The issue is that the EQ settings should account for the occupancy of the venue. The acoustics differ significantly between an empty space and one filled with people. When you tune your PA system in an empty venue, it may sound good initially, but as the venue fills up, the sound quality often deteriorates. This happens because the initial measurements were taken in an empty space, which doesn't accurately reflect the conditions when the venue is full.
@@Dxcter_C0rgi207the high frequencies bounce off the boundary layer of cool air that meets the warm air above people's heads.... basically you want to increase the attack angle of the speaker by getting it high enough and then pointing it down at the heads. This helps the HF penetrate the boundary layer. The extra heat from the bodies can also cause the lower mids to travel faster and cause additional build up of energy.....so by the time the room is full of people you may need to cut some more low end out. Old school method was to place a graphic EQ just before the amps, and then just adjust the bands to suit..... usually 160, 315, 615, 2.5, 3.15, 4k.....check out Dave Natale's EQ curve...he did sound for Van Halen, Tina Turner, Rolling Stones etc...its pretty much the secret recipe for a bus EQ
@@DW_drums thankyou so much, such a detailed response. It did indeed work I have now pointed my spekers down a bit since they were wall mounted but flat against the top of the wall.
@@marcuspainter6861 The high level pro's use RTA like you say, however it's usually to see if there is some kind of failure at the speakers i.e blown sub etc. and also monitor their frequencies as they hit the dB limit ie if you want to go louder without going over the dB limit you can play with certain frequencies.
I'm more old school personally, the volume of the main foh is the benchmark to my way of sorting any venue, it is after all where the main action is focussed when the house has bums on seats, the 3 frequencies indeed do travel at their natural speeds so a simple adjustment to the frequency of each usually realigns any obvious latency issues, the science often gets in the way of the art too much these days and tends to make the process cluttered in my opinion, we use our foh only at room set intention and when it's kicking well in the main audience zone we then during our sound check slowly introduce our side fill and floor stage monitors to come into phase only with the foh from our stage listening as the players first, lots of players sound crap on stage to please the sound guy mainly, both are winners when the foh main is set correctly first, the fills elsewhere around the room if needed are only like monitors for the audience and not considered as able to replicate the main thrust of foh. The drums if conventional are the usual phase-setter and the amps by guitar/bass/keyboard etc are not set higher than the already noisey drummer and when it's right on stage for the players first, it's going to be easier for it to be replicated in the p.a. and stage monitors and audience fills if any are required to match, the mixer if right on foh it can supply a copy to the monitors and it's less fluffing around, the process works in reverse too if required by letting the players get themselves great on stage first then sorting the foh next the monitors next...room fills can be included if needed when those 3 are in phase effortlessly.
I think there is a reason why the SMART is showing the phase reading topmost. If you adjust the eq without doing delay compensation first, your reading might be way off due to interference between multiple speakers. It's like miking a drum kit, first place the mics and make sure the phases are right, then get the eq sorted, otherwise you might be eq-ing the phase issues. Align time first, then eq multiple source systems.
Hello and thanks for the video.😊 I have a question about the monitor. How little / how much time do you spend tuning monitors? Thanks in advance for your help and for sharing your knowledge🙏 🤓
It's a basic process of 'triangulate' to find the optimum listening point to set any p.a. from or for the rest of the space, anything added beyond such is only replication of the ideal listening point which is the same distance directly in front of the p.a. enclosures being apart as foh. ...anything else is mere replication intent or additional monitors or 'fills' as you've referred to them as.
Great video, but shouldn’t time/phase alignment come prior to EQ’ing to the target curve? For example: The bump in low end you just equalized could be due to the speakers misaligned. Fixing the alignment could result in a double increase/decrease of the level. Plus: Crossover points change with level and thus with EQ. Sure, you can only align the whole system at one listening point, but at least you can make sure it's properly aligned >somewhere.
