How best setup multiway PA system

Making it a 3 way system will clean it up alot, add a 12" or 2 x 10" direct radiator mid section, use a high and low pass filter on the subs, bear in mind the narrower bandwidth they have the harder you can drive them which is relevant outdoors / PA. QSC, LAB Gruppen or MC2 are all good choices of amplification, if you dont mind weight MC2 T series are good amps and not too expensive. A DSP Xover is well worth the investment and i wouldnt get hung up on the analogue signal path idea, a good DSP will bring more benefits with accurate X over, time alignment, EQ and limiters. XTA, Linea Research and NST all make very good sounding units and have older models that can be bought at sensible cost, also look at some of the OEM XTA rebadged models that can be lower cost, turbo, martin etc. Proper setup will require measurement. Setup the amp gain structure properly so you drive the DSP input hard and make use of the A/D converter dynamic range.
 
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Hi Art, you bring light into the rabbit hole 🙂

You could lower Fb by extending the ports, put output would be reduced. As a two way, 40Hz is a good compromise Fb.
Soldering the FDS-360 or extending the triangular ports of the ASB6125 is probably not an option since I'm a complete noob here. Does it rule out the FDS-360? If I'm back at the Ashley XR-77/12, then 48 Hz is the lowest I can set the Low/Mid frequency knobs. Does it cut out too much? And would you have any suggestions re Damping level? I maybe going through the pain points that led everyone using DSPs but I like to try it out the hard way (without breaking things of course) 😎

the BSS FCS 966 would be the best of those selections. It's separate HF contour filters could be used to (partially..) compensate for the HF rolloff of your HF/horn driver.
Seems like a good reason to get an additional EQ. While the Ashley already has basic EQ and the SSL Six has one per channel, I assume compensating for the HF rolloff requires another unit. Since everyone seems to prefer parametric eq over graphic eq, should I get PEG instead? I found an Aphex 109 Parametric Tube Equalizer for 400. It probably won't have the HF contour filters ...

DSP offers separate filter frequency choice and EQ, delay, compression, limiting for each output.
I totally see the point. But given our modest needs, I still hope the "filter frequency" topic can be covered by a good active analog x-over, delay is no issue since the speakers sit on each other, SSL Six has suberb compression, and limiting ... let's see. My reasons for avoiding digital are a mix: 1. I'm surrounded by digital day in and out, 2. screens and menu-diving is a complete turnoff for me but plugging and twisting things is fun, 3. I "believe" analog carries or adds some particles that digital can't, 4. I defended the Behringer NX-6000D (DSP) against naysayers but in direct comparison with an QSC PLX1104 I had to surrender.

Cheers
Mike
 
well i'm surprised no one was talked about directivity or coverage, at 1khz a 15 beams quite a bit and two in a vertical orientation means you need to be on axis to get that response...the twelve would alleviate that.
for a live performance rig "coverage" is what it's all about...unless the audience is willing to be stacked vertically or share space in the sweet spot...
i'll likely get dissed for this becuz it's not a "long" enough line but 4 10's or 6 8's in a vertical column would create a much wider coverage area for the mid range to get those "lush synths and acoustic jams" you mentioned out to the audience.
 
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Soldering the FDS-360 or extending the triangular ports of the ASB6125 is probably not an option since I'm a complete noob here. Does it rule out the FDS-360?
No, as I wrote, 800, 1K0, 1K2 would work OK. If it has one of those three cards loaded, OK.
If I'm back at the Ashley XR-77/12, then 48 Hz is the lowest I can set the Low/Mid frequency knobs. Does it cut out too much?
Much of what we hear as low frequency may be upper harmonics.
I put pictures of the note frequencies in my last post, you decide.
And would you have any suggestions re Damping level?
-6dB would be the default setting for two equally matched transducers that were time aligned.
Your transducers are neither equally matched transducers or time aligned if the front of your 60x40 horn is physically aligned with the front of the 2x15".
Seems like a good reason to get an additional EQ. While the Ashley already has basic EQ and the SSL Six has one per channel, I assume compensating for the HF rolloff requires another unit.
The ASHLY XR77/12 has no EQ.
Yes, compensating for the 60x40 constant directivity horn's rolloff requires CD compensation.

