mitigate a problem near the pattern edge. It’s not how good you can make it in front of house, it’s how 'even' you can make the response everywhere.
 

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ernestcarl

ernestcarl

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DETAILED MEASUREMENT COMPARISON

By now, I think most of us are skeptical of the highly optimistic response measurements provided by most manufacturers. Presonus is no different in this regard:

1608656470928.png


Blue line is the measurement found in one of their single-page promotional "tech-spec" info sheets. However, do understand also that the conditions in which measurements I took were non-anechoic and made in-room, at an extremely close-range. Despite that, we do see some resemblance at least -- that way we know these manufacturers aren't lying to us outright!

Let's first start with distortion:

1608657019477.gif


As you can see, my speaker 'optimization' EQ reduces and "levels off" quite a bit of that HF distortion.

1608657034769.gif


But, by god, are my eyes deceiving me?! The much smaller Neumann is actually cleaner in many areas here and there, too. This is incredible performance coming out from such a small, compact speaker! Yes, the second harmonic goes above Amir's 50 dB threshold, but, the third harmonic is still way down, and higher order harmonics are no where near to be seen above 300Hz.

Well, the S8 ain't no slouch here either and is cleaner in much of the range between 200-900Hz. Beating the KH120s primarily when it comes to the bass and lower mids. Notice how it keeps distortion from leaking out away from the fundamental. The Sceptre's can play cleaner and more even sounding bass at louder levels too. In the past, I've measured transient peaks (around 100Hz, I believe) that reached up to 115 dB -- or just hovering around that point and the yet the monitors still sounding great. Of course, and yes, there's going to be a lot of distortion, but with the aide of a sub, you mitigate some of that significantly -- and that port noise issue I mentioned in my listening test impression also goes away. The extreme upper end of the treble response electronically clips with sine sweeps, right around 105dB (extreme nearfield). You will hear swishy re-reflected like sound effect when this happens. But these are with slow sine sweeps, so it possible that faster, more sparse transients with actual music will be able to come through fine without any obvious electronic clipping suppression effects. I certainly haven't noticed it with my occasional, extremely loud listening volume levels with blockbuster movies and bassy psytrance dub-step-like type of music.

Here's Amir's distortion measurement for the KH310 for comparison:
1608657417334.png


BTW, please take note that I'm using a cheapo UMIK-1 and have no idea how good and accurate it really is for these sorts of purposes. My own technique could also be off, but I did the best that I could given the circumstances.

In the past, I took some measurements inside the bass reflex port

1608657831521.png


That prominent second harmonic peak at 1kHz or before (it shifts with mic position) seen in the graphs seems to be originating from the port or maybe deep inside from cabinet resonance; although it's a little less peaky in the right speaker. And there's also a smaller third and second harmonic peak seen here which is reflected in the original graph as well right around 400-500Hz.

1608661906461.png


If you're wondering how these monitors perform with a HP filter enabled, here are the graphs:

1608657039361.gif


It looks like distortion above 300Hz is not mitigated by much with the use of said filters.

1608657043809.gif


The Sceptre does have the option to apply a HPF internally with some button switches right at the back. I noticed that it also applies some subtle phase shaping improvements to the speaker as well that's not seen there when I'm using manually applied cascaded butterworth filters (24dB/oct HP xo) via JRiver.
 
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LTig

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How sure are you that you did not became a victim of expectation bias?
 
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ernestcarl

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Directivity Curves

I mentioned in another thread that I would be measuring the KH120s a little further away at 30cm instead of 15cm to even the the playing field between it and the S8 (for vertical curves mainly). As it turns out, even that was not sufficient, and I was still seeing similar curves with a significant amount of displacement in volume levels due to the drivers not being located concentrically. I took several re-measurements to confirm and it really is the case... sigh. Too bad, my tripod simply is not tall enough and room big enough -- let alone free from reflections! -- to do such measurements the best way possible. Nevertheless, let's proceed anyhow since I already measured these thrice!

