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Building a 2-way small active speaker with software crossover

Now that's very cool.

The engineer deserves serious respect. :)

I edited and corrected "that part" of my writing which now should be read as "But in 2019, I have confirmed it.....". It was just before I started my multichannel fully active exploration.

Of course, he added "But it (make it in fully active) would be all your responsibility! Of course any warranty and even any possible repair service (as whole SP system at YAMAHA factory) with charge will disappear for you!!", and I accepted that.

I carefully incorporate protection capacitors for my treasure midrange-squawker, tweeter, super-tweeter, and also always perform careful ignition/startup sequences and shutdown sequences (ref. here).;)

Very fortunately, since so many units of NS-1000 and NS-1000M have been sold in Japan, I still can rather easily find and purchase used but in good shape (or even excellently refurbished) SP drivers on several web sites for my NS-1000, if needed.
 
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I believe it does have inferior effect on sound quality by "change with age"; especially if electrolyte capacitor(s) are used, and also "the heat" gradually damage the attenuator(s) with year-time.
But that is just the change of the crossover function due to change of the values of some deteriorated parts and not an inherent issue that the "heat" produced at a passive crossover worsens the sound quality.

Details of my DSP EKIO XO/deay configurations (ref. here);
You didn't answer though if those XO configurations were all enabled when you did those woofer transient comparisons?
 
I carefully incorporate protection capacitors for my treasure midrange-squawker, tweeter, super-tweeter,
I assume you carefully measure the capacitor effect then. The capacitor added into the SW crossover chain definitely changes both amplitude and phase responses. This may have an effect to resulting phase at the crossover frequency.
 
But that is just the change of the crossover function due to change of the values of some deteriorated parts and not an inherent issue that the "heat" produced at a passive crossover worsens the sound quality.
Yes, you are right. I was talking about cons of LCR-network and attenuators "in total" and "over years".;)

You didn't answer though if those XO configurations were all enabled when you did those woofer transient comparisons?
Sorry, in my post #504, I was simply "forgetting" and did not clearly declare that "all the DSP EKIO configuration is active and applied".
 
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I assume you carefully measure the capacitor effect then. The capacitor added into the SW crossover chain definitely changes both amplitude and phase responses. This may have an effect to resulting phase at the crossover frequency.

Thank you for your reminder on this point.

Yes, firstly I intensively measured the gains (amplitude) before and after the protection capacitors as shared here #402 and #485.

And, all of my time alignment measurement and tuning summarized in my post here #520 were performed with those protection capacitors.

I do not have capability of precisely and objectively "measure" the phase responses, but I am now very much satisfied with the total sound quality including amazing stereo image (and wonderful feeling/sensation of "SP disappearance") with present setup (ref. #774, #520 and #687) in my room environment/acoustics.

Of course, I always subjectively check and confirm it by listening to my consistent "audio sampler/reference music playlist" (summary ref. here and here) and some of the tracks of "SONY Super Audio Check CD" (ref. here). I believe it is critically important having our own consistent "audio sampler/reference music playlist" throughout our step-by-step audio exploration project.

I usually feel much frustrations that I have little capability of providing further objective data/expressions describing the present amazing total sound quality of my system, but, as always, I can say finally "If you would have chance to be in Tokyo Metropolitan Area in the near future, please come to my listening room for our enjoyable music listening sessions and audio discussions!"
 
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Now that's very cool.

The engineer deserves serious respect. :)
Unless it isn’t true. Honestly, this statement echos around the industry. But I have yet to see measurable distortion differences in active vs. passive that are intrinsic. I have seen and made measurements of reduction in distortion in specific implementations (like a notch for controlling cone breakup modes). In fact, one study referenced in this thread actually shows dramatically reduced distortion with passive, so a perfect voltage source does not guarantee low distortion. In fact, the opposite in the case of controlling breakup modes.

Maybe the study exists that shows a passive filter has intrinsically higher distortion compared to equivalent active, across the board. I’m just not well read or done enough myself.

It does seem that it could be settled with a straightforward set of measurements on a range of typical passive filters vs the active equivalent circuit.
 