Mixer channel EQ is for adjusting individual inputs, not the speaker response.
3 to 4 bands of eq with two semi parametric (sweepable frequency) or at least 4 fixed frequency bands are useful per channel.
The speakers each could benefit from a minimum of 2 bands of fully parametric (variable frequency, and filter width "Q") EQ.

Since everyone seems to prefer parametric eq over graphic eq, should I get PEG instead? I found an Aphex 109 Parametric Tube Equalizer for 400. It probably won't have the HF contour filters ...
1/3 octave EQ are easier for "grab and go" and giving a visual idea of what has been done.
PEQ is more precise.
The Aphex 109 is more of an effect, like a guitar distortion pedal. Fun to insert on an individual channel, not for the entire PA, unless you like the sound of everything through a Marshall amp or the like.
I totally see the point. But given our modest needs, I still hope the "filter frequency" topic can be covered by a good active analog x-over,
Fixed filter frequencies can get close to a decent response, but seldom would be the same frequency for each driver to achieve a flat acoustic crossover.
delay is no issue since the speakers sit on each other,
The acoustic point of origin may be as much as 6" off, around a .5ms time alignment or 90degrees of phase difference, which could make the driver polarity choice equally "wrong" either way.
SSL Six has suberb compression, and limiting ... let's see.
I'm sure it works well for what it was designed to do, but that was not for speaker management.
When SSL consoles were "the thing", the studio monitors were time aligned and equalized..
My reasons for avoiding digital are a mix: 1. I'm surrounded by digital day in and out, 2. screens and menu-diving is a complete turnoff for me but plugging and twisting things is fun,
I also far prefer twisting knobs to poking screens.
The digital control surfaces I prefer have tactile switches and twistable knobs.

That said, once a speaker management system is properly set, it does not require adjustment, any more than you would go in and twist the timing and fuel injection controls on your car engine.
3. I "believe" analog carries or adds some particles that digital can't,
And digital makes manipulation of those "particles" far more convenient.
4. I defended the Behringer NX-6000D (DSP) against naysayers but in direct comparison with an QSC PLX1104 I had to surrender.
Mike, from what you have written thus far, I doubt your comparison was "direct".
Setting up a direct comparison would require level matching within .25dB and an A/B amplifier/speaker switch box.
If your test was not done like that, you are comparing room position, speaker to speaker (or crossover) and level variations.

That said, the amp's (or any) DSP can as easily be set to emulate "a metallic cookie box" sound as "flat response".
I've "fixed" a number of those mistakes on systems that sounded wrong due to inappropriate settings.

Art
 
Don't see the need for digital in our simple setup.
Suddenly you are an expert?
I try to avoid digital with screens unless it's absolutely needed 😉
The notion of "analog only" in a PA system these days naive at best. One of the most important tools DSP brings to the table is driver protection, brushing that off as unimportant when there is big amplifier power on tap will cost you in blown drivers.. one dropped mic or a moment of runaway feedback and there go all you HF diaphrams.
A modern PA processor is perfectly transparent and will have tons more clean headroom than analog components, plus some of the tools it brings cannot be recreated with analog gear. I guarantee you a dbx Venue 360, Ashly Protea, or something from XTA will not be the limiting factor in your system. Instead of having 1 component between your mixer and amplifiers your idea is to string together a chain of used analog gear with who knows how many potentially dodgy I/O connectors and noisy control pots? You're welcome to travel back to 1980 if you like but I was there and heard the bottoming woofers, witnessed the FOH engineer scrambling to figure out why the whole left side of the PA suddenly stopped passing sound. Running a PA system is much better these days with quality digital components in the mix.
 
I know it's a bit experimental but I'm making progress and very thankful for your feedback!

as I wrote, 800, 1K0, 1K2 would work OK. If it has one of those three cards loaded, OK.
Yes that's the case, the seller confirmed that there has been no changes since the BSS FDS-360 was bought.

