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Hello OP @S Moore,

Until today, after your kick-off of this interesting thread, I have been being a careful read-only-member on this thread. Now, please let me participate a little bit having my primitive and general suggestions.

First of all, your present approach seems to be very much similar to my almost-five-year exploration journey (thread) of PC-DSP-based multichannel multi-SP-driver multi-amplifier fully active audio system where I converted passive 3-way SP system into fully active configuration with adding L&R super-tweeters and L&R subwoofers, of course in fully active way.

For your quick understandings on my almost-completed 5-way 10-channel active system, you can find the details of my latest system setup in my post #931 on the project thread. __Furthermore, for your overview perspective and "search" purposes, you can find here and here (exactly same contents) the hyperlink index for my project thread.

I assume these posts and threads under the below spoiler cover would be of your reference and interest:
- In depth insights on SP attenuators and their elimination in multichannel system: #248, #251, #99(remote thread), #100(remote thread), #101(remote thread)

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-6_Summary, discussions, and a little step forward: #404, #405-#409

- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497, #503, #507

- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520
- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687

- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

- A nice smooth-jazz album for bass (low Fq) and higher Fq tonality check and tuning: #910, #63(remote thread)


Thread: Music for Testing Treble (High Frequency) Sound

In this post, I would like to just personally suggest the following only two points.

1. You would please consider (and at least test/evaluate) the relative gain (tone) controls between the SP drivers not only in digital domain using DSP but also in analog domain using HiFi grade pre-amplifiers and/or integrated-amplifiers to enable optimal safe flexible combination and pros; I wrote in my post #931 as follows under the below spoiler cover.
Here in this post (#931), please let me emphasize again about the pros and merits of relative gain (i.e. tone) control not only in digital domain but also in analog domain using pre-amplifiers or integrated-amplifiers (in my setup). I recently wrote again in my post #56 on a remote thread like these;
Yes, as for safe and flexible tone controls (or I can say "relative gain controls among the multiple SP drivers"), my stance (policy) at least, is that we are encouraged to utilize the "best combination" of "DSP configuration in digital domain" and "analog domain tone controls using HiFi-grade preamplifiers and/or integrated amplifiers".

We need to note (and to respect for) that analog domain tone controls (relative gain controls among the multiple SP drivers) give no effect nor influence at all on the upstream DSP configuration (XO/EQ/Gain/Phase/Polarity/Group-Delay). I believe that this is a great merit of flexible tone controls in analog domain. We know well, on the other hand, in case if we would like to do the "tone/gain controls" only within DSP configurations, such DSP gain controls always affect more-or-less on "XO" "EQ" "phase" and "delay" of the DSP settings which will leads you to possible endless DSP tuning spirals every time; within DSP configurations, XO EQ Gain Phase and Delay are always not independent with each other, but they are always interdependent/on-interaction.

Just for your possible reference, my DSP-based multichannel multi-SP-driver multi-amplifier active system has flexible and safe analog level on-the-fly relative gain controls (in addition to upstream on-the-fly DSP gain controls) for L&R subwoofers, woofers, midrange-squawkers, tweeters, and super-tweeters, all independently and remotely. My post here shows you a typical example case for such safe and flexible on-the-fly analog-level tone controls. This my post (as well as
this post) would be also of your interest.

Of course, I know well that I (we) can also perform such relative gain control using DAC8PRO’s 8-channel output gain controllers. I do not like, however, to change the DAC8PRO’s output levels frequently on-the-fly (while listening to music) due to safety and inconvenience concerns; I like to keep DAC8PRO’s analog out gain level always at constant -4 dB which should remain to be usually “untouchable” in my case.

One of the very unique aspects/features of my multichannel audio rig is that I fully utilize four HiFi-grade “integrated amplifiers” plus L&R active subwoofers, each of them have its own gain (volume) controller for safe and flexible relative gain (tone) control in analog domain even on-the-fly i.e. while listening to music.