Unless it isn’t true. Honestly, this statement echos around the industry. But I have yet to see measurable distortion differences in active vs. passive that are intrinsic. I have seen and made measurements of reduction in distortion in specific implementations (like a notch for controlling cone breakup modes). In fact, one study referenced in this thread actually shows dramatically reduced distortion with passive, so a perfect voltage source does not guarantee low distortion. In fact, the opposite in the case of controlling breakup modes.

Maybe the study exists that shows a passive filter has intrinsically higher distortion compared to equivalent active, across the board. I’m just not well read or done enough myself.

It does seem that it could be settled with a straightforward set of measurements on a range of typical passive filters vs the active equivalent circuit.

Not what my comment was about.

I am referring to the contribution of the engineer who worked on those famous loudspeakers. He must have been with Yamaha in 1974 when they released the NS-1000 and still with them in ~1990 when the NS-1000X and NS-2000 were released. A long career producing/designing legendary speakers.
 
About speaker measurement:
I started with IMP (anybody know this ?), performing an, as the name suggests, impulse that was transformed via FFT to frequency response. An upgrade after time was a MLSSA module.
Basically this was and is still sufficient for measuring from above 300 or 400 Hz for 2 way speaker's crossover.
Neither REW nor Hypex provide this out of the box but prefer sinus sweeps, what is ok on it to also adjust to room acoustics improvement.
But for me I prefer to get second opinion by MLSSA via Arta.
If there is congruence I'm fine, if not have to dig a bit.
 
I am referring to the contribution of the engineer who worked on those famous loudspeakers. He must have been with Yamaha in 1974 when they released the NS-1000 and still with them in ~1990 when the NS-1000X and NS-2000 were released. A long career producing/designing legendary speakers.

These engineers (it seems more than one) had even visited Toole at the NRC:

BTW, the designer of the NS-10M and the NS-1000M visited me at my NRC lab in Canada to experience the measurement process and double-blind listening tests. They left with many physical measurements and photographs intending to duplicate some of the facility and processes. The original speakers were designed to exhibit flat sound power (believed, incorrectly, to be what listeners heard in the far field), which my measurements showed they did extremely well. The problem was that the two-way NS-10 ended up with a very non-flat on-axis response, but the three-way NS-1000M, with more uniform directivity with frequency, was an exemplary loudspeaker at the time (1974) - see Figure 18.3 (e).
From: https://www.audiosciencereview.com/...al-music-pros-using.12225/page-10#post-863542

The designers of the NS1000M and NS10M visited me in my lab at the National Research Council in Ottawa. They were totally competent engineers, who simply got mislead by the then popular "flat sound power" target promoted by a few east coast US designers. Measurements shown in my book indicate that they hit the target, and it worked reasonably well for the beryllium-domed three-way NS1000M (which needed only a bit of bass boost to sound much improved) but failed badly for the 2-way NS10M. Ironically the NS10M went on to become a popular near-field monitor among some recording engineers. It was absolutely not neutral, exhibiting a massive midrange hump, but engineers thought of it as a facsimile of crappy consumer loudspeakers; an Auratone with some bass. Now even cheap and cheerful consumer speakers (especially active ones) are sounding much better, yet one sees NS10Ms on console bridges the world over. Habits don't die easily, but there are signs that this one is fading. But then a manufacturer, British I think, has just started manufacturing a replica. Will it never end . . . ?

After the visit Yamaha monitors adopted the flattish on-axis performance target.
From: https://www.audiosciencereview.com/...t-design-available.19024/page-28#post-1417201
 
Not what my comment was about.

I am referring to the contribution of the engineer who worked on those famous loudspeakers. He must have been with Yamaha in 1974 when they released the NS-1000 and still with them in ~1990 when the NS-1000X and NS-2000 were released. A long career producing/designing legendary speakers.
Fair, and I agree with that part.
But would still love to see supporting data. And the alleged exchange has no context. And, this thread is about building an active speaker, and I was hoping you (anyone) might have data.
 
Yeah, we need to remember how bad some of these old (odd) design choices could sound.
It does illustrate how good engineers can design to, and hit, a bad target. Yamaha seemed quite focused on the beryllium drivers, but great drivers don’t make a great speaker. These comments from Toole and are very useful and might explain why I didn’t care for the old Technics SB-M1 when I heard them years ago. I’m a fan of Technics, and have a much less desirable pair than those SB-M1, but they also seem to be designed to flat sound power target.
 

