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Avantone CLA-10 (Yamaha NS-10M Clone) Review

Rate this studio monitor

  • 1. Poor (headless panther)

    Votes: 164 88.6%
  • 2. Not terrible (postman panther)

    Votes: 8 4.3%
  • 3. Fine (happy panther)

    Votes: 5 2.7%
  • 4. Great (golfing panther)

    Votes: 8 4.3%

  • Total voters
    185
@thewas last quote end points really explain a lot and in the meantime from when Toole whose active and wrote the book digital signal processing progressed a lot. So let me formulate the question differently to you. Why it's so hard for you (all of you) to believe how really far from great sounding speaker when paired with good sub's and not pushed to do what it can't but other way around stressed out with a fairly simple EQ can sound pretty good (not to say great)? Meanwhile for many such headphones you took it for granted how they do transform into such the same way with use of simple EQ. There is no either, either and rule is rule or it's not that at all. Of course some things you can't EQ and you can do more than simple EQ. People don't have 120 m² listening room's nor are even studios that big. Why it's then hard to understand that less problematic by design speakers would be easier to handle in such real life far from optimal conditions? You do understand that if you pay attention to it by doing EQ-ing you can even impulse response between speakers (as they won't be at ideally equal same distance from listening spot), that you won't allow them to develop deviations (pre - post ringing and so on) and can even improve impulse response. Also you can make highs beaming much more focused (± deviations from averaged target) by the use of inverted impulse response FIR and slope it down future more if need be by wide PEQ. Of course you will use minimum phase. So actually a lot can be achieved. I never claimed how there were great speakers or great sounding out of the box but how when properly paird and EQ-ed they can become on pair with some of your today's favourite and beloved active DSP-ed one's and how you can control and improve some aspects on passive one's especially in areas which are rapidly progressing in comparison to out of the box active DSP solutions. Finally nothing will change the fact how great and rare mastering engineer's (which by the way are capable of doing that even with a sub pair equipment when they get to know it) produced some great now classics even primary using those in the process. However average Yoes full of them self menaged to destroy ocean of materials no matter how good equipment they had and where actually paid good in the process. Disclaimer is how DSP has it's limits and most limiting factor is the user the same goes for creative mixing - mastering processes and how engineers aren't musical content creator's in any way. Where simple acoustical treatment is more efficient it should be used before or in conjunction with small EQ adjustments like for highs and to a point mids. It's smart to use everything that is in your disposal in the way that it's easier and more efficient to achieve gool and disregarding any meens can be understood as ignorance. Law is not law if its not valid for great majority of cases (statistically in %) and how anomalies that divert from it need to be tracked down and explained and how this is fundamental thing in scientific methodology for any exact science. At least I told something myself instead of vogue quoting authority in the way of in the sense of plucking from the wider context.
 
So it seems you have no RT60 measurements to support your claims. ;)

@thewas last quote end points really explain a lot and in the meantime from when Toole whose active and wrote the book digital signal processing progressed a lot. So let me formulate the question differently to you. Why it's so hard for you (all of you) to believe how really far from great sounding speaker when paired with good sub's and not pushed to do what it can't but other way around stressed out with a fairly simple EQ can sound pretty good (not to say great)?

I didn't and don't see anyone here claiming that in nearfield monitoring (to reduce the impact of their directivity problem) with some EQ they can made to sound decent, it was incorrect claims from you like

If you're precious KH or Genelacs score the same or worse to NS10M's both paird with two closed enclosure Genelac or KH 10" sub's and EQ-ed still having considerably worse time domain how are they better in any regard?

which were commented and corrected.

Finally nothing will change the fact how great and rare mastering engineer's (which by the way are capable of doing that even with a sub pair equipment when they get to know it) produced some great now classics even primary using those in the process.

Which was also commented that a good and experienced mastering engineer can produce great results even with poor tools and that also they were usually not used as primary monitors but rather as a compatibility check and standard for mixers working in different studios as they were quite common.