Optimizing speaker EQ applied
1608662867151.png


Graphs with all red traces belong to the Sceptre. You can clearly see now -- with the verticals separated from the horizontal -- how the pattern of traces bunch up between 5-10 kHz. This is quite a significant finding to me personally. It explains a good deal of why I consistently find the HF pleasantly bright and 'present' (even more lively and alive) when listening on- and off-axis. And although there is some lobing, it's located at 3.5 kHz -- important, but quite a bit further away from the much more important range between 1-3 kHz.

Whereas look at the Neumann: horizontals are superbly controlled but there's also a very wide dip right between 2-7 kHz. @thewas and @Pio2001 have both posted similar MMM graphs as well which showed how the KH120s had a somewhat relatively "dark" characteristic sound when compared to the LSR305, and I believe the Genelec 8030c too. The extreme end of the HF is also attenuated less, making these sound a bit harsher even after applying the monitor's own internal HF shelving switches. With the S8 you see a more obvious gradual decline shelving down which is ideal with home theater type of applications where you might not want the left and right channels to sound so loud if you're sitting further away off-axis from the center seat, and when using an exaggerated toe-in (if on-axis are crossing paths in front of the listening area) for a wider consistent stereo imaging listening area. If you're relying on a phantom center for voices instead of a center channel, this type of speaker calls attention to itself a little less annoyingly so at the extreme ends of the length of your seating area. Logically, it's certainly less annoying than if all the levels stayed constantly loud whichever angle the speakers are pointed at. These speakers are most definitely going to be gentler to the ears.

For the vertical directivity curves, I didn't bother measuring below the driver as essentially you will get very similar curves anyway. This has been confirmed to me by the measurements and review performed by Keith Holland at Resolution Magazine. Again, the pattern is very similar, albeit, with a bit more uneven bunching behaviour between 5-10 kHz. This is not a problem as much of the all important area between 1-3 and even 4 kHz remain strongly intact and smooth. Much, much more consistency and coherency with voices whatever position you are in... that also gently tapers off as should be expected.

Okay, now look at the (exaggerated due to very nearfield measurements) disaster we see with the verticals in the KH120 -- my graph above. Hmmmn.

And compare it with this:
1608665646744.jpeg


I believe many of us may have seen this isobar graph which doesn't look so bad, right? Yeah, I think it glosses over the displacement issue between many non-concentric designs quite a bit. We're also looking at specifically nearfield studio monitor designs here where such potential characteristic changes are more audible the closer you are from the drivers. Actually, it's quite easy to get used to such subtle changes in sound character if you've lived all your life with monitors such as these. But I've had the Sceptres for a quite long time as well, and the slight shifts in sound exhibited by conventional two-way designs like the KH120 (better than many) does not escape my attention no matter how hard I try.

I mentioned my sit-stand desk use-case before... with my sensitivity to the varied height level changes.

Now, look closely at where the vertical lobing appears on the Neumann KH120 graph, quite prominent right around 2.0 and 2.5 kHz. This is not very good. How can it possibly be? Sure it's typical and may be expected for this type of design, but you surely have less wiggle room here. It also might even affect the timbre you hear more than the S8 from desk reflections if you have them sitting flat on the desk without any stands to prop them up. Yeah, that's stupid and who does that anyway? :rolleyes: Eh, actually I've seen people do that with monitors regardless of make from both producers and consumers of music. And if you're sitting down for while and have to stand up, the sound will change and shift quite a bit more noticeably. This can be annoyingly disturbing -- depending on your own tolerance level for this sort of thing. For most people, they probably don't give a hoot.


KH120 curves overlayed with S8

1608667042351.gif


1608667053288.gif


There... I believe my speaker EQ correction is mild and fair in its overall effect. It also makes the Sceptres sound more neutral, and gentler to my ears, at least.

The red traces (S8), while not nearly as smooth, are much more consistent at all angles.

You will see the same case as well when we get into the time domain performance.
 
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ernestcarl

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Standard Overlay Graphs in REW with all directivity curves
and Time-Domain comparison



Clarity (C50)

1608669375196.gif


Red traces tell us that the Sceptres should have better speech clarity over the KH120 -- in theory! Well, I don't really care either way since, IMO, I prefer the voice tonality I hear coming from the latter monitor. Doesn't mean the Sceptre's are bad for speech intelligibility (well, that's total nonsense as it -- actually both monitors -- excel in this regard!) -- it's just not what I prefer myself.


