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Sound Reinforcement - Forums for Live Sound Professionals - Your Displayed Name Must Be Your Real Full Name To Post In The Live Sound Forums => LAB Lounge => Topic started by: Miguel Dahl on April 10, 2021, 08:50:07 AM
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First, I can't believe how rusty I've become after almost a year doing close to nothing due to corona..
My new workplace have Meyer UPQ-1P mains and some older Yamaha DXR12 as delays. The delays only cover three rows.
My utility-case with soundcard and measurment microphone is unaccessible to me at the moment, so I did some adjustments to the system by ear, as it needed some tweaking.
I can't get the delays to play well with the mains. There's some phasing going on. I've tried increments of 1ms, and flipping polarity on one of the speakers for each increment.
There's always some narrow range of frequencies which are out of phase between delays and mains. So I chose the setting which had the least amount of "crap" in the response, and notched the range out of the delays which sounded most horrible. "Do I want it to be in phase at 250, 500, 750, and out of phase between that, or the other way around" as a from the top of my head example.
Is this just as simple as the phase response between the two speakers is just not the same in that region I'm having trouble with? It sounds good up to about 200Hz, I can hear they play well together, then the inverse phase-thing start to happen up to some higher mid somewhere. What's happening in between there in some ranges is just disturbing to listen to.
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As a general rule of thumb, the delay should be about 1mSec per foot for each foot of distance between the mains and the delay speakers (with the delay of course on the delay speakers).
I am unsure of the phase as I have no idea if Meyer and Yamaha speakers utilize the same default phase for their speakers or not. Generally the low frequency response can tell you by ear. If you are out of phase, the woofer frequencies should be noticeably less pronounced than when in phase; however, since I assume you are using 2 speakers for FOH there are usually patterns of cancelation depending on where you stand anyway.
Use only one side at a time to avoid the comb filtering normally seen in a venue while doing your setup.
Your ears are the best instrument .... but I do agree, it is always nice to see measurements that agree with your ears :)
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As a general rule of thumb, the delay should be about 1mSec per foot for each foot of distance between the mains and the delay speakers (with the delay of course on the delay speakers).
I am unsure of the phase as I have no idea if Meyer and Yamaha speakers utilize the same default phase for their speakers or not. Generally the low frequency response can tell you by ear. If you are out of phase, the woofer frequencies should be noticeably less pronounced than when in phase; however, since I assume you are using 2 speakers for FOH there are usually patterns of cancelation depending on where you stand anyway.
Use only one side at a time to avoid the comb filtering normally seen in a venue while doing your setup.
Your ears are the best instrument .... but I do agree, it is always nice to see measurements that agree with your ears :)
+1 what Scott said
JR
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As a general rule of thumb, the delay should be about 1mSec per foot for each foot of distance between the mains and the delay speakers (with the delay of course on the delay speakers).
I am unsure of the phase as I have no idea if Meyer and Yamaha speakers utialize the same default phase for their speakers or not. Generally the low frequency response can tell you by ear. If you are out of phase, the woofer frequencies should be noticeably less pronounced than when in phase; however, since I assume you are using 2 speakers for FOH there are usually patterns of cancelation depending on where you stand anyway.
Use only one side at a time to avoid the comb filtering normally seen in a venue while doing your setup.
Your ears are the best instrument .... but I do agree, it is always nice to see measurements that agree with your ears :)
Same as 3ms pr meter, -ish. Then maybe add a little tiny bit for haas effect for localiztion of the sound to come from the stage, depending on temp.
I was only listening to one side when I did the adjustments. And even if I moved from "here" to there", there was some phase-issues no matter what. When I flipped the polarity the phase issues just changed which range they occured in. Moving up or down by a few hundre Hz. Maybe 200Hz wide ranges.
I measured the relative distance between mains and delays with a laser finder, just to get a ballpark ms-number, and took it from there.
I've dialed in delays a thousand times, but more often than not, it's been the same brand and family of speakers, Meyer, D&B. But this time I was hearing something which was really off, in a dead quitet environment though, as opposed to outdoors.