3 to 4 bands of eq with two semi parametric (sweepable frequency) or at least 4 fixed frequency bands are useful per channel.
The speakers each could benefit from a minimum of 2 bands of fully parametric (variable frequency, and filter width "Q") EQ.
So the SSL Six only has HF and LF but I could insert more later. Re speakers: I could get up to three Klark Teknik DN410 Dual Channel Parametric EQ for 450 each or a bit less. I would add them after the BSS FDS-360 Crossover.

The acoustic point of origin may be as much as 6" off, around a .5ms time alignment or 90degrees of phase difference, which could make the driver polarity choice equally "wrong" either way.
If the 2" CD is perfectly aligned vertically with the woofer would that eliminate the issue? So in effect the ASB6125 subs are about 10 inches behind the front of the horn. The woofers would align with where horn meets CD.

once a speaker management system is properly set, it does not require adjustment
As @conanski mentioned, the dbx Driverack Venue 360 seems good at 96kHz sampling compared to 48kHz of dbx Driverack PA2. But some reviews mention the delay it introduces which can add up when things are constantly redigitized.

I doubt your comparison was "direct"
I had an OP-Z drumloop with some bells and an Roland MKS-50 synth connected to Soundcraft EFX and from there directly into Behringer NX-6000D (with x-over at 1kHz). I set the volume so the OP-Z would reach 90dB and with full keystroke reach 108dB for the MKS-50. Then switched cables and did the same with the QSC PLX1104 and back and forth a few times. With the QSC the kickdrum sounded more cohesive (punchy), the bells more rounded and less harsh and overall much more pleasurable to hear in a 15x15ft room. The room had old single-glass windows that vibrated a lot when the Behringer played but not at all with QSC. So overall, with the Behringer I was happy when my 2 minute test was over while with the QSC I could listen forever.
 
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will cost you in blown drivers.. one dropped mic or a moment of runaway feedback and there go all you HF diaphrams
Ok that's a good reason not be too experimental with it... At the moment it's more a lab environment and I'm progressing very careful.

string together a chain of used analog gear with who knows how many potentially dodgy I/O connectors and noisy control pots
Another good point! So I can do the listening comparison in the lab environment and if there's absolutely no disadvantage of high-quality new digital equipment vs. high-quality old analog equipment then I'm in. There's the delay issue of constantly digitizing and redigitizing and the whole process seems a bit absurd to me. I could go completely digital but then I spend also my free time in front of the laptop.
 
There's the delay issue of constantly digitizing and redigitizing and the whole process seems a bit absurd to me
Yes I get that and wouldn't suggest introducing a bunch of separate digital components, the beauty of the all-in-one PA processor is that there is only one AD and DA conersion, all of the included processing(Crossover, EQ, time alignment, Limiting, level matching) takes place in the same digital domain, and with all system output going through this unit the rather insignificant latency added (1-2ms) is imperceptable. Another benefit of this kind of precessor is the ability to time align multiple speakers down the length of a room/listening area for a distributed system, or if you get to work with live bands you can delay the FOH speakers to the backline which helps glue together all of those sound sources.
 
Yes that's the case, the seller confirmed that there has been no changes since the BSS FDS-360 was bought.
What frequencies are the installed crossover cards?
If the 2" CD is perfectly aligned vertically with the woofer would that eliminate the issue? So in effect the ASB6125 subs are about 10 inches behind the front of the horn. The woofers would align with where horn meets CD.
As a rough estimate, the 4" CD diaphragm should be physically aligned above a point around the 15" dustcap for time alignment if the acoustic crossovers are symmetrical 12 (invert polarity of the HF driver) or 24dB per octave.
The acoustic crossovers won't be symmetrical without individual EQ on each output.