Group Delay

1608669880059.gif


This looks a bit more confusing to interpret. I'll try anyway...

Neumann traces are extremely smooth, but slows down more down to about 1.5 kHz. After that, though, we see an utter chaotic mess (but please, be reminded these are taken at the extreme nearfield -- 15cm from acoustical axis so this isn't going to be necessarily bad [very likely] the further the distance is) -- not to mention all reflections and interferences going against the pure direct sound as sound travels and spreads out further into all sections of the room.

With the S8 there are a lot more prominent undulations but they're also bunched-up together quite tightly throughout, and more linear to t=0. We don't see as much deviation curve to curve all the way down to 500Hz -- other than that area around 3.5 kHz where we saw the horizontal lobing.

Now, it is important to note that we are quite sensitive to time deviations in the mid-range through the HF area. Much more so than in the bass. You can test this yourself by simply playing with the channel delays of your monitors using DSP in software or DAW. I can notice obvious image shifting effects when performing small 0.05 - 0.1ms adjustments. Not so sure how obvious it is below that as I've never bothered to test it for practical reasons.


@bennybbbx wanted to see the Step Response

1608671286895.gif

I am happy to show you all them at all directions measured

I see a small amount of delay and, maybe, "time smear" as we change angles on the KH120. In most cases, I believe we really aren't very sensitive to this -- but it's effects should be possible to hear with very careful listening of certain track samples which highlight this 'smearing' effect e.g. fast moving, densely packed transients. I already provided the track link which I thought best demonstrated this so feel free to test if you can get a hold of similar speaker designs to what I had compared to side-by-side -- or even if you have a pair of linear-phased KH80s or Hedd monitors. I'm not sure but you could probably have one speaker be linear-phase, and the other one not. Just go ahead and try to tell us what you hear of the mentioned, if any noticeable changes anyway. Neumann claims it adds improved transparency -- whatever that is... I have disparaged others before with my own worry-wart skepticism of what others thought as obvious differences between linear-phase systems and traditional 'not so linear' speaker systems -- and might still will, just to be a gadfly. ;) Also maybe the effect is magnified with certain coaxial designs more-so than in traditional multi-way speaker systems. I honestly have no clue at this point in time as I haven't done the extensive research required to know if this can be asserted as verifiably true to most listeners with absolute confidence.

Unfortunately, I'm not going to be making bold claims about an expanded 'stereo-width' or 'soundstage' here after my listening session -- both monitors perform excellently in these domains. What I heard so conspicuously, very specifically, was the S8's superior resolution of HF transients. That's it. Actually, I've even mentioned in my posted observation notes that I found the KH120 clearer and more detailed in the majority of tracks listened to! It's not as if the S8 stomps all over the KH120s at every place in all other performance metrics. And also, just to as a pre-emptive caveat, when I wrote the KH120 are 'harsh', I'm comparing them relative to the S8. Not Dynaudio, Adam, Genelec etc. monitor. Just the S8. The same goes when I say I get fatigued faster on the KH120. Doesn't mean I listen consistently at loud levels most of the time and experience frequent bouts of fatigue listening to the KH120s -- quite rarely, in fact. I can listen to them all day long at reasonable volume levels.


Impulse Response

1608672656520.gif


I see a gradual slowing down and some time-shift displacement with the KH120.

You know what, I think the S8's horn waveguide might be the reason for some of the undulations we see here, as well as in the group delay plot etc. Remember the geometry of the horn waveguide has lots of sharp bends and the after effect of this (disregarding the obvious reflective resonances eliminated via DSP) and interference between the back of the horn and the woofer itself seems to be inescapable no matter what. We see this in the irregularity or unevenness in places around the magnitude response (amplitude over frequency) as well. It's smooth and relatively linear, but far, far from the buttery smooth clean curves exhibited by the KH120 at every angle -- even the vertical curves' lobing/dips look super smooth. :D


Phase

1608673386158.gif


Wow! I used to doubt the concept of phase coherency, but it's very real folks. My, my, my this is very impressive. It's not ruler flat like what we see with other highly corrected two- or three-way designs using serious convolution. But to me, it looks way more natural. I am guessing, with trepidation, that it may be possible to optimize this a bit more if it had a more powerful built-in processor in place-- but that's just my presumption as I don't know the detailed ins-and-outs or limitations of such computer-aided speaker designs.