Hi, thanks for your Video! It really made some things clear and helped me to understand the process :) However, I have a question: Are you doing the same to your Monitors at FOH so the sound you hear when mixing is the approximate sound in front of the Stage?
Hey, do you mean like near field monitors? I've never been in a position where I've had them or felt I needed them but yeah I'd like the response to match the rest of the system
I wouldn't recommend this, if anything I'd say do the opposite. We lose high frequencies disproportionately to low frequencies over distance. If you remove the highs from the delays then you're removing the purpose of the delays, which is to restore the frequency response and level to our target curve or within 6db of it. If you find people are localising the source to the delay speakers, Use a little more delay and thus the precedence effect to make people localise the sound to the stage while maintaining your desired level and tonality
rule #1 use your ears not your eyes I see so many people doing things like subs L-R of stage and then wondering why there are power alleys and valleys. speakers are not lights
I agree, but sometimes you have to do LR placement of subs, because of client or artist (don't want subs in center or it's not possible to put subs in center), it depends. It's great video.
It would've been good to get more information about system tuning when you're not using fills. I know you've got other videos that focus more on standard LR systems, so maybe just rename this video to highlight the difference? Great video though!
In the target trace part you clam that human hate flat system, is there a evdience? Because in contrary in most of Floyd Tooles paper mentions that anechoic flat system were prefered across both normal and trained listeners.
In his paper, The Measurement and Calibration of Sound Reproducing Systems, he quotes a study where listeners were allowed to EQ a flat system to their preference. Untrained listeners prefer more bass and more treble. Trained listeners preferred slightly more bass but a reduction in treble. See Figure 14 p 528. www.linkwitzlab.com/Toole-Room%20calibration.pdf
@@marcuspainter6861 We are actually quoting the same paper. I thought the point of Figure 14 is to tell you that trained listener have a preference that is very close to a anechoic flat frequency response speaker which is written in 4.1 section. Untrained one just want more of everything which is maybe just volume. And so for a room with controled acoustics the right answer is maybe flat. When using smaart we have the choice of complex averaging which is not steady state measurement, it rejects everything outside of the window and thus can be closer to anechoic measurement. I use +12dB Sub as a default too, but I use group send matrix instead of LR send and take that as a compromise for gain structure.
@@Honaiahak It could be that untrained listeners just prefer louder. Louder does tend to sound better, but it is also is similar to Fletcher-Munson equal loudness curve, meaning that untrained listeners may prefer perceptively flat. Are they compensating for the ear's natural unflat sensitivity? Maybe a coincidence.
Let me ask, why do I have to get the measurement of the on-axis and further back the intended listening area? Is the measurement in the intended listening area not enough? Also can I just measure with the sub already on? Please answer thanks!
22:07 I believe there's s mistake here ("apply delay to mains first"), or creates confusion IMO. One doesn't need to apply delay to the mains, unless one is tuning the array of doing some sort of beam steering, and this stuff is often performed in the system processor. If that's the case then yes, correct. But for the purpose of alignment, mains are most of the times the point of reference, because most of the times subs and front fills are nearer to the audience, so those are the ones who need to have delay applied. Correct me if I'm wrong? If we are talking about flown subs then the conversation changes. Cool video, thanks 👍🏻
You're very correct about about the mains being your point of alignment. However I said apply Delay compensation to the measurement with your main before turning on the fills. I can see how that's confusing. To set the record straight, that's applying delay to the measurement within smart to account for the travel time from the mains to the mic.
@@OffshoreAudio yes that's makes more sense. The measurement and the reference signals should be synced to the mains. My bad, I'm not a English native speaker. 🤦🏻
What if for time alignment, you measure the distance from any speaker, to the main PA? Then calculate the delay and apply it on the mixer? Isn't that simpler than using Smart?
You could, but then you're making assumptions about variables that affect the result like temperature and humidity. If you measure using smaart then you're measuring the real world result and aligning actual arrival times. I have done it with a laser measure and ultimately you can do whatever you like. You can do it totally by ear if you want, it'll just be less accurate and less consistent.
