Having heard his work over the decades, I can assure that He does know what he is talking about. During presentations like this, it would be nice to insert the screen image that he is referring to, thus enabling a clearer explanation of the process. Also, If a question or answer is offered, it would also help to restate said comment for the remote listener. I have used the same approach and understanding of concept for over 10 years with digital and agree that this is valid on all counts.
Hi Robert, i have met you twice in India. But, your explanation of consoles bulit optimsed to line level signal is the best i have come across. Three weeks back, i did my best to expalin to my team on the concept of dealing with a line level signal and mic level signal. It also explains why many programmed drums and synrhs sounded as they due low bit rate..
for all corp talking heads i set pres like in a studio, run mixes at 0dbvu minimum and use inline pads to make the room work on shitty powered speakers. i usually set at -20/-25 db to feed the boxes. this way all feeds are optimal. in a proper room with a line array, not needed. i'm talking shitty self powered JBL type boxes. Works a treat for me!
This is absolutely sound thinking. For those who disagree and use gain structure to reflect musical composition you are making a compromise. There’s nothing wrong the approach and in analogue this was normal practice. But it is a compromise. You sacrifice perfect gain structure for musical taste, or you sacrifice nominal fader position for musical taste. There’s no other way out… except subgroups. A VCA is just another fader compromise. So you need to know how your console manages processing in terms of latency. Everything we do in FOH is about compromise. Subgroups are the only way to mix without compromise in the digital world.
Need to ask: If I have a kick that’s too loud, is Robert suggesting I route pre-amp -> fader (unity) -> kick sub group -> whole kit sub group -> master LR? That would require going bus to bus. It would also mean that any time I have a channel that needs its level adjusting (probably most or all of them) I need to send it to a bus before the master LR. I don’t have enough busses to do that. Forgive me, am I missing something or is he seriously suggesting that?
This situation he's describing with the kick that is too loud - is he saying that instead of turning the gain down on the channel (which would reduce the optimal signal quality coming into that channel), and instead of turning down the master LR fader (which is set at its optimal travel point for mix control, and for other instruments that might already be coming through at the right kind of level), and instead of pulling the fader down on the channel itself, you should send that bass drum to a subgroup after it's been processed, and just before the main outs, where that subgroup reduces the level of the signal? If I've got a kick that's coming in way too loud compared to everything else, then if I keep its gain at the optimal level for processing (even though it sounds too loud) then surely I'm going to have to pull it down in any monitor mixes that I have as well? As a practical approach to mixing, I've always set the channel faders at unity and the main mix fader at or slightly below unity, and then I've tried to get a rough mix where each input sounds roughly right. If the visual meters are telling me a signal is way too low or too high coming into the desk, and it still doesn't sound the right volume in the room, it's usually an issue with my amp levels. Thus, what "sounds" right in a balanced mix on the gain pots is my starting point for a balanced monitor and front-of-house mix. I'm not always trying to get things to be peaking at their optimal levels, but rather just using the meters to show me if there's a signal that is far too low or far too high. If I wanted the kick to be lower in the mix (e.g. for aesthetic reasons) I don't understand why it's a problem to pull down the fader. I want the faders to represent the actual mix - so that if something is at -10, then it's probably quite subtle in the mix, and if something's above unity, it's very prominent. I don't know if I'm fundamentally misunderstanding his point, or if I've got it, or if my approach is entirely different to his? I do appreciate that a gain meter that is barely dancing or is constantly peaking is a bad thing, but I've felt that in between those extremes, what my ears tell me is right when all other faders and sends are at unity, that tends to be where I leave the gain knob alone. So please be forgiving if you think I'm being obtuse - genuinely keen to learn.
I sort of agree with you. His point I guess was that if you have to pull the channel fader way down you are out of the realm of "hires". So every movement you have to make from then on out will be far less subtle then when it could have been at unity. But for me (and I'm absolutely not questioning his skills or knowledge, I'm rather convinced that this is not the whole story that he told) there is missing quite.a lot information. First of all, gain structure. in my opinion is more than the little knob on the console. It has to do with the whole chain from signal source to speakers and all the variables that are in between. Furthermore, the point that digital is less forgiving within context of gain and signallevel than analog is a little bit understated. Because especially the headroom issue is not addressed but quite important in my opinion.
What he's saying is mix the drums, for example, into a bus or mix group, then adjust the group's level (fader) to the Master/Main mix. Likewise with other channel groups - vocals, instruments, etc.
