Advice needed for active crossover setting on Yamaha NS1000M

[petarst]

Hello to everyone,

I would kindly ask if anyone here has Yamaha NS1000m hooked to the minidsp flex eight sytem? I have done it and set cross over sections to be 500Hz and 6000Hz. I do have a feeling that system is sounding bright and like the speaker is not balanced as it should be. Is there anyone who can help me with crossover settings? Thank you !

 

[Rick Sykora]

Welcome to ASR!

Normally would suggest you check diyaudio.com for this sort of request. However, suspect we can help here. Specifically, you should check out this thread from @dualazmak:

Multi-Channel, Multi-Amplifier Audio System Using Software Crossover and Multichannel-DAC

Edit: Please find here (on this thread) and here (remote independent thread post) the Hyperlink Index of this thread and some of my related posts in remote threads ************************************************** Edit on August 8, 2023 Jus for your reference,,, - The latest system setup of my...

www.audiosciencereview.com

 

[thewas]

If I were you I would do also distortion and off axis measurements of each driver.

The problem of many loudspeakers of that time is that the crossover point of the mid to the tweeter is chosen very high to ensure higher SPL robustness but compromising a smooth directivity and sound power. For ideal 3" and 1" pistonic drivers without waveguides the crossover frequency should be quite lower (around 3 kHz), but as said please check before if your tweeter allows that from distortion point of view at the SPLs you need.

 

[petarst]

If I were you I would do also distortion and off axis measurements of each driver.

The problem of many loudspeakers of that time is that the crossover point of the mid to the tweeter is chosen very high to ensure higher SPL robustness but compromising a smooth directivity and sound power. For ideal 3" and 1" pistonic drivers without waveguides the crossover frequency should be quite lower (around 3 kHz), but as said please check before if your tweeter allows that from distortion point of view at the SPLs you need.

Thank you guys. I am sorry because I post in the wrong forum section. Thewas your advice is to go for lower CF to be instead 6kHz to be 3Khz, but to do a test via microphone, measurement right? I am new in this and I have never do any kind diy project

 

[dualazmak]

Thank you for your possible kind attention and interests on my project thread. You can find my latest total system setup here post #774 on my project thread.

Please note in my DSP crossover configuration I set -3 dB for midrange Fq zone of 500 Hz - 6 kHz. And, furthermore, I have implemented further flexible relative-gain (tonality) tuning flexibilities in analog level by using four "integrated amplifiers";

As you may already well aware of, the Beryllium dome midrange driver YAMAHA JA-0801, as well as the Beryllium dome tweeter JA-0513 are rather "highly efficient" drivers, and therefore in the original intact NS-1000 and NS-1000M, the "normal position" of the two attenuators is actually/physically about -3 -4 dB suppression configuration.

You would please very carefully read my post #248 and #251 on my project thread for complete elimination of attenuators and use of "parallel" 20 Ohm or 22 Ohm tuning resistors in SP high-level circuits of midrange JA-0801 and tweeter JA-0513 (and also for super-tweeter FOSTEX T925A, in my case).

I would like to highly recommend you to utilize/implement analog-level relative gain tuning capabilities/freedom by using either or combination of;

1. Use analog output gain controllers (if available) within your multichannel DAC

2. Use HiFi-grade pre-amplifiers between your multichannel DAC and power amplifier for flexible on-the-fly (while listening to music) relative gain (tonality) control/tuning

3. Use HiFi-grade "integrated amplifiers" (like in my setup based on "right-person-in-right-place" manner, ref. here #435) for flexible on-the-fly (while listening to music) relative gain (tonality) control/tuning

I assume my recent post here would be also of your interest and reference.

 

[SDC]

Flex eight for dsp works perfectly.

I've used it in high sensitivity beryllium compression drivers. No noise problems.

Shd paired with fusion plate amp for t25d tweeter, M74A midrange also had no problem.

Minidsp are doing decent job with their updated products.

If there is noise, it is probably amplifer or ground loops, so many potential sources.


1. Use low cut passive filters for high and mid driver protection.

Exotic speaker drivers are hard to replace. So it is always safe than sorry.

Accidentally changing the DSP preset of flex can kill the driver in split second.


2. Use measurement microphone.

This is must. Umik is good enough, earthworks or other variants are better, but it depends how much filter you want to apply.

Ruler flat responses with quasi-anechoic measurements will require better calibrated microphones than factory Umik.