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It's hard to do it by ear if you have two different brands, IME having speakers with matching response as close as possible is way easier to align, and most manufacturers have boxes that match ish straight out of the box.
What I do if I don't have a analyzer is set the delay roughly based on distance, then fine-tune it by talking into a mic. Usually works decent enough for speech intelligibility. I like the result better than using music or noise as a signal.
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It's hard to do it by ear if you have two different brands, IME having speakers with matching response as close as possible is way easier to align, and most manufacturers have boxes that match ish straight out of the box.
What I do if I don't have a analyzer is set the delay roughly based on distance, then fine-tune it by talking into a mic. Usually works decent enough for speech intelligibility. I like the result better than using music or noise as a signal.
But to cut to the thing I was originally asking about. Can this thing I'm hearing be because of two different speaker brands, not same family = different phase response? I'm pretty sure I'm hearing at least 120-ish degrees off in those ranges I was speaking about.
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I would sanity check the individual enclosures. If anything is horked up there, you will never be able to dovetail them together. Without seeing a floor plot, it's difficult to predict any behavior characteristics.
For the sanity check, look around everything in the signal chain to see if someone "fixed" something. I suspect that is very likely and you are hearing the results. A janitor could have knocked something loose and reconnected it. Anything is possible.
Once certain the system is doing what it's supposed to do you can move forward with tuning it.
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Since the lows carry better, could high passing the delays allow you to align a smaller frequency range?
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But to cut to the thing I was originally asking about. Can this thing I'm hearing be because of two different speaker brands, not same family = different phase response? I'm pretty sure I'm hearing at least 120-ish degrees off in those ranges I was speaking about.
IME it's usually more of a response-issue than a phase-issue. A delay will stick out if it's way off in frequency response or timing, it's only phase matched for a small area with the mains anyway.
IMHO, YMMV, It Depends etc...
I've turned off delay systems in reverberant rooms sometimes, not because they were bad installs/alignments, but because the reflection from the back wall was more annoying than level/response issues for the last row of seats. Got a nasty sounding slap back from the delays off the back wall. Could this be a similar issue?
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But to cut to the thing I was originally asking about. Can this thing I'm hearing be because of two different speaker brands, not same family = different phase response? I'm pretty sure I'm hearing at least 120-ish degrees off in those ranges I was speaking about.
It's hard to say exactly what's going on without access to a measurement rig, but for what it's worth the published phase responses of those two boxes are nearly identical (although I'd characterize the DXR12 as being more PC125 than PC100).
Is there anything else in the system processing that would cause phase offset? What processor are you using and how is it currently configured?
-Russ
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This is a cases of where actual measurement is required, NOT just listening by ear.
I had a case once in which each of the mains and delays were well behaved, but they had different phase responses.
The amplitude and phase response of each loudspeaker were good, but they were different.
I could either get them to align well above 400Hz or below 400Hz. But not over the entire range.
But simply "throwing a highpass filter in" affected the phase response again.
Some quick measurements (assuming proper amplitude and phase measurements) will go a long way towards solving the problem.
Different phase responses will result in dips and peaks in the combined response. No amount of delay can fix that. If the phase responses are the same, then delay can help fix the issues. Notice I said "help fix", not fix. There are a number of issues that also arise with delay speakers (such as what happens BEHIND the loudspeaker), that have to be addressed in different ways, but cannot be eliminated.
The other suggestions presented are good, making sure each speaker is wired up properly, only doing 1 set of speakers at a time etc are all valid if you can't do measurements
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I don’t need to repeat what other have already said but I will add that in my opinion if a delayed speaker system is done right people will think that the delayed speaker is off. Because of this I have had clients that don’t want delayed speakers installed because they are tired of having to deal with people telling them that the speakers are off. Even though the room and situation calls for delayed speakers to work the best for what they are trying to do. I have said to people that I think it would be great if you can hide the delayed speakers so the people won’t be listening with their eyes.
Are you possibly trying to run these delayed speakers too loud?
