Thanks Michael for your videos on the subject, with each one I rewatch :) I get more info out of it, still a long way to go :) What I'm struggling to understand is where to do the eq for speaker EQ and system EQ. For example at our church we have a custom line array, which was designed by a local shop (I know it's not ideal, but it's what we have) so question is, should we do a speaker EQ on the preamp we have powersoft(if I understand correctly at close range to the speaker to get it flat) then do a system eq also on the preamp? or on the mixer(x32 rack)?
The only way to parse out what tonal anomalies are due to the room vs the speaker itself is to do a near-field-ish measurement of the speaker in a controlled environment, and then apply any necessary processing to get it "flat" (a factory preset, user EQ, etc.), then you can apply another layer of EQ if it's needed somewhere in the chain to get how the speaker is positioned and sounding in the room to sound how you want it.
Hi Michael! As always - great video, lots of useful stuff, thanks for your energy, time and dedication. Have a question about last part of the video (30:40) . So we managed to rule out those peaks and dips on the magnitude graph as a floor reflections, we know that it is not a speaker oddity but a room interference. How can we estimate systems tonal response in that spectral area (low mids) "behind" those ripples? Suppose we should think of it as some kind of average trace between those peaks and dips?
HEllo Michael, what is the use of complimentary EQ, when the magnitude and Phase come back to the initial position? isn't it a waste of processing power of the processor?
Will you be doing a video on the system design with the SRX906 boxes you mentioned in the video? There's not a ton of real world experience reviews and opinions on them.
I hope to in the future once I can actually hear the boxes in person. The data I showed in the video was for a design I helped with, but it was a remote consultation.
We don't. It shows up on the measurement when the room is empty, but with a packed standing audience it's mostly absorbed away. To not be bothered by it while taking a measurement we could put the microphone really close to the ground, thus pushing the comb filter way up in the high frequencies. Or you learn by experience how to filter it away while reading the measurement.
Hi Mike, much appreciate all the quality info you put out for us. Based on this video about EQ and phase I would like to ask you about the following scenario. I have a high quakity Martin audio CDD live 15 self powered speaker and I'm using it with their WLX horn loaded sub. I have the proper DSP setting in my proc for the sub which is low passed at 100Hz with a LR 48/Oct filter. My question is in regard to the CDD live 15 which is should I use just a HPF or use a LR filter to limit the very lowest frequncies from the CDD? I have done OSM measurements and the HPF filter for the CDD's is less disruptive to the phase response of the system. You should be aware that the system DSP has seperate outs for the mid/highs (CDDs) from the subs so they do noyt share a crossover point. Thanks for any input you can share.
It's hard for me to say without looking at the data, but I would choose the filter that provides the smoothest magnitude and phase response transition between the main and the sub. I'd try measuring them side by side, capturing traces with the different settings and see which you can get the phase response to line up better with.
In the case of curved linear array we should start with an observation and response that is perhaps 5 or more octaves wide! "array correction"! I point to this as it is a very common oversight, many of us don't use array correction because we incorrectly feel it is robbing us of potential low hz output. I think that Meyer M array was without doubt the best implementation of array correction ever hands down! Yet most "engineers' scoffed at it. As result the task of "array correction" often unknowingly gets smeared into system tuning as the person does not even realize they have blended tasks that should perhaps not be blended! If AC is dealt with separately in another EQ application point it helps to not modify the tuning SE response because of strange looking filter compilations! That in its self is a subject visual expectations of EQ filter choices. IF commercial aviation trained in the same manner as Live sound we would crash and die at the wrong airport!
Great points here, Gord! It's also interesting to hear how different manufacturers approach the array compensation. Most have opted for a variable low shelf based on the number of boxes. However, to my surprise, I've found that the QSC KLA12 stuff actually adjust the HF when you adjust the "array size" knob, similar to a JBL VRX932LAP. From array 1-5 it's a ~10dB swing.
Yes, with a system processor I'm able to use more filters. If you've only got four I would still use the same approach and start broad, then move to more focused.
@@MichaelCurtisAudio I know those AHM processors have phoenix terminals. How do you connect your system? Do you have it in a rack with panel mount connectors?