In this perspective, my posts #438 and #643 should also give you better understandings. Furthermore, my posts #317(remote thread), #313(remote thread) would be also of your reference and interest.

2. I would like to strongly suggest you evaluating your system at each of your progress steps not only by various objective measurements but also by your careful subjective listening tests using your own consistent music playlist consists of nice-recording-quality tracks selected from various genres of your preference, of course in your consistent listening room acoustics. In this perspective, my independent thread and posts thereof under the below spoiler cover would be your reference and interest:
my hosting thread:
An Attempt Sharing Reference Quality Music Playlist: at least a portion and/or whole track being analyzed by 3D color spectrum of Adobe Audition
and my summary posts thereof:
[Part-18] An Interlude or Provisional Finale of the Post Series: #669
and,
Updated, the latest, Audio Sampler Playlist as of October 20, 2022: #670

In case if you would be seriously interested in the intact tracks of my such "Music Playlist" as well as "the test tone-burst signals I prepared" and "SONY Super Audio Check CD", please simply contact me though PM communication.


I do hope much good luck in your present bi-amping (or further mutli-amping) audio project!:)
 
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We are actually working on an active version of the speaker right now. You can try this as a starting point. The last node on the tweeter is at 10,000 Hz. I should also mention that the amps on the tweeter and mid have 2.5 dB more input gain than the woofer amps so you will have to scale accordingly based on the amps you are using.

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Thank you for sharing this.

It is very interesting at least for me finding you are applying IIR filters (XO and EQ) all the way in your planned fully active configuration, since I too use IIR filters of DSP software "EKIO" all the way. As shared here, EKIO uses IIR filters; cascade 2nd order direct form II biquad in 64 bit floating point.

Just for possible reference and interest of OP @S Moore as well as people onboard this thread, let me share my latest EKIO XO and minimal EQ configuration specific to my 5-way 10-channel active setup (and SP drivers, cabinets, room acoustic environments, thereof) (ref. my post #931 on project thread) under the below spoiler cover. Please note I still use transparent-in-the-Fq-zone (I actually validated so, ref. here) physical protection capacitors for my treasure Beryllium-dome midranges, Beryllium-dome tweeters and metal horn super-tweeters.
Fig03_WS00007533 (6).JPG


Fig11_WS00007525 (4).JPG


Fig12_WS00007541 (4).JPG


Fig14_WS00007522 (11).JPG
 
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My first hypothesis is the wavelength of 2.7kHz is 12.7cm, very close to the 11cm diameter (3127Hz) of the LDW6 midrange driver. That means -2.64dB of 11cm signal is reaching the midrange driver, resulting in edge diffraction from the surround and conical directivity ("beaming").

The wavelength of 2kHz is 17.2cm, significantly larger than the 11cm LDW6 diameter. At 2kHz, -30.7dB of 11cm signal is reaching the midrange, a massive reduction. This avoids the frequency range where beaming and surround diffraction become problems.

We chose the higher crossover frequency in the passive crossover because it resulted in a better power response and DI than every variation of lower frequency crossovers we tried. For the active, we saw something similar but haven't had enough time to play around to see if this is something we can address active. Your theory would be the opposite of what would happen. Lowering the crossover frequency would put more energy in the area of the the woofer surround which would cause more diffraction, not less. What is happening is the lower crossover point is creating a peak in a region that people often perceive as "more detailed" as well as increasing the definition of the off axis horizontal mismatch.

2000 Hz.png
2000 Hz Crossover.png
2700 Hz.png
2700 Hz.png

I was wondering why Kerry chose 48dB. See my first response to Kerry was a surpise at that steep slope! I'll configure 24dB/2kHz and see what I think. I doubt if I will hear any difference. It'll still keep the 11cm away from the midrange.

Please explain.

Since the drivers are non-coincident (i.e. not a coaxial), higher order slopes minimize vertical interference and result in a better controlled vertical directivity.