But crossover filters, some notches, and shelving filters for gently smoothing the FR, simple Umik will suffice.


3. Don't push the drivers to hard.

Low crossover frequ and high slope can push the drivers to it's limit.

It's no good for the driver. And sound quality don't magically improve with lower XOs.

What matters is choosing the right point and slope for desired properties.

Using Vituixcad and multi presets of Minidsp helps with that, giving user more flexibility with less work.



For amplifier selecting, I'll prioritize reliability, noise performance second.

 

[thewas]

Thank you guys. I am sorry because I post in the wrong forum section. Thewas your advice is to go for lower CF to be instead 6kHz to be 3Khz, but to do a test via microphone, measurement right? I am new in this and I have never do any kind diy project

Since you like say you seem to have no experience in this field I can recommend you two ways, you either get yourself a basic book about loudspeaker design and measurements (or read a lot in the corresponding forums) or try finding someone to do it for you as otherwise you not only you won't beat but won't even reach the performance of the original crossover.

 

[dualazmak]

1. Use low cut passive filters for high and mid driver protection.

Exotic speaker drivers are hard to replace. So it is always safe than sorry.

Accidentally changing the DSP preset of flex can kill the driver in split second.

Fully agree with you.

In addition to DSP's XOs with high-pass (low-cut) filters, I too have several safety measures including the protection capacitors (low-cut) for midranges JA-0801, tweeters JA-0513 and supertweeters FOSTEX T925A as shown in the diagram shared in my above post #5 (for the details refer to #184, #485, #886, #890).
Please also refer to;
- Renewal of SP cabling boards beside SP systems: #906

Spoiler

I also very carefully start-up and shut-down my whole audio system (ref. #776).

Spoiler

Standard "Startup/Ignition Sequences":
01. Wake-up the AC Circuit Breaker (100V/15A, if it has been shutdown)
02. Power-on the root switch of 24-port AC power strip (series of four 6-port strips)
______This automatically wakes up DIY-12-VU-Meter-Array, KORG DS-CAC-10, BEHRINGER DS2800
03. Power-on YAMAHA A-S3000 after making sure the volume is at -infinity dB (no sound)
04. Power-on ACCUPHASE E-460 after making sure the volume is at -infinity dB (no sound)
05. Power-on SONY TA-A1ES after making sure the volume is at -infinity dB (no sound)
06. Power-on YAMAHA A-S301 after making sure the volume is at -infinity dB (no sound)
07. Power-on L&R sub-woofers YAMAHA YST-SW1000 by remote controller (preset volume 14:00 o'clock)
08. Power-on OKTO DAC8PRO; make sure the volume (master gain) is less than -75 dB
09. Power-on Windows 11 Pro PC
10. Check & confirm wake-up and running of ASIO BRIDGE at taskbar
11. Check & confirm wake-up and ready of DIYINHK ASIO at taskbar
12. Check & confirm all of the Window's sound I/Os (KERNEL, WASAPI, WMD, Mic, etc.) are muted-off
13. Launch JRiver MC (or Roon); make sure the master volume is at about 5 % (-77.5 dB)
14. Start playing JRiver the 6th track of my music sampler/reference playlist (piano solo)
15. Slightly volume-up JRiver to around 10 % (-55.0 dB); check the JRiver's small VU-EQ meter is properly dancing
16. Launch software DSP EKIO
17. Open/load the EKIO's standard configuration file into EKIO
18. Start playing EKIO
19. Check and confirm that all of the EKIO's I/O VU meter bars are properly moving in very low gain
20. Volume-up A-S3000 to 11:55 o'clock (-18 dB)
21. Volume-up E-460 to -17.0 dB
22. Volume-up TA-A1ES to -17.0 dB
23. Volume-up A-S301 to 09:45 o'clock (ca. -19 dB)
24. Volume-up DAC8PRO to -4 dB; check to properly hear the sound in very small volume
25. Carefully and slowly volume-up JRiver MC (as master volume) to usual listening volume using mouse wheel