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The Meyer probably has a couple of ms of latency and the DXR probably also has a couple of ms of latency. Between the two you may have as much as 5ms or more depending on the latency of the mixer and the Foh system to the subs? So doing a straight conversion of footage between the main hang and the delay speakers can often be off by several feet.
As mentioned before, having a PA with matching units is much easier because they will sound the same. If one speaker has a much different sonic character, trying to hear where things lock-in is hard to do. Also, you are working with two speakers that have totally different crossover topologies. One operates with fairly typical filters ( the Meyer ) and the other utilizes FIR filters ( the DXR ), which can create some weird phase shifts that are frequency-dependent when compared to another conventional product. I.E. you can get it right in one frequency range, but not another. The use of all-pass filters creates a similar effect. In either case, unless you know the processing and have access to processing that can emulate it, it is a fruitless endeavor.
Typically, assuming you have a well-balanced system with products that have similar attributes, you can hear the two parts LOCK-IN when you adjust the delay. Often you can guesstimate the footage and in place delay, and then as you slowly add or subtract delay you can come to a point where you hear it lock in. All of a sudden 1ms is either right or wrong. The delay will just seem to go away and you just hear the FOH sound become more apparent. Your eyes and ears will simply be drawn to the stage as opposed to realizing the delayed speaker. But if the two systems are far enough separated by processing tricks ( FIR, delay, latency, or frequency response ) the task of time aligning becomes very difficult to do by ear; quite literally requiring a DSP and FFT solution to correct.
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I don’t need to repeat what other have already said but I will add that in my opinion if a delayed speaker system is done right people will think that the delayed speaker is off. Because of this I have had clients that don’t want delayed speakers installed because they are tired of having to deal with people telling them that the speakers are off. Even though the room and situation calls for delayed speakers to work the best for what they are trying to do. I have said to people that I think it would be great if you can hide the delayed speakers so the people won’t be listening with their eyes.
Are you possibly trying to run these delayed speakers too loud?
Exactly. You should not realize they are on, until you turn them off, and you notice the difference.
I have had jobs in which I HAD to shorten the delay, so the customer could "hear" the speakers, and THEN we would get paid. They would not pay us until they "heard" them working, even though I had demonstrated that turning them off made a big difference.
Oh well. The worst thing is that in years afterwards if a true audio guy comes in and finds out that I tuned the system, and then says I didn't know what I was doing because the delay wasn't set right----------
HEY, at least I got paid, and that was the whole purpose of setting it up wrong.
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Exactly. You should not realize they are on, until you turn them off, and you notice the difference.
I have had jobs in which I HAD to shorten the delay, so the customer could "hear" the speakers, and THEN we would get paid. They would not pay us until they "heard" them working, even though I had demonstrated that turning them off made a big difference.
Oh well. The worst thing is that in years afterwards if a true audio guy comes in and finds out that I tuned the system, and then says I didn't know what I was doing because the delay wasn't set right----------
HEY, at least I got paid, and that was the whole purpose of setting it up wrong.
The customer (who pays the bills) is always right, even when wrong...
JR
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The DSP is a Galileo 408, running 3.12 FW,
That's my apporach to tuning delays. They should not be heard, but one can hear the difference when turned off.
I found a soundcard and threw up a condenser, not a measurment reference mic, but a high quality (can't remember which one) condenser, at least, just to look at the relative differences, not important if the mic itself is "off" comparing to what's really happening... Only measured up to 2K looked at the problem area which is lower than that. The SPL-response in REW told me what I wrote earlier what I was hearing, about 200Hz-ish wide bands which either have a dip or peak when I flip the polarity. Can't get the mains and the DXR's to align over that spectrum, it's either this response, or that..
I don't know if using that mic was worth anything to be honest, but even if the mic is "the wrong one", the relative response would be the same. Maybe it was just a coincidence that what I saw in REW matched with what I heard earlier.
Delays are high passed at 120, subs XO mains at 70. The delay settings were in the ballpark after confirming later with the "delay-finder". When doing "fine-tuning" times by ear I usually use a metronome. When I hear one click and sound is the same when flipping polarity I call it good enough.