I had an OP-Z drumloop with some bells and an Roland MKS-50 synth connected to Soundcraft EFX and from there directly into Behringer NX-6000D (with x-over at 1kHz). I set the volume so the OP-Z would reach 90dB and with full keystroke reach 108dB for the MKS-50. Then switched cables and did the same with the QSC PLX1104 and back and forth a few times.
There are over a dozen "1kHz" crossover options available with the NX-6000D.
Each has a different phase (and potential polarity) response.
Depending on the output configuration of your connectors, there is a potential to have powered a single woofer, both, or reversed polarity on one or the other.

The QSC PLX1104 has no crossover, so could not be configured as the NX-6000D.
50% chance of relative driver polarity being correct in either set up, which could amount to a huge difference in frequency and vertical polar response between the two.
The room had old single-glass windows that vibrated a lot when the Behringer played but not at all with QSC.
Proof enough in itself the comparison was not matched in level, frequency or phase.
So overall, with the Behringer I was happy when my 2 minute test was over while with the QSC I could listen forever.
You determined you heard configuration differences, not amplifier differences in the 2 minute test.

Best of luck with the rest of your system choices!
 
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Thank you for guiding a noobie through the analog jungle 😉

I hooked the Ashley XR77/12 up and it seems to do its job but there's buzzing/humming when connected to the SSL Six. To check I plugged the OP-Z directly into the amp (connected to JBL 2450H) and turned all volumes up without anything playing. When adding the XR between it doesn't add any more noise. BUT when connecting the SSL Six to the Ashley, it adds buzzing/humming that has to do with electricity (e.g. when putting the Six's external power supply on the main output cable, the buzzing/humming is reduced). All are on the same electricity circuit.

The LR24dB/oct BSS FDS-360 sounds far better than the ASHLY XR77/12, is dead on accurate and has (peak) limiters
The problem is the outputs of the FDS-360 are unbalanced. Given the buzzing/humming I already see, that is probably not helpful.

What frequencies are the installed crossover cards?
If he says nothing had changed I assume they're the ones you posted the screenshot of. Should I explicitly ask for it?

if the acoustic crossovers are symmetrical 12 (invert polarity of the HF driver)
Since the Ashley is 12dB, does it mean I need to invert its polarity? Per JBL’s doc, positive voltage to the black terminal causes diaphragm motion toward the phasing plug. So inverting polarity, I just reverse these connections? How do I know what is + and what - coming from the amp via speakon cable?

The acoustic crossovers won't be symmetrical without individual EQ on each output.
So I need one equalizer per per output, low and high in this case. I hooked up a BSS FCS 966 graphic EQ before the Ashley XR, so I need two parametric EQs for after the XR. Or could/should I use two graphic EQs?

50% chance of relative driver polarity being correct in either set up, which could amount to a huge difference in frequency and vertical polar response between the two.
So how I got the NX-6000D refunded and got a QSC PLX2402 in good condition and with the QSC PLX1104 I spend about the same. Both weren't used in touring and look almost new.

Best of luck with the rest of your system choices!
Thank you, it will all be good due to all these excellent suggestions here. When I'm happy with my analog setup, I will order a DBX DriveRack and see if it improves and if yes, I keep it.

BTW: DriveRack Pro vs. DriveRack Venue? I'm mostly interested in sound quality and less on speaker management for a stadion 😉
 
BUT when connecting the SSL Six to the Ashley, it adds buzzing/humming that has to do with electricity (e.g. when putting the Six's external power supply on the main output cable, the buzzing/humming is reduced).
Sounds like you have "ground loop" noise. If using TRS cable, "+" is carried on the Tip, "-" on the Ring, and the Sleeve should be connected to the cable shield.
The shield and ring may be internally connected on one or more of your units.
Shield connections may be connected to chassis (earth) grounds on one or more of your units.
The shield connection may have to be "lifted", (cut) to eliminate the "ground loop".
Convention was to lift the shield at the receive end.
The problem is the outputs of the FDS-360 are unbalanced. Given the buzzing/humming I already see, that is probably not helpful.
As I recall, the FDS-360 was electronically balanced using the current standard of XLR pin 1 shield, pin 2 "+", pin 3 "-". It's pin 1 shield connection may have to be "lifted", (cut) to eliminate "ground loops".
Back in the "bad old days", "shield lift" XLR cables were common.