It fits the classic East Coast vs West Coast sound of US made speakers and the more refined and detailed Japanese sound of late 1970s and 1980s speakers, which was, to my tastes much more accurate and enjoyable.

Unfortunately, many UK loudspeaker designers (before they all got bought out) started following the boom-tizz JBL/Harman sound in the 80s and 90s. A notable exception was many of the early 80s Infinity acoustic suspension/sealed designs using the famous Emit tweeters and poly cone bass drivers. I actually liked their sonic signature.

At least some manufacturers are starting to rediscover the dedicated midrange driver again. Good thing in my book because we've had 25+ years of missing midrange detail IMO.
 

Oh, just amazing...

I really thank you so much for your sharing those information which greatly encourage (and support?) my efforts (ref. my project thread and the latest setup here) in my DSP-based multichannel multi-amplifier audio exploration using renovated (fully active setup) YAMAHA NS-1000 (definitely not NS-10M)!
 
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About speaker measurement:
I started with IMP (anybody know this ?), performing an, as the name suggests, impulse that was transformed via FFT to frequency response. An upgrade after time was a MLSSA module.
Basically this was and is still sufficient for measuring from above 300 or 400 Hz for 2 way speaker's crossover.
Neither REW nor Hypex provide this out of the box but prefer sinus sweeps, what is ok on it to also adjust to room acoustics improvement.
But for me I prefer to get second opinion by MLSSA via Arta.
If there is congruence I'm fine, if not have to dig a bit.
I use both Arta and REW, and also response to a single rectangular impulse of 10 - 20 us. It was my University electroacoustic lecturer Professor Merhaut who was a promoter of the impulse method back in the seventies. In room conditions, a rectangular impulse method suffers from ambient noise and gives too low resolution. There is not enough low frequency energy in the single impulse. Comparing MLSSA and REW sweep, REW gives the best results. It also works with impulse response calculated back from the sweep.
 
A perfect voltage source does not guarantee low distortion. In fact, the opposite in the case of controlling breakup modes.
I think it is case related, strictly. I also have some measurements showing the effect of current drive to lower distortion. However, a true current drive, which is usable only in case of the active speaker ;).

 
Nice continuous promotion of your project and approach ;):D.

I am sorry and sincerely apologize that I wrote too often and too much on your nice thread here which sometimes made your precious train out of your right rail.

Now I will be gradually going to be in ROM (read-only member) league here, and will return to my project thread, even I know it maybe already too late, sorry again.:facepalm:

My next topic on my thread will be precise calibration of my gem ECM8000 microphone (Made in Germany, 2008) for which you kindly pointed (ref. here #809, here #813, here #816 and here #819).

You would please kindly understand that I am still learning a lot on this thread from your invaluable efforts and discussions so far intensively and generously shared with us.
 
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I am sorry and sincerely apologize that I wrote too often and too much on your nice thread here which sometimes made your precious train out of your right rail.

Now I will be gradually going to be in ROM (read-only member) league here, and will return to my project thread, even I know it maybe already too late, sorry again.:facepalm:

My next topic on my thread will be precise calibration of my gem ECM8000 microphone (Made in Germany, 2008) for which you kindly pointed (ref. here, here and here).

You would please kindly understand that I am still learning a lot on this thread from your invaluable efforts and discussions so far intensively and generously shared with us.
No problem and no need to apologize :).
Hachiju no tenarai :).
 
I have been playing with bass response a bit. The result is an extension of low frequency range, with still acceptable distortion at the listening level. Near field measurements, SPL absolute value not calibrated.

EKIO bass response1.png


EKIO bass response distortion.png


28_09_2023 Bassboost.png


Interestingly, the VituixCad optimizer did not work in this case and the iteration of shelf filter parameters had to be done manually by trial/error method.
 
You may want to drill a hole in it with some rainpipe to check out a reflex config. Would be interesting to see the difference in distortion. Probably like 60 or 70 Hz tuning would work well.
 
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