@@josh88745 I don't think so. Calculating the delay is distance over spead of sound in air. And most consoles you just input the distance. Hence me saying it's simpler🤷
Greetings from Boston Massachusetts USA! What a great tutorial. That helped me understand a bunch of things in a clear way. (No pun intended). Btw, is that a Scottish accent??? Whatever it is, I love it!
Simply not true… Ever worked on a large-scale festival with multiple hangs, fills and audience zones? Tell the system techs to do it without proper simulation and tuning. There is a reason, today’s festivals are bigger and offer a better listening experience. It all comes down to the development of measurement tools and software, too.
@lhenn_ To add to your comment - since I began methodically tuning systems from venue to venue my mixes have become a million times more consistent. And you can really hear the differing responses of each PA manufacturer.
The issue is that the EQ settings should account for the occupancy of the venue. The acoustics differ significantly between an empty space and one filled with people. When you tune your PA system in an empty venue, it may sound good initially, but as the venue fills up, the sound quality often deteriorates. This happens because the initial measurements were taken in an empty space, which doesn't accurately reflect the conditions when the venue is full.
yea thats true, how would we tackle that?
@@Dxcter_C0rgi207the high frequencies bounce off the boundary layer of cool air that meets the warm air above people's heads.... basically you want to increase the attack angle of the speaker by getting it high enough and then pointing it down at the heads. This helps the HF penetrate the boundary layer. The extra heat from the bodies can also cause the lower mids to travel faster and cause additional build up of energy.....so by the time the room is full of people you may need to cut some more low end out. Old school method was to place a graphic EQ just before the amps, and then just adjust the bands to suit..... usually 160, 315, 615, 2.5, 3.15, 4k.....check out Dave Natale's EQ curve...he did sound for Van Halen, Tina Turner, Rolling Stones etc...its pretty much the secret recipe for a bus EQ
@@DW_drums thankyou so much, such a detailed response. It did indeed work I have now pointed my spekers down a bit since they were wall mounted but flat against the top of the wall.
Dual channel analysers like SMAART can measure during a live show. Does anyone do this?
@@marcuspainter6861 The high level pro's use RTA like you say, however it's usually to see if there is some kind of failure at the speakers i.e blown sub etc. and also monitor their frequencies as they hit the dB limit ie if you want to go louder without going over the dB limit you can play with certain frequencies.
I'm more old school personally, the volume of the main foh is the benchmark to my way of sorting any venue, it is after all where the main action is focussed when the house has bums on seats, the 3 frequencies indeed do travel at their natural speeds so a simple adjustment to the frequency of each usually realigns any obvious latency issues, the science often gets in the way of the art too much these days and tends to make the process cluttered in my opinion, we use our foh only at room set intention and when it's kicking well in the main audience zone we then during our sound check slowly introduce our side fill and floor stage monitors to come into phase only with the foh from our stage listening as the players first, lots of players sound crap on stage to please the sound guy mainly, both are winners when the foh main is set correctly first, the fills elsewhere around the room if needed are only like monitors for the audience and not considered as able to replicate the main thrust of foh.
The drums if conventional are the usual phase-setter and the amps by guitar/bass/keyboard etc are not set higher than the already noisey drummer and when it's right on stage for the players first, it's going to be easier for it to be replicated in the p.a. and stage monitors and audience fills if any are required to match, the mixer if right on foh it can supply a copy to the monitors and it's less fluffing around, the process works in reverse too if required by letting the players get themselves great on stage first then sorting the foh next the monitors next...room fills can be included if needed when those 3 are in phase effortlessly.
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Another great video: keeping it simple keeps it clear!
Thanks. That's exactly what I was going for!
Very very well done!
Thank you!