Some XLR connectors are wired with pin 1 to the case, that can prevent a "shield lift" cable from doing it's job of breaking a ground loop.

There is a lot of old analog gear with grounding schemes so poorly designed that galvanic isolation (transformers) are required to eliminate ground loop noise.

If he says nothing had changed I assume they're the ones you posted the screenshot of. Should I explicitly ask for it?
I posted ~25 different card options may be installed (there are almost as many more above 1.5kHz) and only three of those that should be usable with your loudspeakers.
Yes, you should ask what the cards are.
Since the Ashley is 12dB, does it mean I need to invert its polarity?
Read it's manual, see if you interpret it the way I did in post #13.
Per JBL’s doc, positive voltage to the black terminal causes diaphragm motion toward the phasing plug.
That is backwards (inverted) from the rest of the industry.

Technical Notes Volume 1, Number 12C
Polarity Conventions of JBL Professional Transducers and Systems

Beginning in the early 1990s, JBL began a transition to the positive-to-red standard in accordance with practices recommended by the AES, EIA, IEC, and other standards organizations. The process has been underway since that time, with changes being applied to new transducer models and loudspeaker systems. The decision was made that no “legacy” transducers in existence at that time would be changed during their remaining catalog lifetime. In most cases, JBL loudspeaker systems, even if they make use of negative polarity transducers, have been internally wired so that they “behave” as positive systems. That is, a positive signal at the red or positive terminal of the system will cause the low frequency cone to move outward.

Cone loudspeaker speakers can be checked with a 9volt battery to see the direction of movement.
So inverting polarity, I just reverse these connections?
Yes, reversing speaker connections inverts the polarity from what it was.
How do I know what is + and what - coming from the amp via speakon cable?
The Speakon connector's wiring can be tested with an ohm meter to determine if it is wired correctly, and continuity between the 2, 4, or 8 connections are not crossed or shorted. It's easy to swap polarity when assembling those cords..

Amplifiers output is alternating current, tools like an oscilloscope or REW (freeware)
https://www.roomeqwizard.com/help/help_en-GB/html/graph_splphase.html
can give you an idea of what is going on in the land of phase.

It is interesting to see how many audio components don't adhere to the current standard of XLR pin 1 shield, pin 2 "+", pin 3 "-", or invert from input to output.
So I need one equalizer per per output, low and high in this case. I hooked up a BSS FCS 966 graphic EQ before the Ashley XR, so I need two parametric EQs for after the XR. Or could/should I use two graphic EQs?
You can use whatever EQ you want wherever you like.
Graphic EQ has less "surgical" control than PEQ, but easier to "see" what the EQ's response may resemble.
When I'm happy with my analog setup, I will order a DBX DriveRack and see if it improves and if yes, I keep it.
I was never happy with Ashly analog crossovers.
BSS analog crossovers were a great improvement by comparison.

The control and flexibility a DBX DriveRack PA afforded allowed a great improvement in the system's sound quality in comparison to any analog speaker management I had used prior.
All the current DBX DriveRack are considered to be of a higher sound quality than the 20 year old unit I still use.

Optimally setting a system's protection limiters, HP filters, crossover frequencies, slopes, EQ, time and phase alignment requires knowledge, experience and measurement.

Here is an example of a 2-way JBL SRX725, and what a DJ managed to do to the response with improper crossover settings, the stock passive crossover response on top in black and white, his bi-amped response below in blue:
SRX Gonzo.png

DJ Gonzo had been "adjusting" the active crossover on gigs for several months, managed to turn a +/-3dB response into a disaster.

Unfortunately, a response like that is often what inexperienced operators accomplish when they attempt to integrate separate components into a speaker system.

Took about 15 minutes for me to discover what he had done wrong and bring the system into alignment.