I'll be able to look at it properly the next week.
We also have some UP-Juniors of some kind, maybe that would be better to hang instead of the DXR's as delays to get more of the same sonic signature.
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For voice centric shows in large spaces ( convention centers and arenas ) I generally look to delay speakers for improvements in articulation only and high pass them accordingly sometimes as high as 400-500. There are certainly applications where delays are more akin to separate zones and wider bandwidth makes sense but if your listeners are in direct line with the main speakers try a high cut off and see if that cleans things up.
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Delays are high passed at 120, subs XO mains at 70. The delay settings were in the ballpark after confirming later with the "delay-finder". When doing "fine-tuning" times by ear I usually use a metronome. When I hear one click and sound is the same when flipping polarity I call it good enough.
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For voice centric shows in large spaces ( convention centers and arenas ) I generally look to delay speakers for improvements in articulation only and high pass them accordingly sometimes as high as 400-500. There are certainly applications where delays are more akin to separate zones and wider bandwidth makes sense but if your listeners are in direct line with the main speakers try a high cut off and see if that cleans things up.
That's a good point. Thanks.
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For voice centric shows in large spaces ( convention centers and arenas ) I generally look to delay speakers for improvements in articulation only and high pass them accordingly sometimes as high as 400-500. There are certainly applications where delays are more akin to separate zones and wider bandwidth makes sense but if your listeners are in direct line with the main speakers try a high cut off and see if that cleans things up.
Exactly.
The delays should bereproducing "what is missing" from the main coverage.
Sometimes I have highpassed around 800Hz or so. But yeah, 300-500Hz is VERY common for delay highpass
The one thing I do to keep the delays from "feeling thin", is to use a more gradual highpass than normal. Usually a 1st or 2nd order filter.
I would start by measuring in the main coverage area, then move to the delay coverage area and see what is missing. You will see the mid - highs start to roll off.
Start by putting a HP where they start to trail off, and go from there. The quality of the mic doesn't matter, as long as the same mic is used for both locations, and all you are trying to do is match the other response. NOT to provide a flat response.
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Exactly.
The delays should bereproducing "what is missing" from the main coverage.
Sometimes I have highpassed around 800Hz or so. But yeah, 300-500Hz is VERY common for delay highpass
The one thing I do to keep the delays from "feeling thin", is to use a more gradual highpass than normal. Usually a 1st or 2nd order filter.
I would start by measuring in the main coverage area, then move to the delay coverage area and see what is missing. You will see the mid - highs start to roll off.
Start by putting a HP where they start to trail off, and go from there. The quality of the mic doesn't matter, as long as the same mic is used for both locations, and all you are trying to do is match the other response. NOT to provide a flat response.
It's a not very long throw from the mains, but there can be some lamps in the way which casts a shadow (heh), and the mains are some degrees downward and inward, to cover the closer seats. But that would mainly only affect the highs.. So high passing higher, and maybe just do it so high that the troublesome area is out of the equation is a great tip. I'll try that. But still I feel like I'd want to try the UP-Juniors for the tonality.
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First, I can't believe how rusty I've become after almost a year doing close to nothing due to corona..
My new workplace have Meyer UPQ-1P mains and some older Yamaha DXR12 as delays. The delays only cover three rows.
My utility-case with soundcard and measurment microphone is unaccessible to me at the moment, so I did some adjustments to the system by ear, as it needed some tweaking.
I can't get the delays to play well with the mains. There's some phasing going on. I've tried increments of 1ms, and flipping polarity on one of the speakers for each increment.
There's always some narrow range of frequencies which are out of phase between delays and mains. So I chose the setting which had the least amount of "crap" in the response, and notched the range out of the delays which sounded most horrible. "Do I want it to be in phase at 250, 500, 750, and out of phase between that, or the other way around" as a from the top of my head example.
Is this just as simple as the phase response between the two speakers is just not the same in that region I'm having trouble with? It sounds good up to about 200Hz, I can hear they play well together, then the inverse phase-thing start to happen up to some higher mid somewhere. What's happening in between there in some ranges is just disturbing to listen to.