From the sound of it, you will need one subgroup for each “problem” input channel to optimise the sound - if you have the same band performing every week, I guess it is doable. But I usually have to set up shows within 2 hrs from scratch for different bands because venue hire is expensive. That’s just barely enough time to unload everything from the dock, connect and sound check, assume the talents arrive on time. So I just don’t think I can afford to do this level of optimisation. I am also confused on the concept of bit rate - I thought a DAC/ADC conversion bit rate is always constant regardless of the input voltage. The perceived drop in sound quality when the input is too low is due to signal to noise ratio. Happy for someone to correct me here.
The bitrate is the same but if your peaks are only reaching half the full scale range, then you’ll only use half of the available bits. Look it like hooking up a 1080p device to a 4k screen
Everything's good except that instead of lowering the levels using subgroups (if the PA is overpowered), I would lower the volume as far as possible in the chain in order to keep an optimal level up to the Master Fader. I would ideally dim down the power amps with the benefit of reducing the self noise of the PA, otherwise the PA processor, or as a last resort lower the post master output on the console.
Each bit represents 6db of dynamic range. When he says "hanging at zero" he means the "0" (unity) position on the fader/trim meters which is actually -18dbfs ...which equals 3 bits. Hard to tell without seeing his slide/graphics. A signal hitting 0dbfs (the digital ceiling) would in fact use up all 24 bits like you're saying.
So just to make sure I understand correctly: ideally, assuming we gain staged our inputs correctly, every fader would be set to 0dB, and we would be mixing with strictly subgroups?
I guess I'm confused, how do you make subtle balance adjustments if everything is ideally kept at 0db? I can' imagine making a sub-group for each individual channel as well as subgroups for groups of instruments, like an all drum group.
@@RonMikeGabe He does not make subgroups for each channel only the instrument groups. As a rule keeping the individual faders between +10 and -10, he calls this high resolution mixing. He uses the subgroups to adjust for the level differences between screaming guitar amps and thundering drums and comparatively low level vocals.
i know this is like a year later...but is there a way were i can get those notes on the screen just because i want to keep it like in a pdf or somethig
So, when is the signal at line level exactly... when the RMS at the input gain meter is properly set or when in addition to that the channel fader is also at unity?
You'd start with the faders at unity (initially) to ensure full resolution of the signals. Then, gain stage your signals to reach the "line level" of +4dBu (0dBVU on a voltage meter) by driving gain into the signal with the preamp. That's the "sweet spot." This level is roughly equivalent to 0dBVU on an analog voltage meter or -18dBFS on a meter within a digital environment like a DAW. Your signal's average should hover around this spot on your meters. Some meters--like those in some DAWs--show RMS. In that case, this is also where you could stage the signals, especially if you are working with any analog hardware or plugins. For live sound, you'll be looking to get weaker signals like a "mic line" boosted up to the +4dBu target.
This is fantastic information - I was puzzling between a DCA and a Subgroup to make this work correctly- I am having to run my Subgroup master faders quite low on the console (around -20 db) - Is this normal? Or is it perhaps that there is to much power in the PA itself, and this needs to be adjusted down?
You can just pull your master fader further down to push the subgroup masters further up to gain more control over them. Alternatively you can turn down your PA amps to gain more control over your master fader too.
@@danscu5278 Brilliant, Thank you! I have considered this and will work on this in my workplace. Using subgroups now in my mixing has been a game changer! I wish I knew this years ago
Someone correct me if I'm wrong but... Line level = ~1V (about 1V of signal) Inst Level = (basically anywhere in between) Mic level = mV (milliVolts of signal)
Line level = around 1 volts + low to medium impedance (from several ohms to about 1-2k) Mic level = below 50mV + low impedance (most are less than 300 ohms) Instrument level = high impedance(10k to 1M), the voltage varies - Passive single-coil guitars produces weaker signals, while active-pickup metal guitars and electro-acoustic guitars can pump out voltages close to line level
Forgive my ignorance here, but what is in fact the difference between a VCA and a "master subgroup" that he describes near the end? Are they just the same thing, but one is pre and the other is post fader or something?
Main difference between a VCA (called DCA on a Digital Console) is that on a VCA when you move the fader it is the same as adjusting all the Faders individually in that VCA group where with a Control Group you are simply just adjusting the master volume that's being sent to that Control Group. My problem is that many digital consoles only have DCA's and not Control Groups so you don't have the option. Unless a Matrix can do the same job?
@jacomeintjes9709 I have yet to see a digital console that doesn't have groups options available. Which console have you used that doesn't have this feature?