Art
 

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Hello again, I did some testing with my setup with the Ashley XR-77/12 XOR. It works but adds too much noise/hums so I will return it. Conclusion is to use a DSP 😉 I also got a BSS FCS 966 GEQ for almost free and not noisy but it will be redundant when going digital. I also got some full-range speakers (FRS) dirt cheap that I may use as monitor. I wasn't prepared to spend several grants on amps so just got another used one very cheap. So here's my current setup ...

Speakers
  • JBL ASB6125 (35 Hz - 1 kHz, 135 dB SPL max, 1600W 2h, 1350W 100hrs)
  • JBL 2450H with JBL 2385A horn (500 Hz to 20 kHz, 111 dB SPL, 150W cont.)
  • JBL AC18/95 (60 Hz - 20 kHz, 117 dB max, 250W cont.) >>> for monitoring or as 3-way system?
Amps
  • QSX PLX 1104 (20-20kHz, -108dB, 2x500W 4Ω) >>> used for the horns but perhaps better for the FRS
  • QSX PLX 2402 (20-20kHz, -108dB, 2x700W 4Ω, 2x1200W 2Ω) >>> used for the ASB6125 @4Ω now,
The ASB6125 specs say "4 ohms in parallel-drive, 2x8 ohms in discrete-drive mode" but can/should I run them at 2Ω ?

The t-racks are great value
Yes but they can only be controlled remotely via Windows whereas DBX Driverack PA2 allows Mac and iOS. Plus it has Auto-EQ. So with an Behringer ECM8000 for measuring and free Audionet CARMA software, perhaps I get a dummy-proof system 😉 But PA2 only samples with 48 kHz vs. 96 kHz of the t.racks DSP 206. Does it outweigh the benefits of the PA2?

Yes, you should ask what the cards are
1600 kHz so it's out of the game. The Ashley as well because of noise/hum. So now the question is: which DSP? I can get a high-rated t.racks DSP 206 with 96 kHz sampling for 300 or same price used dbx driverack 260 with 48 kHz sampling but both don't support Mac & iOS so they're useless for easily going remote. Behringer DCX2496 has 96 kHz, iOS support and "some" auto-EQ. I tend towards PA2 because it also has feedback suppression. What do you think?

Yes, reversing speaker connections inverts the polarity from what it was.
So I could just swap the cable ends and listen if there's a difference and choose the best sounding connection?

Optimally setting a system's protection limiters, HP filters, crossover frequencies, slopes, EQ, time and phase alignment requires knowledge, experience and measurement.
Do you think a DSP-based LMS could get me there together with Behringer ECM-8000 microphone and Audionet CARMA software? How about Driverack PA2 auto-eq does it do its job well? I will be using the PA system indoors and outdoors.

Unfortunately, a response like that is often what inexperienced operators accomplish
So a DSP-based LMS (perhaps with software like CARMA) could protect me from evil? And if so, what should I look for? 🙏

BTW: I want to build a wooden box around the 2450H 2"CD and horn. Since the CD is quite heavy, does it need some support, e.g. a wooden beam to hold its weight? Else the whole weight is connected to the box only via a few screws at the front of the horn. Basically the CD is just hanging in the air.
 
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JBL AC18/95 (60 Hz - 20 kHz, 117 dB max, 250W cont.) >>> for monitoring or as 3-way system?
It's a little 8"/1.5" diaphragm 2-way box, you could use it for a monitor.
117dB is about -20dB (one quarter as loud) as your 2x15" rated output.
The ASB6125 specs say "4 ohms in parallel-drive, 2x8 ohms in discrete-drive mode" but can/should I run them at 2Ω ?
Two 8 ohm speakers in parallel presents a four ohm load to the amplifier.
"Discrete-drive mode" means running each 8 ohm speaker from it's own (discrete) amp side.
The PLX 2402 can put out 425 watts per channel at 8 ohms.
Running all four of your 15" in parallel would be a 2 ohm load.
In a pinch, you could run all four from one side of an amp rated for 2 ohm operation.
Using the PLX 2402, the four speakers would have 1200/4=300w per driver, rather than 700/2=350w per driver as you are running now.
350 watts is not much for a speaker rated for 5400 watts peak power handling capacity, the 2x15" speakers could safely handle more than another 10 dB of peak power (3500 watts).
So I could just swap the cable ends and listen if there's a difference and choose the best sounding connection?
You can make any choice you want as to what "sounds best".
Now that you have the AC18/95, you can compare your choices to it's response.
But PA2 only samples with 48 kHz vs. 96 kHz of the t.racks DSP 206. Does it outweigh the benefits of the PA2?
Feedback suppression and Auto-EQ are somewhat like random application of Auto-Tune, the results are seldom "best sounding".
I have not compared the two DSP platforms you are considering, but I can't hear a difference between sampling rates above the usual CD sampling rate of 44.1kHz even in a tightly controlled A/B situation.
Go with the interface you prefer.