It's been ages since I've swum in these waters...but, I have been following this thread with interest. The first thing that occurred to me was that maybe there was some DSP action going on in the Yamaha boxes. Not being familiar with the DXR12 (which, I know, makes me a freak of nature on this forum :-[) I looked them up and found this: "D-CONTOUR is an intelligent multi-band compressor that gives you powerful and consistent sound throughout all output levels. By constantly monitoring the output of multiple frequency bands and calculating the optimum EQ adjustments for each, even the maximum sound output maintains outstanding clarity and musicality. With the DXR Series, D-CONTOUR provides a more detailed tuning of your sound with two different settings: FOH/Main mode or Monitor mode."
I would say that if that function is turned on, you have little to no chance of getting them to play nice with the UPM's.
And, as others have suggested, it could be radically different voicings between the two brands of speakers, but if you can access the controls on the DXR12's, I'd see if you can bypass all that "cleverness" and just get them to be a 2-Way speaker.
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It's been ages since I've swum in these waters...but, I have been following this thread with interest. The first thing that occurred to me was that maybe there was some DSP action going on in the Yamaha boxes. Not being familiar with the DXR12 (which, I know, makes me a freak of nature on this forum :-[) I looked them up and found this: "D-CONTOUR is an intelligent multi-band compressor that gives you powerful and consistent sound throughout all output levels. By constantly monitoring the output of multiple frequency bands and calculating the optimum EQ adjustments for each, even the maximum sound output maintains outstanding clarity and musicality. With the DXR Series, D-CONTOUR provides a more detailed tuning of your sound with two different settings: FOH/Main mode or Monitor mode."
I would say that if that function is turned on, you have little to no chance of getting them to play nice with the UPM's.
And, as others have suggested, it could be radically different voicings between the two brands of speakers, but if you can access the controls on the DXR12's, I'd see if you can bypass all that "cleverness" and just get them to be a 2-Way speaker.
In did not know that D-contour was as "clever" as stated. I uasually turn off all "helping processing". But Maybe I need to check this one again. But what I know is that there's an internal HPF engaged on the boxes. Hmm. I'll take another listen to it tomorrow when I for sure know evertything is turned off. HPF is the least a Galileo can do for me.
So there's an actual multiband compressor inside the box. I did not run hard at all for testing, but maybe the circuit even if it's doing nothing actively can do some stuff to the output, if engaged. Huh, there I learnt something new, never actually checked what the D-contour does. Slap on my fingers for not checking the equipment I'm using.
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A few comments in general about "delay" speakers. I will listen to what is coming off the back of the speaker to help me determine a good HPF. Often the energy going the wrong way can be very detrimental as its is much more delayed then it should be. Having large speakers with real pattern control is very helpful here, and will help you keep your HPF low(er).
Additionally, i have on multiple instances rolled my own FIR coefficients in the field to get phase response to line up between two different types of loudspeakers. I am not making any EQ adjustments with the FIR, just broadband phase. What is sad is when speakers within a company's own line up aren't even phase matched (ie, Vertec 4886, 4888).
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{...} One operates with fairly typical filters ( the Meyer ) and the other utilizes FIR filters ( the DXR ), which can create some weird phase shifts that are frequency-dependent when compared to another conventional product. {...}
Meyer Ultra-series loudspeakers use quite a bit more than "typical" crossover filters. The phase-wrap caused by the crossover filters is all-passed flat, so the phase response is flat from ~20 kHz down, with the first wrap only happening at 100 Hz. As I mentioned above, this produces a phase response that looks very similar to the DXR12's published FIR-based one.
{...}
I found a soundcard and threw up a condenser, not a measurment reference mic, but a high quality (can't remember which one) condenser, at least, just to look at the relative differences, not important if the mic itself is "off" comparing to what's really happening... Only measured up to 2K looked at the problem area which is lower than that. The SPL-response in REW told me what I wrote earlier what I was hearing, about 200Hz-ish wide bands which either have a dip or peak when I flip the polarity. Can't get the mains and the DXR's to align over that spectrum, it's either this response, or that..