Law is not law if its not valid for great majority of cases (statistically in %) and how anomalies that divert from it need to be tracked down and explained and how this is fundamental thing in scientific methodology for any exact science. At least I told something myself instead of vogue quoting authority in the way of in the sense of plucking from the wider context.

I hope this is more a language barrier but such comments of you seem to be in significant discrepancy from your posted content.
 
The thing is that the Yamaha NS-10s just worked out of the box for these mixing engineers to get the mixes to translate to most speaker systems, I don't think many of them had a clear idea of why that was the case as most of them probably never analyzed it in detail. These speakers just worked as a tool to quickly get a good balance between the sound objects in the highly "crowded" midrange (as it was a very midrange-focused loudspeaker), and that made the music sound good on everything from really great full-range systems while also sounding fairly decent on lesser systems which mostly covered the midrange frequency range.

We can of course analyze the reasons...
The midrange is the main thing to get right in a mix to get it to translate to most speaker systems out there. You can make a mix sound great on a speaker system with good bass extension even when the balance between the different sound objects isn't the best in the midrange area, it's like the extended bass covers up for the flawed mismatched balance in the midrange, but as soon as you listen to the mix on the lesser extended system, it can become more obvious that something is off in the balance of the mix as those sound objects with more bass-extension no longer is heard as expected and drown in the mix because of the lack of energy in the midrange area.

When looking at the charts below...
Most of the "punch" of a kick drum is in the range of 50Hz-100Hz, but the "fullness" is higher up at 100Hz to 250Hz and the "attack" is way up in the region of 3kHz to 5kHz. In a speaker system with a good bass extension, the "punchiness" and the lower bass range will somewhat cover up the lack of energy in a not-so-greatly balanced midrange, but in a speaker system that lacks the bass extension, the kick drum may hardly not be heard at all if it lacks the natural energy in the midrange area. As can be seen in the chart, a kick drum covers a wide range from about 50Hz to 8kHz (sometimes even further up than a guitar), so a good balance with all the rest of the instrumentation is needed for it to be heard naturally even higher up in the frequency range.

1723536815723.png


Toole suggests that the same could be achieved with EQ and that is probably true, but the main thing here is that the NS-10s worked out of the box for these mixing engineers and they could easily and quickly set them up without the need for further analyzing and fiddling, it just worked for them out of the box.

But the NS-10s also had other "qualities" other than just the midrange-focused sound when looking at the data found here: https://dt7v1i9vyp3mf.cloudfront.net/assetlibrary/n/ns10m.pdf?jQWj8tYIeZeymRCNXitG9Qfwq9mLf1t0.
Mixing engineers found it particularly easy to judge the amount and length of reverbs, and also the attack and the release times of the compression. Whatever the technical reason for that may be, also that just worked out of the box for them.

The main thing is that these speakers worked out great for many mixing engineers, and as it all comes down to achieving a great result it doesn't matter what tools were used to achieve just that. They made great-sounding reference-level mixes on the NS-10s over a long time which are still considered reference-level, and that is all that should matter to us consumers of high-quality music. How can we argue against that, what works for some people may not work for others, it's always the result in the end that matters, nothing else. :)
 
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I’m at this site because I listen to experts like Amir. He’s an expert in the field of audio electronics, measuring hi-fi gear and the science of what we hear.

I listen to quite a few producers because they’re expert in their field, and I find it interesting to hear how they’ve accomplished certain sounds and recordings, etc.

All the producers I know who’ve commented say they know the tonal balance of the Yamaha is ‘wrong’, but know it so well that they know how to work around that.

That may sound counter-intuitive to you and I, but as I say, I listen to the experts. If they say they can do it, I trust them.

I’m seeing a lot of producers saying now that so many excellent, neutral speakers are available for such relatively little money, that almost everyone is using these, so I think it’s a bit of a moot point.

Good, balanced comments from Warren here:


He uses Genelecs these days.
There were a lot of small monitors around c2000 or so that mimicked the NS10 to a greater or lesser extent, all with a 2khz or so response lift*and a straight line down-tilt below (in the UK 1980s, we had Linn Kans which did similar things if not moreso!). I still don't believe that NS10s were used as primary mixing tool, but rather as a magnifying-glass on the critical upper mids as *part* of the mixing process.