A few other reviewers have performed time-domain measurement evaluations on the S8 so I'll include excerpts from two of them.


Keith Holland (Resolution Magazine):
1608674046062.png


I mistakenly thought the delay was over 3ms, but my own measurement confirms a similar value.
1608674169311.png


Barely any delay. This means you can use this monitor for real-time tracking or playback of instruments without any fear/annoyance due to delay caused by the time-domain DSP correction.


Fei-Fei from exbound.com (Chinese audio webzine -- but I think they moved to another web address now?):

There's a translation in English archived by Presonus in their list of "reviews" for this product at their webpage. It isn't the easiest to understand as some of the wording sounds so wrong and out of place. So I used google to translate direct from the Chinese script, and it made even less sense to me! I've edited the ff. excerpt the best way I could make sense of it.

1608674516547.png

Using Smaart real-time analyzer


And here are my own single-point measurements:
1608675132077.png


1608675143562.png



As I already had all the curves, I thought I might as well attempt to make a vector average of all of them just to see how the results look.


1608675190330.png


1608675200777.png


Actually, the phase looks quite good in both if you average it all out into one trace.


Vector averaged step response

1608675327122.png


1608675341521.png


1608675348168.png


1608675352633.png


Lastly, some zoomed-in Wavelet Spectrographs using 1/12 resolution setting.

1608675502948.png


1608675507780.png


1608675513119.png


1608675517461.png



Oh yeah, I took some pictures inside the cabinet a while back:

1608677668950.jpeg


1608677684098.jpeg


1608677688701.jpeg


Yeah... not quite as I had so optimistically imagined them as seen with Fulcrum Acoustics's more expensive drivers:
1608677781078.png


Oh well.

I did not disassemble the amp plate, so here's just a one point of view perspective:
1608677859973.jpeg


There are clearer pictures from Fei Fei's own review archived here: https://pae-web.presonusmusic.com/uploads/products/reviews/Spark_Digital_Sceptre_S8_review_EN.pdf
 

bennybbbx

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many thanks that you post the step response. very intresting your measurement. I notice that the S8 have lots bigger bass/mid as the KH120. but the s8 is faster in step response, but the s8 have higher swing out levels. i can not say on measurements which speaker is more precise in hearing.

maybe you can record my testsong. you need only 1 microphone and 1 speaker the way i now do. in testsong highhat is on left side so can easy hear which side you play. put my testsong in a DAW. switch the DAW output to mono or use a VST in master that create mono. Now pan the testsong in the DAW mixer channel to full left. now create 2 mono channels mamed left and right which have as input source the same microphone. pan the left channel to full left in mixer and the right channel to full right. mute the microphone channels so no feedback can happen.

because you have in the testsong original channel pan to left you need record arm the track with the microphone input that is panned left.
now press record in DAW and make sure testsong channel is not arm for record. only input of your microphone. so all stay sample sync. when you have done record the left channel, disable the record arm for left channel in microphone track and enable record arm for your right track with the same microphone. Now pan in the testsong original track to full right and press record button to record right channel. now you have both channels sample sync. mute the original testsong track disable microphone and unmute the 2 microphone tracks left and right. see screenshot of studio one. there is a small free studio one version available which allow this records too in studio one. important is only that input of microphone must be on same soundcard as output of the teststong. use diffrent soundcards for output and input in asio for alls never work for me, because happen sometimes crackle because they are not sample sync then. now whenyou have the 2 mono tracks. set the markers from start to end and choose song export mix as a mp3 with 320 kbit(is good enough) or as wav and name it as the speaker you record. now you have a stereo file which can later correct with EQ. If you think there can not hear a diffrence you can ofcourse test me with my testfiles from this thread https://www.audiosciencereview.com/...of-the-synth-sound-in-the-attached-mp3.18401/ and change with eq the testfiles so i can only detect on width. my english is not so good. hope you understand what i mean