I don't know if using that mic was worth anything to be honest, but even if the mic is "the wrong one", the relative response would be the same. Maybe it was just a coincidence that what I saw in REW matched with what I heard earlier.
{...}
Something is better than nothing, especially since you're doing comparative measurements (i.e. whatever distortion the microphone is inducing will affect all your measurements equally and therefore shouldn't matter much), but if you're just looking at the magnitude response then you're looking at the wrong information.
Since you now have some measurement tools, here's what I'd suggest: At the point where you want to hand off from the mains to the delay, take a measurement of the main speaker in isolation, then take a measurement of the delay speaker in isolation, then do whatever you need to do to make the phase traces of the two speakers be within 55 degrees of each other for any frequencies where the level from the two speakers is within 10 dB of each other. If you can post screen captures of these two separate measurements that might be helpful, too.
{...} But still I feel like I'd want to try the UP-Juniors for the tonality.
UPJuniors will be a perfect tonal (and phase-response) match for the UPQ-1Ps, and allow you to avoid any concerns about "in-box magic", but based on the published phase and frequency response of the DXR12s I'd be surprised if it actually solves your problem (and if it does then I'd extra like to see the what phase traces your system is putting out).
{...} Additionally, i have on multiple instances rolled my own FIR coefficients in the field to get phase response to line up between two different types of loudspeakers. I am not making any EQ adjustments with the FIR, just broadband phase. What is sad is when speakers within a company's own line up aren't even phase matched (ie, Vertec 4886, 4888).
I've had to do this too, but IIR. For the OP, Galileo doesn't support FIR, but each output has three all-pass filters available, which should be enough to get the job done.
(As an aside, your comment about phase-matching is one of the reasons I appreciate Meyer's approach: boxes from the same line historically have had the same phase response, and more recently their Galaxy processors make phase-matching between lines as simple as selecting from a drop-down menu, even including some of their older speakers.)
-Russ
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You guys are much more serious than I am about the delay speaker matching :).
I use the click track method. When the click slap goes away, the delay is right .... crude, but fast.
As pointed out, trying to unroll the difference in DSP between 2 speakers from 2 different companies is a tough lift.
Great thread though. Very informative.
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You guys are much more serious than I am about the delay speaker matching :).
I use the click track method. When the click slap goes away, the delay is right .... crude, but fast.
As pointed out, trying to unroll the difference in DSP between 2 speakers from 2 different companies is a tough lift.
Great thread though. Very informative.
I thought I tried the metronome method (or it may have been a looped recording of one) and I found I could because of the loop that I could be off my many Clicks and it sounded like it was right. I then changed to using a clicker that I had someone click into a mic and they were told to not do it at a fixed pace.
Then I started using SMAART many years ago and I no longer had any issue at all with setting up delay speakers.
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{...} As pointed out, trying to unroll the difference in DSP between 2 speakers from 2 different companies is a tough lift. {...}
It can be (although FIR makes that much easier); adding matching rolls to each speaker can be quite a bit easier, though.
-Russ
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A few comments in general about "delay" speakers. I will listen to what is coming off the back of the speaker to help me determine a good HPF. Often the energy going the wrong way can be very detrimental as its is much more delayed then it should be. Having large speakers with real pattern control is very helpful here, and will help you keep your HPF low(er).
This is VERY TRUE. It can be especially important to pay attention to in places like Churches with balconies that have delays hung over the main seating (to cover the balcony).
The energy coming off the back and bottom of the loudspeaker can play havoc with the seats below the speakers, if the speaker does not have pattern control down low enough.
With any loudspeaker, you need to pay attention to not only the seats it is covering, but ALSO the seats that are out of the coverage, but are affected by the loudspeaker (loss of pattern control).
Pattern control down low is a good thing. But comes at a price in physical size.
So often compromises have to be done on the front or back end of the job. Understanding what is happening is key however.
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This thread got me to thinking about how much time there is to play with FIR on delays, latency that is normally completely intolerable to live.