@@WilliamOgden that would be the way to go, yes. Alternatively you can go digital in and out of the box and have a stagebox somewhere for those outputs. But thats an edge case that adds needless complexity to the system
@@MichaelCurtisAudio sometimes I use powered speakers which have EQ built in , so I get 3 more EQ filters if I run out from the mixer. I rarely need more than 4 filters anyways :) If you think about it, there's 4 filters from Matrix LR and there's another 4 from Main LR as well. If you used up all 8, you might as well run the system EQ flat at lower levels. LOL
@@MichaelCurtisAudio i am Mastering engineer in Oslo Mastering. How do you cope to work with a flat curve? In my ears is not sound natural and is isn't. Read some Aes paper from Tool. The mixer who use flat curve in studio, often have to much bass, and to little treble. Nice video btw
@Michael Curtis Thanks for that info. Will it work on win 11 pro? Also will I need an interface and reference mics to get started? I really want to learn all this especially for driver, sub, fill alignment, and for checking speaker and horn driver phase etc.. The videos are already bogging down my brain 🧠 I've always gave great mixes in my time of mixing but it's time to master the visual tools 🔧
Thank you so very much! I appreciate the feedback. I'll have to check out your videos soon. Looks like you hang out more in the studio/home theater space? Or do you do live as well?
You have a fantastic teaching ability, very clear to understand. Great channel, thanks Michael.
You're very welcome!
thank you so much for taking the time to create this, just started learning PA tuning, and this is extremely helpful!!
I tried this yesterday in my small PA system and it got me really nice results, thanks!
Glad to hear that!
hey Michael! if i were to turn the subs around and get the driver closer to the wall, would i then flip the phase to 180 to compensate? thanks!
Thanks Michael curtis for always being informative 👍👍👍👍
Pure Gem, much respect to you!
Nice vid...Pat is great!
Amazing video once again, thanks Michael!
Much appreciated!
Thanks Michael for your videos on the subject, with each one I rewatch :) I get more info out of it, still a long way to go :)
What I'm struggling to understand is where to do the eq for speaker EQ and system EQ.
For example at our church we have a custom line array, which was designed by a local shop (I know it's not ideal, but it's what we have) so question is, should we do a speaker EQ on the preamp we have powersoft(if I understand correctly at close range to the speaker to get it flat) then do a system eq also on the preamp? or on the mixer(x32 rack)?
The only way to parse out what tonal anomalies are due to the room vs the speaker itself is to do a near-field-ish measurement of the speaker in a controlled environment, and then apply any necessary processing to get it "flat" (a factory preset, user EQ, etc.), then you can apply another layer of EQ if it's needed somewhere in the chain to get how the speaker is positioned and sounding in the room to sound how you want it.
Hi Michael! As always - great video, lots of useful stuff, thanks for your energy, time and dedication. Have a question about last part of the video (30:40) .
So we managed to rule out those peaks and dips on the magnitude graph as a floor reflections, we know that it is not a speaker oddity but a room interference. How can we estimate systems tonal response in that spectral area (low mids) "behind" those ripples? Suppose we should think of it as some kind of average trace between those peaks and dips?
That's exactly right! And the deeper the ripple, the stronger the reflection. The smaller the ripple, the weaker the reflection.
Thanks Man,just what I needed,as if you know what my brain just thinking this morning,there you are again enlightening Us again,sincerely Your Student
Glad you found it help, Montel.
HEllo Michael, what is the use of complimentary EQ, when the magnitude and Phase come back to the initial position? isn't it a waste of processing power of the processor?
Will you be doing a video on the system design with the SRX906 boxes you mentioned in the video? There's not a ton of real world experience reviews and opinions on them.
Yeah I would like to see this too. Maybe a comparison against HDL6A
I hope to in the future once I can actually hear the boxes in person. The data I showed in the video was for a design I helped with, but it was a remote consultation.
What a cool video.... how could we correct the floor reflection comb filter??
We don't. It shows up on the measurement when the room is empty, but with a packed standing audience it's mostly absorbed away. To not be bothered by it while taking a measurement we could put the microphone really close to the ground, thus pushing the comb filter way up in the high frequencies. Or you learn by experience how to filter it away while reading the measurement.
Which software did you use to simulate the SRX900? :) 6:40
jbl lac 3
Hi Mike, much appreciate all the quality info you put out for us. Based on this video about EQ and phase I would like to ask you about the following scenario. I have a high quakity Martin audio CDD live 15 self powered speaker and I'm using it with their WLX horn loaded sub. I have the proper DSP setting in my proc for the sub which is low passed at 100Hz with a LR 48/Oct filter. My question is in regard to the CDD live 15 which is should I use just a HPF or use a LR filter to limit the very lowest frequncies from the CDD? I have done OSM measurements and the HPF filter for the CDD's is less disruptive to the phase response of the system. You should be aware that the system DSP has seperate outs for the mid/highs (CDDs) from the subs so they do noyt share a crossover point.