EDIT: on my measure
kali(6.5 inch) reach highest in 0.5 ms.reach 0(first 0 reach) at 1.25 ms. reach 80% again at 5,5 ms
mtm(2* 3.5 inch) reach highest fast (not measureble time). 0 at 0.8 ms reach 70% after 6 ms.
Canton+Kali (4 inch mid) reach highest in 0.25 ms and 0 in 1 ms reach 40% after 6 ms
celestion (10 inch) speaker reach highest in 0.1 ms and 0 after 3.75 ms reach 40% after 5 ms

on your step response can not see so good but i guess
KH120 (5,25 inch) highest 0.7 ms reach 0 at 1.8 ms (1.1 ms from highest to 1. 0) reach 40% after 10 ms
SPectre S8(8 inch) highest fast reach 0 at 1 ms reach 50% after 7 ms


studio one speaker rec_2.jpg
 

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bennybbbx

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In this post I’ve linked to quite interesting measurements with K+H O500:
https://www.audiosciencereview.com/...ds/time-domain-measurements.12951/post-543807

intresting article. It seem there is much diffrence in hearing from people. 1 ms is frequency 1 khz . 10 ms=100 hz. 0.1 ms=10 khz. I know i hear not more as 18 khz and a diffrence between a tweeter that reach 40 khz or only 25 khz or AMT i did not hear. I hear only the big diffrence in the mid range. so i have not the" high end ears". and diffrence between amps i hear not. If somebody not hear a diffrence in my tests in stereo width and stereo separation, then the speaker is good enough for him. for my hearing Kali LP6 sound very worse in stereo and dull. some german testers at delamar or bonedo report too what can interpret as a transient problem. but others are happy with the sound. when LP6 for hearing only then is ok when like the smaller sound better. but it is not precise. and when you mix and want make a wide pad synth it sound very small on Kali. so you tend to increase stereo width(when you hear diffrence).maybe only few hear so much diffrence what i hear, but also for me it help that kali sound much better more realistic with this ozone settings. i think about do another thread poll and let people choose on the original example process with diffrent smaler stereo width which the think smallest. ears are really strange. because when i enhance the LP6 with this settings it loose the muddy sound and sound brighter, but when i use this setting on mtm or canton then it sound not brighter only wider and sound as in a bigger room sit. normaly LP6 sound for my ears as i hear music thru the door and not sit in the room
stereo width for kali.jpg
.
 
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ernestcarl

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many thanks that you post the step response. very intresting your measurement. I notice that the S8 have lots bigger bass/mid as the KH120. but the s8 is faster in step response, but the s8 have higher swing out levels. i can not say on measurements which speaker is more precise in hearing.

maybe you can record my testsong. you need only 1 microphone and 1 speaker the way i now do. in testsong highhat is on left side so can easy hear which side you play. put my testsong in a DAW. switch the DAW output to mono or use a VST in master that create mono. Now pan the testsong in the DAW mixer channel to full left. now create 2 mono channels mamed left and right which have as input source the same microphone. pan the left channel to full left in mixer and the right channel to full right. mute the microphone channels so no feedback can happen.

because you have in the testsong original channel pan to left you need record arm the track with the microphone input that is panned left.
now press record in DAW and make sure testsong channel is not arm for record. only input of your microphone. so all stay sample sync. when you have done record the left channel, disable the record arm for left channel in microphone track and enable record arm for your right track with the same microphone. Now pan in the testsong original track to full right and press record button to record right channel. now you have both channels sample sync. mute the original testsong track disable microphone and unmute the 2 microphone tracks left and right. see screenshot of studio one. there is a small free studio one version available which allow this records too in studio one. important is only that input of microphone must be on same soundcard as output of the teststong. use diffrent soundcards for output and input in asio for alls never work for me, because happen sometimes crackle because they are not sample sync then. now whenyou have the 2 mono tracks. set the markers from start to end and choose song export mix as a mp3 with 320 kbit(is good enough) or as wav and name it as the speaker you record. now you have a stereo file which can later correct with EQ. If you think there can not hear a diffrence you can ofcourse test me with my testfiles from this thread https://www.audiosciencereview.com/...of-the-synth-sound-in-the-attached-mp3.18401/ and change with eq the testfiles so i can only detect on width. my english is not so good. hope you understand what i mean