Standard "Shutdown Sequences":
01. Stop playing JRiver MC (or Roon)
02. Volume-down the "Master Volume" of JRiver MC to less than 5% (-77.5 dB)
03. Stop playing software crossover EKIO
04. Volume-down OKTO DAC8PRO to -99 dB
05. Volume-down YAMAHA A-S301 to -infinity dB (no sound)
06. Volume-down SONY TA-A1ES to -infinity dB (no sound)
07. Volume-down ACCUPHASE E-460 to -infinity dB (no sound)
08. Volume-down YAMAHA A-S3000 to -infinity dB (no sound)
09. Power-off A-S301
10. Power-off TA-A1ES
11. Power-off E-460
12. Power-off A-S3000
13. Power-off L and R active sub-woofers YAMAHA YST-SW1000 by remote controller
14. Exit/Shutdown EKIO
15. Exit/Shutdown JRiver MC
16. Shutdown Windows 11 Pro PC
17. Soft-shutdown (Mute-off) DAC8PRO by remote controller
18. Switch-off the root switch of 24-port AC power strip (series of four 6-port strips)
______This automatically shutdowns DIY-12-VU-Meter-Array, KORG DS-CAC-10, BEHRINGER DS2800
19. Down the AC Circuit Breaker (100V/15 A, if needed for long shutdown; longer than a week or so)

Although these may look somewhat complicated at your first glance, I usually complete the startup/ignition) sequences in less than two minutes, and I complete the shutdown sequences in less than one minute.

Furthermore, for the tuning in DSP configuration, I mean any change/modification of DSP parameters, I also take utmost careful procedures in order to avoid possible damages to SP drivers, as I recently shared here.

Spoiler

I wrote there:
I know that XOs (many parameters), time-alignment, phase tuning, EQ and gain in DSP would be more-or-less interdependent with each other, right?
None of them can be changed/modified completely independently.
On the other hand, analog-level gain controls do not affect the upstream DSP configurations.

One of the other pros of HiFi analog-level relative gain tuning (tonality control) by knobs/dials would be it is very safe on-the-fly compared to on-the-fly DSP gain control.

In case if you would like to do it in DSP, especially on-the-fly, you always have possibility of mis-adjustment of gains, i.e. and e.g. 20 dB boost instead of intended 2.0 dB boost by numerical keyboard mistyping the value (and/or it would happen even using mouse wheel up-and-down in some DSP software tools) which may harm and/or destroy your precious SP drivers.

Consequently, I seldom change DSP parameters (especially relative gains) on-the-fly, while listening to music track; I always stop playing and set the master volume in digital music player to minimum (minus infinity dB) position when changing the DSP parameter(s), and after the DSP modification and after start playing music, I very carefully and slowly gain-up the master volume (in my case JRiver MC) to check the given change in DSP parameters; in this way, I can avoid possible (but rare) harm/damage to SP drivers.

For further safety purposes, I also use protection capacitors for my treasure Be-midranges, Be-tweeters and metal-horn super-tweeters (summary ref. here).

I use DSP EKIO, and EKIO has very nice gain up-and-down by mouse wheel rotation (0.1 dB granularity); I have been always strongly recommending, therefore, to use mouse wheel rotation (not keyboard numeric typing), if you use EKIO and would like to do on-the-fly relative gain control (e.g. ref. here).
(Mr. Guillaume BADAUT of LUPISOFT who develops EKIO has very quickly implemented the excellent mouse wheel operations responding to my sincere request.)

 

[MAB]

Treols Gravesen reverse-engineered the NS-1000
Read his thread, he has a detailed discussion of the crossovers and speaker response, with measurements and simulations.

Yamaha-NS1000

The speakers tend to sound lean in many rooms. No surprise given sealed design and the woofer response,.

The resulting on-axis measurements:

This speaker has a tendency to sound bright in my experience too. This was a common feature (or flaw) in Yamaha speakers. The NS-1000 not so bad, and placing them close to the wall, combined with attenuating the mid and tweeter will help.

Troels developed a new passive crossover. I would read his discussion. It's a good read, with measurements and simulations, and a good starting point for an active design in my opinion.

Yamaha-NS1000

His mods address some of the choices Yamaha made and should result in a speaker that isn't as intrinsically bright, but still leaves me wondering about the bass response. Here is his final in-room response:

The left graph is 50cm from the front-wall, the right graph is only 27cm from the wall. I think this bass can be improved with DSP, like a Linkwitz-transform.

You should seriously consider integrating a capacitor on both the tweeter and the midrange to preventing damage. You can either use large values as protection capacitors, or you can use values that combine with the DSP filters to hit a target slope. If you integrate them, you will want to keep the circled caps (albeit different values based on your DSP settings, as well as taking into account the fact that you will likely bypass the rest of the components and the padding, which will change the crossover point.