Thanks for any input you can share.
It's hard for me to say without looking at the data, but I would choose the filter that provides the smoothest magnitude and phase response transition between the main and the sub. I'd try measuring them side by side, capturing traces with the different settings and see which you can get the phase response to line up better with.
@@MichaelCurtisAudio Thanks for the quick response Mike. Perhaps one day a video about LR, Butterworth and Bessel filters!
Where can I download the Michael Lawrence target curve from?
drive.google.com/open?id=1wLFuF0yMybNNJE1NM2KD65b66SdNF_IV&usp=drive_fs
What is that
Sir Michael target curve
Great video! What software/interface are you using for your measurements?
Thanks, Martin! I use the Audient EVO 8 and SMAART v9.
In the case of curved linear array we should start with an observation and response that is perhaps 5 or more octaves wide! "array correction"! I point to this as it is a very common oversight, many of us don't use array correction because we incorrectly feel it is robbing us of potential low hz output.
I think that Meyer M array was without doubt the best implementation of array correction ever hands down! Yet most "engineers' scoffed at it. As result the task of "array correction" often unknowingly gets smeared into system tuning as the person does not even realize they have blended tasks that should perhaps not be blended!
If AC is dealt with separately in another EQ application point it helps to not modify the tuning SE response because of strange looking filter compilations! That in its self is a subject visual expectations of EQ filter choices.
IF commercial aviation trained in the same manner as Live sound we would crash and die at the wrong airport!
Great points here, Gord!
It's also interesting to hear how different manufacturers approach the array compensation. Most have opted for a variable low shelf based on the number of boxes. However, to my surprise, I've found that the QSC KLA12 stuff actually adjust the HF when you adjust the "array size" knob, similar to a JBL VRX932LAP. From array 1-5 it's a ~10dB swing.
Great stuff!!!
Thanx!!!
How did you have the luxury of using so many EQ filters? Most of us only get 4 paramatic filters on a digital mixer on the outputs.
He brings an Allen and Heath system processor for system tune and alignment.
Yes, with a system processor I'm able to use more filters. If you've only got four I would still use the same approach and start broad, then move to more focused.
@@MichaelCurtisAudio I know those AHM processors have phoenix terminals. How do you connect your system? Do you have it in a rack with panel mount connectors?
@@WilliamOgden that would be the way to go, yes. Alternatively you can go digital in and out of the box and have a stagebox somewhere for those outputs. But thats an edge case that adds needless complexity to the system
@@MichaelCurtisAudio sometimes I use powered speakers which have EQ built in , so I get 3 more EQ filters if I run out from the mixer. I rarely need more than 4 filters anyways :) If you think about it, there's 4 filters from Matrix LR and there's another 4 from Main LR as well. If you used up all 8, you might as well run the system EQ flat at lower levels. LOL
Hi first of all thanks for all your videos..i would like to know if it is possible to sound tuning a system with an ipad plzz ?
You can use the Open Sound Meter app on an iPad and connect an iOS compatible interface to it with your measurement mic.
I don’t have a pc but an ipad pro and i digital mixer 😢
You should have a target curve also in studio. Harmon curve
I personally prefer flat, but the Harman curve can be used in certain situations.
@@MichaelCurtisAudio i am Mastering engineer in Oslo Mastering. How do you cope to work with a flat curve? In my ears is not sound natural and is isn't. Read some Aes paper from Tool. The mixer who use flat curve in studio, often have to much bass, and to little treble.
Nice video btw
I wish I had your brain and knowledge. I'd purchase smaart if I could get some grip on things.
Start with Open Sound Meter and get the hang of measurement software, then upgrade to Smaart when you're ready.
@Michael Curtis Thanks for that info. Will it work on win 11 pro? Also will I need an interface and reference mics to get started? I really want to learn all this especially for driver, sub, fill alignment, and for checking speaker and horn driver phase etc.. The videos are already bogging down my brain 🧠
I've always gave great mixes in my time of mixing but it's time to master the visual tools 🔧
Your videos are excellent! Would love to chat sometime.
Thank you so very much! I appreciate the feedback. I'll have to check out your videos soon. Looks like you hang out more in the studio/home theater space? Or do you do live as well?
@@MichaelCurtisAudio Mostly home theater, home audio, and I do car audio stuff for my own learning experience. I sent you a message on IG.