@@conorm2524 The Yamaha TF series mixer does not have subgroups. The closest thing you could do on those is send your channels to an Aux Send, and then use that Aux as a subgroup, and feed all your different Aux sends into a Matrix for output to the amps. But if you use your Aux Sends as subgroups like that, then you lose the ability to use them for more important things like stage monitors.
@garrisonaw So you can still use them as sub groups. Cool. Still waiting on the other guy to explain how "many digital consoles only have DCA's and not Control Groups"...
Subgroups are buses (sub-mixes) in which audio signals are actually merged and then regulated by a single fader, so it not only provides a collective volume control for a selection of channels but also makes it possible to apply public processing to them - For example, applying a general EQ for all lavalier mics of same brand to calibrate their frequency response and to alleviate feedback problems, or applying a general compression for all mics recording sound from a drum set. However signal from a subgroup cannot be routed to auxes - It can only go to the main stereo bus or to matrices. This could cause a problem if aux-fed signals are used. VCA's are batch-operation remote controls which turn down the volume of their member channels proportionally under command from a single fader. Signals from member channels of a VCA are still separated and never premixed, so all aux-fed things like FX's or monitors would work fine. However VCA doesn't allow the signals to go through the same process like what a subgroup does.
So I have a question. Ive been doing sound on call at a bar, and I've done about three gigs. The system hasn't been tuned yet, and though I want to get in, and do that, for one, I'm not the main FOH guy there, and also, they don't have much gear, or mics/stands etc, so I'm already doing a lot to bring a lot of my own mics, compressors etc. Here's my question though. I totally understand gain staging etc, and once I go through sound check, and get all my levels, and I'm pretty good at getting the most out of the Pre's in the cheaper mixer they have, but anyway, the biggest struggle is, with bands that play loud, I definitely feel like I'd like the drums to cut more, mainly the kick/toms, and both are all individually mic'd) the issue is definitely always that, and vocals. I can basically have almost no guitar, and bass going on in the PA mix, and vocals are still drowned out, but not by a lot, I just always wish I had like literally 5-10 more DB's that I cant seem to get. Now, I noticed the PA's (just two) , and 2 floor subs, are set half way, and Iv'e never touched them, because they are basically taped over, I guess by the main engineer, I havent asked, but I know the bar owner mentioned random people have messed with them before, so maybe he taped them. At any rate, they are pretty good size speakers, and the subs hit pretty good, and they are cheaper, but not to horrible, like the old Yorkville Elite classics lol. Think I could just turn them up to maybe like 3 quarters, and that might help with the issue, because to give an idea, I'll pretty much have my main mix faders at unity, the Kick a bit higher than unity, the snare a bit lower, toms about unity, guitars, and bass usually always less, except when I need to boost for solos or some automation, but vocals are almost close to maxed on the faders, and just verging on feedback lol. I do a good job at avoiding feedback for sure, but just saying it's right on the edge, but I still need like 5-10db's ideally. Also the meters on the board are always close to maxed, so I thought maybe turning up the PA speakers would maybe help in some way? Like maybe I could have most stuff a bit lower on the board to get the volume I was getting with the PA's where they are now, and I could push the vocals abit more maybe?
I think you are exactly right, the PA needs to be louder to make more room for the vocals. However, if it is a separate it could be the amp is already at maximum volume for the specification of the speakers. Another consideration is whether the overall sound is too loud. Could the guitar and bass amps on stage be turned down a bit? Sometimes guitar players are more willing to do this is they have more guitar in their monitor. Also, consider that the vocals mics will also pick up the guitar amps. So it may be worth considering ways to reduce that.
@@me_fault Ya okay. So, they are just powered speakers, but pretty sure they aren't close to maxed, but maybe a precaution where it's taped, because there's no limiter either. That said, I totally always try to get the guitarists to turn down, and then I give them some guitar in the monitors like a normal person, and those sets go excellent, but 70% of the bands are just hard headed, and think they need to have their Marshall on 7-8, and I even tell them I can give them whatever guitars they need in their monitors(theres 4 monitors) Stage L/C/R/Drums, but they just don't listen, or if they do, 2 songs in it's back up. I try to tell them the PA's are in front of you so if you turn down, it may seem quieter than you're used to on stage, but think of the monitors as your own world, and as long as you guys can hear what you need, don't worry about volume, because I can crank it more through the PA's, but it will just be a bit quieter on stage than your used to, but just as loud out front. It's like bands arent used to proper sound, so they think monitors are just for vocals or something. With those bands, it's actually so loud that I can hardly use the guitar mics. It is a smaller venue, but still, on the bands that do keep it more chill, it always sounds so much better, and then drums/vocals aren't an issue. Im just going to crank the PA's to like 3 quarters next time, and just not say anything. Then ill just be a bit more conservative on my master fader to start. Also, It's hard sometimes to argue with guitarists, especially when it's like a veteran band thats so used to having those volumes, so I usually ask nice before, if it's to loud, I might ask once, but after that I let it go. If they wanna yell at me they cant hear there vox all night, and same with the crowd, I just say, go talk to the guitarist, and tell him to turn his Amp down hah.