EDIT: on my measure
kali(6.5 inch) reach highest in 0.5 ms.reach 0(first 0 reach) at 1.25 ms. reach 80% again at 5,5 ms
mtm(2* 3.5 inch) reach highest fast (not measureble time). 0 at 0.8 ms reach 70% after 6 ms.
Canton+Kali (4 inch mid) reach highest in 0.25 ms and 0 in 1 ms reach 40% after 6 ms
celestion (10 inch) speaker reach highest in 0.1 ms and 0 after 3.75 ms reach 40% after 5 ms

on your step response can not see so good but i guess
KH120 (5,25 inch) highest 0.7 ms reach 0 at 1.8 ms (1.1 ms from highest to 1. 0) reach 40% after 10 ms
SPectre S8(8 inch) highest fast reach 0 at 1 ms reach 50% after 7 ms


View attachment 101110

Hi! Yep, sure, I think I might be able to do this after the holidays. I think I need to take a short hiatus for now.
 
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ernestcarl

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If I didn't set up a 10 PEQ speaker EQ limit... and used REW's total maximum of 20 in the generic EQ setting.


What would be the result?


1609439483362.gif



The curves will look a little smoother here because of the windowing applied, but these are the exact same measurements taken previously.


1609439570929.gif



Original 'Basic' 10 PEQs ~vs~ 'Extreme' 20 PEQs w/out ERB smoothing
1609439583514.gif



Top curve is all directivity traces simply averaged together.
1609439689720.gif



Separated and more detailed view of Original 'Basic' 10 PEQs ~vs~ 'Extreme' 20 PEQs w/ ERB smoothing
1609439753220.gif



Transparent Overlay
1609439850895.png


I listen off-axis so primacy of flattness is given to somewhere between ~ 15-30 degrees.

Since we're just using IIR filters, how would it affect the phase?
1609439907105.gif


KH120 -1dB internal HFS ~vs~ S8 'Extreme' Speaker EQ
1609439915678.gif


No, the phase is not flattened to t=0 at all angles combined.

Although, it does look smoother.


My EQ abuse(?) might make some eyes pop out in outrage so I pasted it last.

1609440553665.png
 
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ernestcarl

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DRIVER COMPONENTS ~ LOW FREQ & HIGH FREQ DRIVER XO

I opened the right speaker cabinet in order to unplug the power leading to either one of the drivers -- which yielded the ff. measurements.

1609559035105.png

*Distortion should be a dB or two higher. Never mind the relative levels.


Above is an overlay I made of the magnitude and distortion views.

Notice that when measured outside of the port, the noise/distortion shifts to the right.


And separate views for clarity:

1609559211316.png



REW's alignment tool gives the ff. summed prediction based on the phase information:

1609559271016.png


Seems about right.

*Sorry, no, actually it looks like the predicted summing didn't quite use the phase information as expected since I was not able to use a time reference for the woofer -- forgot about that. It seems to be nothing more than a predicted sum if phases were "optimal" -- which they are, I believe.

However, after manually applying to what looks like the appropriate time offset for the woofer (based on the referred summed step response) + some additional windowing, I get the ff. superimposed result:

1609567621443.png


2.4 kHz just looks to be the "mid-point" between the critical zone (+- 60 degrees or additively) area or interval where LF & HF magnitude responses both significantly affect each other.
 
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ernestcarl

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I believe the separate driver components measurement is telling a half-baked story... The reason is we're not really seeing the 'raw' response -- and for that, I have to connect each driver to another external amplifier bypassing any DSP.

1609564745207.png


For example, that huge dip in the tweeter's response possibly (just a guess) might have been deliberately added to kill-off some audible resonance, otherwise.


Another coax design from Fulcrum Acoustic (RM28ac):

1609565201028.png


1609564992015.png


Even in the above with much more detailed specs, I also don't think that these two graphs tell us enough as I believe the light yellow trace is already the combination of the upper 8" bass driver and horn compression driver (combo). Purple trace looks like the 2nd (lower) bass driver.

To gather more insight in the design, I'd probably have to strip the cables and connect each driver to an appropriate ext. amp -- measure the impedance and raw unprocessed phases too. See how the DSP transforms the raw response into the 'finished' one -- i.e. so reverse engineer the DSP of the whole speaker system?