* https://dt7v1i9vyp3mf.cloudfront.net/assetlibrary/n/ns10m.pdf
 
@thewas
"I hope this is more a language barrier but such comments of you seem to be in significant discrepancy from your posted content"
Which content of mine would that be? At least I have valid master degree in formal logic and did much more scientific methodology and research then you ever will.
"So it seems you have no RT60 measurements to support your claims. ;)"
Couple of room measurements thread's around with plenty examples. Insisting on something stupid doesn't make you look more intelligent, on contrary.
Yes it whose all previously discussed and it's pretty much pointless to do so future with possible exceptions of "fantasy scenario" think that whose a new one.
Only valid scientific research into audio reproduction or rather a collection of over a hundred of them with a sufficiently representative (all together) sample is the one about equal loudness compensation and all do they don't agree entirely with each other results are in line with eachother for a most part. Everything else is exactly opposite of it and to make things worse no one is even trying to do any which could be considered as such.
 
Aren't these used a lot less in the last 10 years than decades back? If so, has the quality of the mix worsen as a result of it?
Still used a lot, but good question, he he!
Could the fatigueing sound also be related to Gennies? :eek: :D
 
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@thewas
"I hope this is more a language barrier but such comments of you seem to be in significant discrepancy from your posted content"
Which content of mine would that be? At least I have valid master degree in formal logic and did much more scientific methodology and research then you ever will.
Sorry but such pathetic title bragging (which you have done also before) about an obviously for the topic useless degree is just a confirmation that when people cannot argue with facts they fall back into fallacies like argument from authority. You should have noticed that people with real knowledge on topics never state their titles in discussions because they don't need to and a title while it can help (if its relevant) it is not a guarantee for quality posts.

Couple of room measurements thread's around with plenty examples. Insisting on something stupid doesn't make you look more intelligent, on contrary.
None of them for the NS10 or CLA-10 (even more done on the same rig to compare with some good BR examples), so insisting on rather does that to you.
 
Aren't these used a lot less in the last 10 years than decades back? If so, has the quality of the mix worsen as a result of it?

You should reverse that question.
Has the quality of the mixes gone up in the last 10 years because of the less use of NS-10s as mixing tools in the studios?

On one hand, thanks to the shift to digital audio workstations from purely analog gear have made it possible to clean up the recorded audio tracks with much greater precision, but on the other hand, the shift to “we fix the problems later in the post-processing” instead of getting it right in all the previous steps throughout the audio production has, in my opinion, made things sounding more artificial than real. In the past, audio productions were more about documenting “a band playing in a real room”, while nowadays it's more about creating a clean-sounding artificial “reality”.

But none of the above has anything to do with the choice of mixing monitors, now as well as back in the days it has always been better to use the speakers that worked better for you than using speakers that worked better for someone else. It’s all about the resulting quality, not which tool is used to reach that goal. :)
 
But it does not necessarily work vice versa. At least not for me (producing EDM mostly). A big plus of these speakers when it comes to unmasking problems. For the same reasons I keep the sub off most of the time, or why I sometimes use bandpass EQ settings on the monitoring chain.
For example, on my KH-120s, with full kick and bass, I don't hear, if the cymbals bus has transients that are still too hard, while the NS-10 poke the ears. Once noticed on the NS-10, it becomes obvious on the KH-120, like it becomes hearable everywhere.
Or when it comes to the attack of softsynths, especially with fast notes like low sequences. On my KH-120 a lo seq can sound good, but on the NS-10 it immediately stands out, if it has the typical VSTi attack.

I've heard a lot of silly nonsense in my time, but "typical VSTi attack" is a new one and probably takes the cake. Like what on earth are you talking about?
 