I don't quite understand the last part, or I should say I don't see a scenario when using a VCA to turn an input down 20db is a problem for aux levels, If you do foh and monitors from one console, you most likely send your inputs to auxes in pre fader and if you are doing only monitors for example why not just use digital trim ?
Scovill is not talking about Aux Pre problems for "Monitors from FOH" scenario, but for AuxPost sends feeding the parralell compression processing (ie. Parallel Drum Bus) that will be disrupted once you lower the VCA on the drumkit (AuxPost sends to DrumSquash compressor will be lower, under the threshold on the comp). Hope that all is clarified now 👋
Lol I thought the same thing.. I am super new, so I appreciated it but yea, very basic.. Where do you get that basic information?? I don't have a college near by that offer's any of this stuff.. Idk where to start other than YT.. I am finding it difficult to find direct information tho.. Lots of sales stuff.
@@rhinoskin7550it doesn’t help that RUclips is FULL of confidently-wrong misinformation. You can’t go wrong with Scovill tho! He’s one of there greatest. When I was coming up in the early 2000s, the Yamaha Sound Reinforcement Handbook was a great manual for proper audio fundamentals, live and studio. But it might be somewhat dated now
I’d say most people know, they just need to be reminded… because they’re thinking too much about the tiny details and sometimes forget the main purpose of things.
@@rhinoskin7550my textbook for audio and live sound is the yamaha sound reinforcement handbook, 2nd edition. Try starting there. It’s a difficult subject to learn without the hands on practice of working with a sound reinforcement system but you can definitely learn the theory and practice via reading and vicariously through RUclips. At some point, though, you’ll need to find a venue or school or something that can give you hands on practice
Isn’t this the guy who used to travel around on behalf of avid, claiming you had to gain as much as possible to «preserve all the bits». Never lost respect for anyone that quickly in my life.
This is flawed teaching. Subgroups or sub-masters should not be used for the presented reasons. Also, too much PA should be dealt with using the matrix feature, so your matrices become control room volume.
Why not use subgroups to set the volume in the room? How do you do it? Any better way to do it? You know he sends the subgroups to a matrix for the PA?
I assure you Robert Scovill knows that he is talking about. He has forgotten more about audio than you will ever know. Not only does his artistic/mixing skills surpass most FOH engineers, I think you would be hard pressed to find a more "technically minded" person than Robert. Did you know that Robert is credited as the first person to use virtual sound checks and live multi-track workflows? (Something he started using back in the "analog days" while working with Tom Petty). You should really use Google to learn about the people you are criticizing before you post.....
@@BrianSimmonsNo seriously. Though he has had a succesful career. His understanding of the basics of audio, and especially digital audio is deeply flawed. Fortunately/unfortunately, there is enough headroom and leeway in modern gear to let him keep his job;-)
@@Seanalbertt I thought they were designed to operate with digital audio. Could you look in the back of yours and tell me what inputs do you have? Also could you point me to the part where the mixer "operates" with line level?
Having heard his work over the decades, I can assure that He does know what he is talking about. During presentations like this, it would be nice to insert the screen image that he is referring to, thus enabling a clearer explanation of the process. Also, If a question or answer is offered, it would also help to restate said comment for the remote listener. I have used the same approach and understanding of concept for over 10 years with digital and agree that this is valid on all counts.
You caught both issues that would hugely improve remote learning!
Almighty master of sound u need 1 mic for students.
News flash: he’s not the one that set up the video shoot.
Hahahah true
A lot of subtle wisdom packed in 20 min. Love the simple, practical thought process.
Robert is by far the best at explaining gain
Where can I see the full masterclass?
A few bookmarks for me:
Some great reasons to keep the bit depth high: 9:04
Hi Robert, i have met you twice in India.
But, your explanation of consoles bulit optimsed to line level signal is the best i have come across.
Three weeks back, i did my best to expalin to my team on the concept of dealing with a line level signal and mic level signal.
It also explains why many programmed drums and synrhs sounded as they due low bit rate..
¹😊you q⁰the
I gained alot of good info :)
I see what you did there! 😬
I had a very basic understanding of how important gain structure is. You explained it very well (bit rate) in great detail. Good video.