LOL! Eh, in another life time, maybe. Not this time.
 
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ernestcarl

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INTERNAL DSP HIGH PASS FILTERS

Due to the high-order roll-off design of these monitors, applying ordinary external HP filters e.g. LR4 or cascaded butterworth filters are going to cause additional delay in the phase. If you want to keep it as low as possible, one might prefer to use the monitor's own internal HP filters.

1609604566176.png



1609604578335.png



INTERNAL DSP HIGH SHELF FILTERS

For those who would rather not dabble with external DSP, the monitors do have their own built-in high-shelf filters. Unfortunately, though, these are not refined enough to satisfy me fully.

1609605019050.gif


Honestly, there should have been a -2dB and -3dB setting. Arghh! For the longest time, I've actually found the -1.5dB shelving good enough... but overall these shelving filters are just too crude. If you want the best sound out of these (that will best fit your own personal preferences), you really need to play around with a DSP parametric equalizer.


"EXTREME" EQ OF THE SCEPTRE S8 COMPARISON WITH GENELEC's "THE ONES"

I came across Genelec's time domain plots thanks to fellow member @Hephaestus.

Since Genelec obviously doesn't use REW, I had to try to make my own measurement plots comparable enough with theirs.

These are some of the earlier models -- AES paper was published in 2017.
1609602051976.png


IMPULSE
1609602101342.png


GROUP DELAY
1609602112472.png

no windowing, just ERB smoothing

WAVELETS
1609602159962.png

Default resolution. With my own EQ to even out the frequency response. Without it, the HF is a tiny bit hot in places.

1609602270613.png

Normalized to peak.


QUICK CONCLUSION:

Time domain-wise, there really isn't all that much of a big difference between the Presonus Sceptres and Genelec "The Ones". Both are excellent.

The area where the Genelecs perform above-and-beyond these more physically, traditionally horn-loaded coaxes is in the directivity smoothness and consistency -- much, much less diffraction and interference due to their ultra-smooth geometry. But that isn't really surprising at all as their waveguide tech is newer and significantly more advanced, while also taking up a lot more room now covering the entire front fascia of the monitors. This is very good as unlike the very exposed waveguide horn which could be damaged easily by careless handling, their tweeter (protected with a grill) and with the mid driver are kept flat and shallow enough that it's not really likely for one to accidentally damage or mishandle them. Another advantage outside the time-domain would be their much lower distortion levels -- I'm not so sure about the smallest model -- but the bigger ones, definitely.

In terms of price to performance, uh, c'mon... I think the answer it pretty obvious. ;) Though, you still would need to use some extra external DSP to even out the S8's response to taste, IMHO.
 

bennybbbx

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The genelcs you post are 3 way and have good impulse response and are very expensive.
 
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ernestcarl

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The genelcs you post are 3 way and have good impulse response and are very expensive.

They are expensive, true. But the cost is peanuts to others. The S8 and S6 are kind of "ugly" to be honest, but I don't care myself as they do the job required of them -- and they give a totally different perspective from my other monitors. ;)

BTW, I modded one of the Sceptres. Used along with my current sub, the improvement is substantial -- I think : https://www.audiosciencereview.com/...ces-of-mains-in-a-2-1-setup.19039/post-627254
 
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ernestcarl

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That prominent second harmonic peak at 1kHz or before (it shifts with mic position) seen in the graphs seems to be originating from the port or maybe deep inside from cabinet resonance; although it's a little less peaky in the right speaker. And there's also a smaller third and second harmonic peak seen here which is reflected in the original graph as well right around 400-500Hz.

Realized I've been looking at the port measurements wrong. I should not have offset the dark blue (mid-bass woofer) and green (tweeter) traces way down in level.

1609921848842.png


Now I know why that 1 kHz distortion peak could not be eliminated with my port blocking and internal damping experiment -- it's not likely a harmonic distortion (entirely) but part of the tweeter's fundamental response -- just very low in level -- I think.
 
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ernestcarl

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Co-axials do soud different! Co-herent!

Well, Presonus calls them Co-Actual for marketing purposes. Dunno why they bother re-inventing the name as it doesn't make the speakers look any prettier. :p
 
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