@thewas you didn't even try to provide any facts or measurements or any empirical data, all you did is quote other people. And you didn't show me that majutic content of mine. You can find measurements and form NS10's around the net (S.O.S. had a serie of articles including measurements as much as I remember among many others if you dig deep enough). What I said is that time domain is valid for behind 120 Hz and you can see that form pretty much almost any speakers measurements in the RT60 decay plots and as you will hardly find any for closed enclosure speakers you can instead use one's with ported speakers and such sub's (including mine) or even better sub's standalone comparative measurements regarding spotting the differences. Surely I don't have to find them for you at least that much you can do on your own if it interests you. Regarding degrees and titles I am modest (but surely not shy) in comparison with lot others hire, in your case it's quote's instead. Have a nice life, regarding me this discussion is over.
 
@MAB that whose sarcasm. I really, really won't quote myself but if you would be kind enough to read closing point in #1136 you would see how I exactly already pointed the same thing. And it's not only pore implemented crossover but combination of that and cabinet resonance prior to it. Ironically creators of CLA 10 praised around how they did a much better job regarding crossover, and they clearly didn't. Space between the line's represent possible headroom, of course you can't correct that entirely with EQ, not even with EQ+phase correction never the less calculated score and measurements are that what there are not my phantasy and done by our host and who ever did calculate EQ, everything else is a subjective bias and if you want to question that be my guest and take it with them. Enjoy your time and spend it wisely.
analyzing the impedance curve of this speaker, you can boldly list the housing parameters. F0=100Hz Qtc=0.7.

Now, if we use the same active filter with the same parameters, we will get an acoustic high-pass filter Linkwitz LR24dB/Oct at 100Hz. I guess you can already write what parameters a subwoofer should have to get a linear Amplitude graph. And what I really care about is where these old recordings come from, e.g. Pinkfloyd, which sound so dull and dark on linear Monitors and on NS10 it's ok
 
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The midrange is the main thing to get right in a mix to get it to translate to most speaker systems out there.

Indeed. And it's not only a matter of assuring the mid range is well defined and balanced, it's also about makings sure your mix still works on other speakers with weak bass performance. It might take some extra work on the bass drum and bass sound to assure they don't get lost and still drive the song. When done well, this is not detrimental to the sound. That's what lots of people fail to understand.
 
@thewas you didn't even try to provide any facts or measurements or any empirical data, all you did is quote other people.
I wasn't the one who made unproven claims, if you had asked for proof for my claims (like for example directivity mismatch) I would have easily showed such to you like @MAB did.
You can find measurements and form NS10's around the net (S.O.S. had a serie of articles including measurements as much as I remember among many others if you dig deep enough). What I said is that time domain is valid for behind 120 Hz and you can see that form pretty much almost any speakers measurements in the RT60 decay plots and as you will hardly find any for closed enclosure speakers you can instead use one's with ported speakers and such sub's (including mine) or even better sub's standalone comparative measurements regarding spotting the differences. Surely I don't have to find them for you at least that much you can do on your own if it interests you.
Well, since you refer to that infamous by SOS quoted paper lets compare its measurements of the NS-10 with a decent ported monitor of that time (the later Neumann successors of it have even better designed ports):

1723555200516.png

1723555217034.png

The spectral decay of the NS-10 shows to be only quicker below 70 Hz where the port of the K&H operates simply as it doesn't have such and also no serious bass too, though nothing that supports your above 120 Hz claim.
 
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Indeed. And it's not only a matter of assuring the mid range is well defined and balanced, it's also about makings sure your mix still works on other speakers with weak bass performance. It might take some extra work on the bass drum and bass sound to assure they don't get lost and still drive the song. When done well, this is not detrimental to the sound. That's what lots of people fail to understand.

Yes, good translation means the mix will be better for all sound systems. A well-balanced midrange will not just sound better on a bass-limited system, it will also sound better on a high-quality full-range system. It’s just that it can be harder sometimes to hear what is needed to make the midrange better as a capable low-end can somewhat compensate for a lacking midrange.