We are still continuing to evaluate more crossovers. The one I posted was just a starting point after a morning of measuring and tweaking around the original crossover design. We haven't had time because of the holidays and a few life circumstances this month to get back in for testing yet but just started up again this week. We are looking at some lower order slopes as well as going to odd order, which in the initial modeling looks like it might result in the best power response so far. I'll try and throw up a few more options next week.
 
We chose the higher crossover frequency in the passive crossover because it resulted in a better power response and DI than every variation of lower frequency crossovers we tried. For the active, we saw something similar but haven't had enough time to play around to see if this is something we can address active. Your theory would be the opposite of what would happen. Lowering the crossover frequency would put more energy in the area of the the woofer surround which would cause more diffraction, not less. What is happening is the lower crossover point is creating a peak in a region that people often perceive as "more detailed" as well as increasing the definition of the off axis horizontal mismatch.

View attachment 423327View attachment 423329View attachment 423328View attachment 423330


Since the drivers are non-coincident (i.e. not a coaxial), higher order slopes minimize vertical interference and result in a better controlled vertical directivity.


We are still continuing to evaluate more crossovers. The one I posted was just a starting point after a morning of measuring and tweaking around the original crossover design. We haven't had time because of the holidays and a few life circumstances this month to get back in for testing yet but just started up again this week. We are looking at some lower order slopes as well as going to odd order, which in the initial modeling looks like it might result in the best power response so far. I'll try and throw up a few more options next week.
Awesome response and excellent product support! I saw the new 2.9kHz hump during the tweaking process and tossed in a -3dB PEQ to tame it. It was a sloppy job because I simply slapped the PEQ on the tweeter channel. I'm still trying to navigate the MiniDSP software. I still haven't learned how to implement a "system wide" PEQ (i.e. adjust the whole input signal before it is broken up) vs. "individual channel" PEQ. I didn't make enough measurements to determine if the hump was coming from the midrange, tweeter, or both. But since the midrange seemed to be working fine when the corner frequency was 2.7kHz, I figured the hump was coming from the tweeter.

Your original active crossover screenshot had a +3dB 2.5kHz PEQ filter on the midrange and +4dB 2.3kHz on the tweeter. I knew that my new 2.0kHz crossover knee frequency would directly screw up your original settings, so part of implementing my new knee frequency included troubleshooting new problems.

In regards to the midrange surround, it's my understanding that when the wavelength is equal to the diameter of the surround, it experiences separation at the geometric transition from conical to planer. It's not the movement (or "energy") of the surround, it's the boundary effect due to change of geometry. Shorter wavelengths stay in the conical boundary, which leads to "beaming". Longer wavelengths don't distinguish the transition between conical and planer because the length of the wavelength overwhelms the transition. This is how I understood things when I was building full-range driver systems.
 
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In regards to the midrange surround, it's my understanding that when the wavelength is equal to the diameter of the surround, it experiences separation at the geometric transition from conical to planer. It's not the movement (or "energy") of the surround, it's the boundary effect due to change of geometry. Shorter wavelengths stay in the conical boundary, which leads to "beaming". Longer wavelengths don't distinguish the transition between conical and planer because the length of the wavelength overwhelms the transition. This is how I understood things when I was building full-range driver systems.

Yes, having some obstruction or sharp change at the distance of the wavelength is what causes diffraction. That's not what I was saying though. With the crossover at 2.7 kHz - and I haven't checked your math on the wavelength but going by that assumption that the surround is that wavelength away - the tweeter is down 6 dB at that frequency. With the 2 kHz crossover, the tweeter response is elevated 6 dB over the 2700 Hz crossover at that point and there is more energy at all frequencies below it as well. This will cause any diffraction from 2.7 kHz and directly below as the wave travels over the surround and into the woofer cavity to be worse, not better.
 
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We've done some additional testing and here is where we are at currently. We've tried numerous different crossover points but right around 2700 Hz seems to be the sweet spot for minimizing off axis flare. This version is slightly smoother off-axis than the version I posted above but much flatter on-axis.
tempImageIV9Dvj.jpg
 
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