I like your idea. I have been thinking of doing something like this to a NS-1000 as well. But I do want to caution, these are rare speakers, Yamaha made some odd design choices that can be improved on for sure. But that improvement is going to take some measurements and knowledge/experience to make better. I think the midrange are special, the woofer and tweeter are great but unremarkable. But the midrange looks like it has some odd resonances, and it would be good to measure and address. This is not exactly a first-time DIY project. But I applaud your enthusiasm. If you want to proceed, I would copy Troels' topology to start with.

Hope this helps.

edit: added some missing words from a sentence.

 

[dualazmak]

Yes, I too shared the "normal/intact" -3 -4 dB attenuation for Be-midrange JA-0801 and Be-tweeter JA-0513 by the original attenuators in my above post #5 (for the details ref. here #248 on my project thread).

I also have shared this post on my project thread:
- Useful web articles on Yamaha NS-1000 and NS-1000M: #084

Spoiler

I assume everyone here is curious about my fascinations and enthusiasms on YAMAHA NS-1000 (not NS-1000M) and its SP units. Even though you can find so many review articles on NS-1000 and NS-1000M, here let me just inform you one rather recent review article;
https://www.hifinews.com/content/yamaha-ns-1000m-loudspeakers
https://www.hifinews.com/content/yamaha-ns-1000m-loudspeakers-page-2
https://www.hifinews.com/content/yamaha-ns-1000m-loudspeakers-sidebar-family-values
https://www.hifinews.com/content/yamaha-ns-1000m-loudspeakers-lab-report

The SP units are common for NS-1000 and NS-1000M. Little people knows, however, the difference between NS-1000 and NS-1000M; there are considerable differences in enclosure thickness and weight;

As described in my previous post, I have fully renovated the SP terminals and LC-network+attenuators of my NS-1000 to be ready for this multi-channel, multi-amplifier project.

Edited to add at 23:30 May 4, 2020 Japan time;
One of my audio-enthu friends in Japan kindly informed that these two links are also nice for your kind notice on NS-1000 and NS-1000M;
http://www.troelsgravesen.dk/Yamaha-NS1000.htm
http://www.thevintageknob.org/yamaha-NS-1000.html

Edited to add at 08:22 July 5, 2020 Japan time;
https://liquidaudio.com.au/legendary-yamaha-ns-1000m-loudspeaker-review/

 

[dualazmak]

Dr. Floyd Toole's brief comment on NS-1000 dated July 14, 2019...

 

[dualazmak]

Another short comment of Dr. Floyd Toole on NS-1000M dated July 31, 2021 can be find here:

Spoiler

Dr. Toole wrote there:
.....
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).

I could go on, but fortunately I don't need to because the recording industry has done it for me; it has moved on. The fad has substantially passed, and the current norms for monitor loudspeakers (including Yamahas) are not different from the current objectives for neutral sounding domestic entertainment loudspeakers, which is as it should be.

 

[petarst]

Treols Gravesen reverse-engineered the NS-1000
Read his thread, he has a detailed discussion of the crossovers and speaker response, with measurements and simulations.

Yamaha-NS1000

The speakers tend to sound lean in many rooms. No surprise given sealed design and the woofer response,.
View attachment 359027

The resulting on-axis measurements:
View attachment 359028

This speaker has a tendency to sound bright in my experience too. This was a common feature (or flaw) in Yamaha speakers. The NS-1000 not so bad, and placing them close to the wall, combined with attenuating the mid and tweeter will help.

Troels developed a new passive crossover. I would read his discussion. It's a good read, with measurements and simulations, and a good starting point for an active design in my opinion.

Yamaha-NS1000

His mods address some of the choices Yamaha made and should result in a speaker that isn't as intrinsically bright, but still leaves me wondering about the bass response. Here is his final in-room response:

View attachment 359030
The left graph is 50cm from the front-wall, the right graph is only 27cm from the wall. I think this bass can be improved with DSP, like a Linkwitz-transform.