I have seen many people make the wrong assumption that making a mix better for a less bass-capable system will make it sound worse on the better system, but that wouldn’t make it a very good translation. :)
 
Translation is an art, a skill and a talent. Not just something you achieve by buying the right speakers or treating your room. Of course for me, a neutral room + speaker helps me trust my translation decisions faster, but there's more to it than that. I don't use NS-10s but I can see why a speaker that puts a magnifying glass on the midrange where our ears are most sensitive could help translation in the right hands. The number of esteemed mixes done on them is undeniable. There are other ways to achieve this obviously like using filters, checking on other systems / headphones, using metering etc. etc. etc.

This is also why the 'circle of confusion' seems like a flawed premise to me, at least as I understand it. This is art; there is no 'correct' amount of lows, mids, highs. There's no way of knowing what decisions were speaker coloration and what were artistic intent. There is no objective metric for success of 'translation' or what is the 'perfect mix'. All these things are going to vary wildly based on the artist, the genre, the producer and many more variables. And they SHOULD.
 
This is also why the 'circle of confusion' seems like a flawed premise to me, at least as I understand it. This is art; there is no 'correct' amount of lows, mids, highs. There's no way of knowing what decisions were speaker coloration and what were artistic intent. There is no objective metric for success of 'translation' or what is the 'perfect mix'. All these things are going to vary wildly based on the artist, the genre, the producer and many more variables. And they SHOULD.

Perhaps you are misconstruing the concept of "circle of confusion". This short article might explain it more clearly.


Jim
 
Perhaps you are misconstruing the concept of "circle of confusion". This short article might explain it more clearly.


Jim

Sort of; I definitely see the value in advocating for neutral monitoring (or a standard) so that decisions can be made with artistic intent as opposed to compensating for speaker or room deficiencies. Studios have been striving for these things for a very long time. I still just find it a bit reductive of the whole process of production and mixing.
 
Sort of; I definitely see the value in advocating for neutral monitoring (or a standard) so that decisions can be made with artistic intent as opposed to compensating for speaker or room deficiencies. Studios have been striving for these things for a very long time. I still just find it a bit reductive of the whole process of production and mixing.

You seem to totally understand the premise, but something in your heard won't let you accept it as fact. I'd be trying to figure out what that thing is. If you truly find it reductive you simply don't have the experience in production or mixing to see how true it really is.
 
Sort of; I definitely see the value in advocating for neutral monitoring (or a standard) so that decisions can be made with artistic intent as opposed to compensating for speaker or room deficiencies. Studios have been striving for these things for a very long time. I still just find it a bit reductive of the whole process of production and mixing.

Look at all the writers and musicians in the world today, and how they are threatened by AI. All their life, they've been perfecting their craft and honing their skills. Now there's a computer algorithm that can potentially take the wind out their sails. Is it not understandable that they resist it?

In the same way, is it possible that you find solutions to the "circle of confusion" reductive because the very premise threatens to diminish your artistic skills? Like writers and musicians, production teams depend on personal skills, developed over time and sometimes with a great deal of effort.

Just as free trade eventually destroyed the medieval guilds, so may progressive solutions in the recording industry destroy the dominance of personal artistry. Unfortunately, time and tide wait for no man.

Jim
 
Perhaps you are misconstruing the concept of "circle of confusion". This short article might explain it more clearly.


Jim

“Accurate reference monitoring” will not put an end to the “circle of confusion” by itself, the real solution is using well-known “accurate reference recordings” set to a controlled loudness level for regular comparisons and reality checks during the mixing of your song.

The reason why accurate reference monitoring by itself will not be enough is that the recorded content you are working on is not necessarily tonality-wise accurate to begin with. But still, your hearing will quickly acclimate itself to that tonality and set it as a norm, and when you add more tracks to the mix you will likely drift towards the tonality your hearing has already acclimated itself to, and before you know it the whole mix can be pretty far from having a neutral tonality without you even noticing it. This will happen no matter how accurate or not your choice of a studio monitor is, and the best way to set this straight is regular comparisons/reality checks with well-known accurate reference recordings to recalibrate your hearing back to accurate tonality.
 
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