I think basically, the FIR time available comes down to the regular old delay setting needed.
I mean, if a delay needs say 100ms to sync up, that's worth 9600 taps of FIR playtime (100ms latency, impulse centered @48kHz.)
That much FIR could make just about any speaker have a shot at emulating the mains' response, both mag and phase. At least within linear operating range.
Heck, delays could do what mains can't ...have flat phase, better impulse response, yada yada......which then wouldn't tie together with the mains. Lol.
Anyway, new tuning tools are no doubt gonna get better it appears.
Couple futuristic processing with acoustic designs to mitigate the rear radiation problem, like what Fulcrum is doing with passive-cardioid, or like DSL is doing with size; and who knows, someday install delays may become the best seat in the house !!
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This thread got me to thinking about how much time there is to play with FIR on delays, latency that is normally completely intolerable to live.
I think basically, the FIR time available comes down to the regular old delay setting needed.
I mean, if a delay needs say 100ms to sync up, that's worth 9600 taps of FIR playtime (100ms latency, impulse centered @48kHz.)
That much FIR could make just about any speaker have a shot at emulating the mains' response, both mag and phase. At least within linear operating range.
Heck, delays could do what mains can't ...have flat phase, better impulse response, yada yada......which then wouldn't tie together with the mains. Lol.
Anyway, new tuning tools are no doubt gonna get better it appears.
Couple futuristic processing with acoustic designs to mitigate the rear radiation problem, like what Fulcrum is doing with passive-cardioid, or like DSL is doing with size; and who knows, someday install delays may become the best seat in the house !!
The other thing that you can do with delays, that might not be acceptable with the mains is linear phase multiband limiting for speakers with passive crossovers. You could do it with conventional crossovers to separate out the bands, limit them individually, and sum them back together, but you would re-add all that phase shift. However, with plenty of time on your side, you can use FIRs to achieve per driver limit thresholds (to a point) with no phase deviation at all.
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The other thing that you can do with delays, that might not be acceptable with the mains is linear phase multiband limiting for speakers with passive crossovers. You could do it with conventional crossovers to separate out the bands, limit them individually, and sum them back together, but you would re-add all that phase shift. However, with plenty of time on your side, you can use FIRs to achieve per driver limit thresholds (to a point) with no phase deviation at all.
That's pretty neat limiting solution for passives !
And I can see how it can even target very narrow frequency ranges that need limiting.
It kinda reminds me of something Linea Research / DSL have in their processors albeit with much less flexibility than you showed.
Their VX Limiters, that require extra latency to operate.
I keep finding it amazing what can be done when there is processing time to work with.
Below is an example of a 3-way main, where each driver Mid, HF, and VHF has it's own fader.
Green traces are simple on-axis mag and phase, that were tuned to on-ax only for purpose of this experiment, and is why they are so flat.
Purple is where I raised the Mid driver +10dB, cut the HF -10dB, and then raised the VHF +10dB.
I wanted to test how much phase is affected by level changes between sections, when using the steep linear phase xovers i had in place.
Reason being: Faders on each driver is a setup I routinely use for testing. Helps to isolate things quickly.
I've always been struck by the amount of tonal change that's been available using this technique, without sounding whacky.
There's been waay more room to move spectral balance around than with regular EQ's...
So i speculated it might be due to less phase change, and wanted to see how much fader levels could be altered before phase started messing up like with IIR EQs...
Found out faders can be moved a lot !
Anyway, as you can see, the purple phase with its absurd mag trace, barely peeks out from under the green phase trace.
Still amazes me. Realtime major mag change...no phase change.
I hope this wasn't too much of a sidetrek, but it seems to show an example of the power of being able to adjust mag and phase independently .....with delays, or fills, or anytime when distance provides some natural room for processing latency.
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I followed an earlier tip and just highpassed the delays higher. Couldn't hear the weirdness any more, perfect. But still some work to do. The tips in this thread are of great value.
I'll have to leave it as it is for now as another production is moving in to that stage in a few days, and I don't want to tamper with the system when there's another tech using it on a daily basis. That could lead to confusion for him.