You should seriously consider integrating a capacitor on both the tweeter and the midrange to preventing damage. You can either use large values as protection capacitors, or you can use values that combine with the DSP filters to hit a target slope. If you integrate them, you will want to keep the circled caps (albeit different values based on your DSP settings, as well as taking into account the fact that you will likely bypass the rest of the components and the padding, which will change the crossover point.
View attachment 359031

I like your idea. I have been thinking of doing something like this to a NS-1000 as well. But I do want to caution, these are rare speakers, Yamaha made some odd design choices that can be improved on for sure. But that improvement is going to take some measurements and knowledge/experience to make better. I think the midrange are special, the woofer and tweeter are great but unremarkable. But the midrange looks like it has some odd resonances, and it would be good to measure and address. This is not exactly a first-time DIY project. But I applaud your enthusiasm. If you want to proceed, I would copy Troels' topology to start with.

Hope this helps.

edit: added some missing words from a sentence.

Thank you all of you for everything you have done here for me. I want to avoid any parts from my amps to the speakers therefore i go with active solution.
How can I damage them, I don't get it? I will user Troels values and I will see what will happen.

 

[MAB]

Thank you all of you for everything you have done here for me. I want to avoid any parts from my amps to the speakers therefore i go with active solution.
How can I damage them, I don't get it? I will user Troels values and I will see what will happen.

I didn't understand what you man by 'avoid any parts from my amps to the speakers'.

If you hook an amp directly to a speaker, any DC offset, any turn-on or turn-off noises, and any connection mistake, or filter mistake (like accidentally pushing a button on the MiniDSP filter console), and you no longer have working tweeter or midrange. I hope this makes sense. If not, you really need to take a step back and consider if you want to start learning with a rare NS-1000.

 

[petarst]

I didn't understand what you man by 'avoid any parts from my amps to the speakers'.

If you hook an amp directly to a speaker, any DC offset, any turn-on or turn-off noises, and any connection mistake, or filter mistake (like accidentally pushing a button on the MiniDSP filter console), and you no longer have working tweeter or midrange. I hope this makes sense. If not, you really need to take a step back and consider if you want to start learning with a rare NS-1000.

Ok i didnt know that...

 

[dualazmak]

I didn't understand what you man by 'avoid any parts from my amps to the speakers'.

If you hook an amp directly to a speaker, any DC offset, any turn-on or turn-off noises, and any connection mistake, or filter mistake (like accidentally pushing a button on the MiniDSP filter console), and you no longer have working tweeter or midrange. I hope this makes sense. If not, you really need to take a step back and consider if you want to start learning with a rare NS-1000.

I essentially (and generally) agree with you.

After I carefully reviewed all the past posts on this thread, I assume and it seems that OP @petarst is still in the learning process of fundamentals of DSP configurations especially the XO=crossover filters as well as several "safety" measures/implementations (ref. my above post #8).

In case if he would be still really serious in stepping into DSP-based active multichannel setup with his (treasure!) NS-1000M, I highly recommend him first to test/practice/train-himself using other (affordable/existing, from used market?) 3-way SP unit(s); even though at least in Japan we still can get fairly in-good-shape SP drivers of NS-1000 and NS-1000M in second-hand (auction) markets.

Just for reference and learning materials for OP @petarst (and for other DSP-naive people looking at this thread), as an example case of DSP multichannel multi-amplifier active audio setup using Yamaha NS-1000, let me share again my DSP (including XOs) configuration under the below spoiler cover.

Spoiler

You would please very carefully read through my post here #774 on my project thread for the details of my latest active setup with Yamaha NS-1000. The essences in DSP configuration (in my case I use "EKIO") can be summarized in these two diagrams.

In any way, I do hope much good lucks for OP's near-future progress in DSP-based multichannel active audio setup possibly using still-amazingly-excellent Yamaha NS-1000M.

 

[AuricGoldfinger]

If you're implementing an active crossover it will require some knowledge of the drivers' response behaviors, or the electrical crossover's response. I can do both, but your drivers will vary from mine or anyone else's. I could characterize the output curves from the crossover using resistors or the drivers, and this will put you in the ballpark to get the attenuation and slope correct. Measuring with the drivers of course accounts for the inductive rise in impedance with increasing frequency, which makes the slope shallower.