Thank you for this amazing video! Great Tutorial! This is pure gold ❤
You're so welcome! Very glad to hear it's helpful
I’ve been looking for a video on how to tune a system I’ve read books this by far is the best thing I’ve ever found for mixing a system
It really is a fantastic piece of internet this! 👌
Aw thank you so much. This means a lot ❤️
This is an area I have struggled to get my head around. I kinda get it, but this really made the penny drop!
Thanks dude 🤘🔊🔊
So glad it was helpful. It's definitely a complex topic so if I can help you on your way then I'm happy
I think there is a reason why the SMART is showing the phase reading topmost. If you adjust the eq without doing delay compensation first, your reading might be way off due to interference between multiple speakers. It's like miking a drum kit, first place the mics and make sure the phases are right, then get the eq sorted, otherwise you might be eq-ing the phase issues. Align time first, then eq multiple source systems.
This is the best video I’ve seen yet on time alignment!! Cheers from AUSTRALIA 🇦🇺
Aw you're too kind! Cheers!
Hello and thanks for the video.😊
I have a question about the monitor.
How little / how much time do you spend tuning monitors?
Thanks in advance for your help and for sharing your knowledge🙏
🤓
It's a basic process of 'triangulate' to find the optimum listening point to set any p.a. from or for the rest of the space, anything added beyond such is only replication of the ideal listening point which is the same distance directly in front of the p.a. enclosures being apart as foh. ...anything else is mere replication intent or additional monitors or 'fills' as you've referred to them as.
Great video, but shouldn’t time/phase alignment come prior to EQ’ing to the target curve? For example: The bump in low end you just equalized could be due to the speakers misaligned. Fixing the alignment could result in a double increase/decrease of the level.
Plus: Crossover points change with level and thus with EQ.
Sure, you can only align the whole system at one listening point, but at least you can make sure it's properly aligned >somewhere.
I would agree. Changing the time alignment can affect the summation in the main/sub crossover region.
Hi, thanks for your Video! It really made some things clear and helped me to understand the process :)
However, I have a question:
Are you doing the same to your Monitors at FOH so the sound you hear when mixing is the approximate sound in front of the Stage?
Hey, do you mean like near field monitors? I've never been in a position where I've had them or felt I needed them but yeah I'd like the response to match the rest of the system
You remove the highs from the delay speakers (shelving) to give the impression the sound is coming from the main speakers ie the stage.
I wouldn't recommend this, if anything I'd say do the opposite. We lose high frequencies disproportionately to low frequencies over distance. If you remove the highs from the delays then you're removing the purpose of the delays, which is to restore the frequency response and level to our target curve or within 6db of it. If you find people are localising the source to the delay speakers, Use a little more delay and thus the precedence effect to make people localise the sound to the stage while maintaining your desired level and tonality
👌
rule #1 use your ears not your eyes I see so many people doing things like subs L-R of stage and then wondering why there are power alleys and valleys. speakers are not lights
I agree, but sometimes you have to do LR placement of subs, because of client or artist (don't want subs in center or it's not possible to put subs in center), it depends. It's great video.
@@tomazklancnik7760 yeah I’m talking about the people that preferred to set up their subs left right because it looks like it’s the most even coverage
@@tomazklancnik7760 I'd rather put them together on one side or the other.. sounds cleaner
Center sub is to prevent power alleys and valleys.
@@jaken528 I’ve been doing that lately you get so much more output on top of getting rid of the dead spots
What microphones and software are you using to set your eq level?
It would've been good to get more information about system tuning when you're not using fills. I know you've got other videos that focus more on standard LR systems, so maybe just rename this video to highlight the difference? Great video though!
If the right mic, is in the right place, with the right desk and correct speakers, is eq necessary? 🧐
There is no "right" in audio. Everything is a compromise.
There is no answer to your question because what you are asking about doesn't exist.
@16:45 did you measure Top and Sub together ?
Yeah. You can see the measurement change when I turn the sub on.
In the target trace part you clam that human hate flat system, is there a evdience? Because in contrary in most of Floyd Tooles paper mentions that anechoic flat system were prefered across both normal and trained listeners.