So with an Behringer ECM8000 for measuring and free Audionet CARMA software, perhaps I get a dummy-proof system 😉 So a DSP-based LMS (perhaps with software like CARMA) could protect me from evil?
No tools make a system dummy-proof, or protect the innocent from evil 😉
BTW: I want to build a wooden box around the 2450H 2"CD and horn. Since the CD is quite heavy, does it need some support, e.g. a wooden beam to hold its weight?
Yes, without support it's weight may crack the plastic horn on a sharp bump, especially in cold weather.
You never know what way a dropped cabinet may land, so a circular support would be best, like this example:

d491ec2b875c4b9a2a468307a9f7c9ed.jpg.80b2f3904b8812b9c793186aeb088993.jpg

That said, at 4.8 kg (10½ lb), the 2405J is only about half the weight of the above EV driver or the JBL 2445, so less likely to crack the 2385A horn.

Art
 
The ASB6125 specs say "4 ohms in parallel-drive, 2x8 ohms in discrete-drive mode" but can/should I run them at 2Ω ?
2ohms operation is not recommended for PA applications.. particularly for low frequencies, and you can't force an amp to operate at a particular impedance anyway... the speakers connected determine that.
Yes but they can only be controlled remotely via Windows whereas DBX Driverack PA2 allows Mac and iOS. Plus it has Auto-EQ. So with an Behringer ECM8000 for measuring and free Audionet CARMA software, perhaps I get a dummy-proof system 😉 But PA2 only samples with 48 kHz vs. 96 kHz of the t.racks DSP 206. Does it outweigh the benefits of the PA2?
Your focus on remote control is a bit misplaced, really once the processing is setup to optimize output from the speakers there is little need to access it on site... altough in this day and age I kinda get it when just about anything can be controlled wth an app.
I can get a high-rated t.racks DSP 206 with 96 kHz sampling for 300 or same price used dbx driverack 260 with 48 kHz sampling but both don't support Mac & iOS so they're useless for easily going remote. Behringer DCX2496 has 96 kHz, iOS support and "some" auto-EQ. I tend towards PA2 because it also has feedback suppression. What do you think?
Auto EQ can be a somewhat useful tool in certain applications... like where a sound system is installed and never moved, but for a mobile system not so much. The software is dumb and can't tell the difference between direct and reflected sound and can't compensate for room nodes, so the operator has to know how to interpret the results.. know what to keep and what to throw out, it's not a run and done proceedure.
The DCX has auto time alignment not EQ, this feature does work fairly well though. Feedback supression can be useful but you're more likely to need it on stage monitors than the main PA. There is a version of the t-Rack with FIR processing, that is the latest/greatest these days.

Do you think a DSP-based LMS could get me there together with Behringer ECM-8000 microphone and Audionet CARMA software?
Have you looked into REW, it's free to use... https://www.roomeqwizard.com/
BTW: I want to build a wooden box around the 2450H 2"CD and horn. Since the CD is quite heavy, does it need some support, e.g. a wooden beam to hold its weight? Else the whole weight is connected to the box only via a few screws at the front of the horn. Basically the CD is just hanging in the air.
Yes some support for the compression driver is a very good idea.
 
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