A couple warnings:

1. Do not cross the tweeters lower. I'll add that the tweeter has a resonance in the region of 2.4-3kHz. With the 6kHz crossover in place it's negligible and doesn't impact the time domain measurements. If the tweeter is crossed lower, it will lead to an increase in harmonic distortion. If you are going active, then a notch filter can be centered on the resonance, but it's not absolutely needed. Should you wish to pursue controlling the energy there, you'll need to determine exactly where to place the notch and how much attenuation your tweeters require to achieve the desired acoustic slope. As you notch the tweeter, watch the phase, and the midrange low-pass (high frequency response of the mid) will have to be carefully raised by a small amount to bring the acoustic slopes together for a flat merge.
2. Add a protection device to each tweeter and midrange. I used ultra-fast blow solder-in type fuses, 0.5A. They contribute only half a decibel of attenuation. Since you are going active, absolutely use a capacitor on each tweeter and mid driver, even if you read elsewhere it's not needed.
3. I can't recommend using someone else's passive "upgrade crossover" in this speaker. The understanding is you are going active, but I just want to clarify this point. I've tested many of these drivers spanning different production years, and they varied by some margin in impedance modulus, sensitivity and response, to such extent that its not much wonder people have widely differing views of the model. An upgrade crossover is not perfectly one size fits all when drivers vary by 4dB. I found it interesting Yamaha had different capacitor values for their tweeter in different schematics for this speaker model over the years, and that may have been to account for the different impedance and response. If an owner is going to outlay for a new crossover they should accommodate their drivers to trim everything up evenly using a new measurement system. I found the tweeter and mid with the original crossover were time coincident, converging nicely in step response as a single event. The upgrade crossovers out there were made at tweeter height, ignoring the time coincidence.

 

[petarst]

Thx mate, i am planning to order Troel Gravesen crossover and i am going passive with biamping. But now I am reading what you say...maybe it is the best to sell them with their oroginal crossovers...what else to do...when the things are so complicated with them...

 

[AuricGoldfinger]

No worries, just trying to straighten any winding paths.

I guess my question is, what problem are you trying to solve? What is it that you feel needs fixing?

I think if you are in search of a tangible payoff, then biamping as you mention, with an active crossover and some bass extension EQ on the woofer may be worthwhile. I would set the original crossovers aside and leave them unmodified so that whatever direction you take the speaker can still be put back to how it was in the beginning. This can be very good for comparisons and later measurements.

It only looks difficult and you just need some learning before engaging a speaker like this. It uses a very simple second-order crossover with drivers having first and near second-order acoustic slopes. The combined response is quasi-Linkwitz Riley and was probably a conscience choice as the tweeter and midrange are time coincident with the listener height near the middle of the baffle. To further explain, L-R crossovers are used in time-aligned speakers and that is what you have here in the mids and treble. The midrange is brilliant, it incorporates acoustic baffle step compensation thus making the crossover simpler and avoiding more complexity. You just need some test equipment and a practice speaker to learn on. If you’re concerned that you may damage them and would regret it, I would then suggest leaving them as they are.

I wouldn’t use that upgrade crossover on the Troel’s site and I could fill a page with reasons why. To summarize, that upgrade crossover was made while ignoring the coincident response of the original speaker, the mic sitting at several dozen degrees positive on the Y axis. I see he has updated his measurement now a decade later, but no word on any correction to his design. It also reduces the sensitivity; his before vs after graphs should have been done in the same conditions but weren’t; the tweeter response is very suspect as it’s nowhere near that flat in reality and can’t possibly be so with only a single notch filter added; the tweeter is shown -10dB at 15kHz in the upgrade graphs but not so in the before upgrade graphs, yet I see nothing in the crossover to achieve that (some odd mic placement may have been used); and the kit’s cost is largely inflated by audiophile woo capacitors.

Several people who purchased that kit later reverted back to the original Yamaha crossover, and with some searching you may find those discussions. Their impressions were that once the initial “wow, this is mellow and deep” wore off, the Troels crossover had negatively impacted the soundstage and detail. I’ll also note that my set of NS-1000M have much greater extension compared to the set he based his design on. I set my levels to normal, and if anything, all the speaker might benefit from is a few dB of flattening from 300-900Hz. Nothing else.

 

[spiral scratch]

all the speaker might benefit from is a few dB of flattening from 300-900Hz. Nothing else.

Shouldn't that be achievable with the pots on the front? And thanks for sharing your insights on this speaker. I picked up a pair last fall and have been having some fun with mine.

To the OP - if you're doing active you should be measuring frequency response. And I think part of what auricgoldfinger's saying is there's no one size fits all approach here. So chucking it all in to just get a different set of speakers might not get you any further ahead. I would think one of the benefits of going active is getting proper room response which is achieved by measuring (and a lot of trial and error). They're nice speakers, if you like the way they sound I would stick with it. Adding some protection for the drivers is a good idea unless you live in Japan, the replacements are pretty expensive.