In his paper, The Measurement and Calibration of Sound Reproducing Systems, he quotes a study where listeners were allowed to EQ a flat system to their preference. Untrained listeners prefer more bass and more treble. Trained listeners preferred slightly more bass but a reduction in treble. See Figure 14 p 528. www.linkwitzlab.com/Toole-Room%20calibration.pdf
@@marcuspainter6861 We are actually quoting the same paper. I thought the point of Figure 14 is to tell you that trained listener have a preference that is very close to a anechoic flat frequency response speaker which is written in 4.1 section. Untrained one just want more of everything which is maybe just volume. And so for a room with controled acoustics the right answer is maybe flat. When using smaart we have the choice of complex averaging which is not steady state measurement, it rejects everything outside of the window and thus can be closer to anechoic measurement.
I use +12dB Sub as a default too, but I use group send matrix instead of LR send and take that as a compromise for gain structure.
@@Honaiahak It could be that untrained listeners just prefer louder. Louder does tend to sound better, but it is also is similar to Fletcher-Munson equal loudness curve, meaning that untrained listeners may prefer perceptively flat. Are they compensating for the ear's natural unflat sensitivity? Maybe a coincidence.
Let me ask, why do I have to get the measurement of the on-axis and further back the intended listening area? Is the measurement in the intended listening area not enough? Also can I just measure with the sub already on? Please answer thanks!
To ensure the whole listening area is consistent. I explain at 16:00 - 16:30 in the video
22:07 I believe there's s mistake here ("apply delay to mains first"), or creates confusion IMO. One doesn't need to apply delay to the mains, unless one is tuning the array of doing some sort of beam steering, and this stuff is often performed in the system processor. If that's the case then yes, correct.
But for the purpose of alignment, mains are most of the times the point of reference, because most of the times subs and front fills are nearer to the audience, so those are the ones who need to have delay applied. Correct me if I'm wrong?
If we are talking about flown subs then the conversation changes.
Cool video, thanks 👍🏻
You're very correct about about the mains being your point of alignment. However I said apply Delay compensation to the measurement with your main before turning on the fills. I can see how that's confusing. To set the record straight, that's applying delay to the measurement within smart to account for the travel time from the mains to the mic.
@@OffshoreAudio yes that's makes more sense. The measurement and the reference signals should be synced to the mains.
My bad, I'm not a English native speaker. 🤦🏻
@@DiogoCocharro no worries! it is confusing. it's a complicated topic and difficult to make it straightforward
What if for time alignment, you measure the distance from any speaker, to the main PA? Then calculate the delay and apply it on the mixer? Isn't that simpler than using Smart?
i’m pretty sure this is what he was saying
You could, but then you're making assumptions about variables that affect the result like temperature and humidity. If you measure using smaart then you're measuring the real world result and aligning actual arrival times. I have done it with a laser measure and ultimately you can do whatever you like. You can do it totally by ear if you want, it'll just be less accurate and less consistent.
@@OffshoreAudio I understand. Thx a lot.
@@josh88745 I don't think so. Calculating the delay is distance over spead of sound in air. And most consoles you just input the distance. Hence me saying it's simpler🤷
Greetings from Boston Massachusetts USA! What a great tutorial. That helped me understand a bunch of things in a clear way. (No pun intended). Btw, is that a Scottish accent??? Whatever it is, I love it!
Thank you so much! So happy to hear this. And yes! No one ever guesses the accent correctly hahaha 😁
System tuning is overrated
Simply not true… Ever worked on a large-scale festival with multiple hangs, fills and audience zones? Tell the system techs to do it without proper simulation and tuning. There is a reason, today’s festivals are bigger and offer a better listening experience. It all comes down to the development of measurement tools and software, too.
@lhenn_ To add to your comment - since I began methodically tuning systems from venue to venue my mixes have become a million times more consistent. And you can really hear the differing responses of each PA manufacturer.