As a reference, the signal to noise on a 2" tape machine is in the area of 11-12 bit.
for all corp talking heads i set pres like in a studio, run mixes at 0dbvu minimum and use inline pads to make the room work on shitty powered speakers. i usually set at -20/-25 db to feed the boxes. this way all feeds are optimal. in a proper room with a line array, not needed. i'm talking shitty self powered JBL type boxes. Works a treat for me!
This is absolutely sound thinking. For those who disagree and use gain structure to reflect musical composition you are making a compromise. There’s nothing wrong the approach and in analogue this was normal practice. But it is a compromise. You sacrifice perfect gain structure for musical taste, or you sacrifice nominal fader position for musical taste. There’s no other way out… except subgroups. A VCA is just another fader compromise. So you need to know how your console manages processing in terms of latency. Everything we do in FOH is about compromise. Subgroups are the only way to mix without compromise in the digital world.
Need to ask:
If I have a kick that’s too loud, is Robert suggesting I route pre-amp -> fader (unity) -> kick sub group -> whole kit sub group -> master LR?
That would require going bus to bus. It would also mean that any time I have a channel that needs its level adjusting (probably most or all of them) I need to send it to a bus before the master LR. I don’t have enough busses to do that.
Forgive me, am I missing something or is he seriously suggesting that?
Excellent argument. Agree 100%
This situation he's describing with the kick that is too loud - is he saying that instead of turning the gain down on the channel (which would reduce the optimal signal quality coming into that channel), and instead of turning down the master LR fader (which is set at its optimal travel point for mix control, and for other instruments that might already be coming through at the right kind of level), and instead of pulling the fader down on the channel itself, you should send that bass drum to a subgroup after it's been processed, and just before the main outs, where that subgroup reduces the level of the signal?
If I've got a kick that's coming in way too loud compared to everything else, then if I keep its gain at the optimal level for processing (even though it sounds too loud) then surely I'm going to have to pull it down in any monitor mixes that I have as well?
As a practical approach to mixing, I've always set the channel faders at unity and the main mix fader at or slightly below unity, and then I've tried to get a rough mix where each input sounds roughly right. If the visual meters are telling me a signal is way too low or too high coming into the desk, and it still doesn't sound the right volume in the room, it's usually an issue with my amp levels. Thus, what "sounds" right in a balanced mix on the gain pots is my starting point for a balanced monitor and front-of-house mix. I'm not always trying to get things to be peaking at their optimal levels, but rather just using the meters to show me if there's a signal that is far too low or far too high. If I wanted the kick to be lower in the mix (e.g. for aesthetic reasons) I don't understand why it's a problem to pull down the fader. I want the faders to represent the actual mix - so that if something is at -10, then it's probably quite subtle in the mix, and if something's above unity, it's very prominent.
I don't know if I'm fundamentally misunderstanding his point, or if I've got it, or if my approach is entirely different to his? I do appreciate that a gain meter that is barely dancing or is constantly peaking is a bad thing, but I've felt that in between those extremes, what my ears tell me is right when all other faders and sends are at unity, that tends to be where I leave the gain knob alone.
So please be forgiving if you think I'm being obtuse - genuinely keen to learn.
I sort of agree with you. His point I guess was that if you have to pull the channel fader way down you are out of the realm of "hires". So every movement you have to make from then on out will be far less subtle then when it could have been at unity.
But for me (and I'm absolutely not questioning his skills or knowledge, I'm rather convinced that this is not the whole story that he told) there is missing quite.a lot information. First of all, gain structure. in my opinion is more than the little knob on the console. It has to do with the whole chain from signal source to speakers and all the variables that are in between. Furthermore, the point that digital is less forgiving within context of gain and signallevel than analog is a little bit understated. Because especially the headroom issue is not addressed but quite important in my opinion.
What he's saying is mix the drums, for example, into a bus or mix group, then adjust the group's level (fader) to the Master/Main mix. Likewise with other channel groups - vocals, instruments, etc.
Thanks
From the sound of it, you will need one subgroup for each “problem” input channel to optimise the sound - if you have the same band performing every week, I guess it is doable. But I usually have to set up shows within 2 hrs from scratch for different bands because venue hire is expensive. That’s just barely enough time to unload everything from the dock, connect and sound check, assume the talents arrive on time. So I just don’t think I can afford to do this level of optimisation. I am also confused on the concept of bit rate - I thought a DAC/ADC conversion bit rate is always constant regardless of the input voltage. The perceived drop in sound quality when the input is too low is due to signal to noise ratio. Happy for someone to correct me here.
The bitrate is the same but if your peaks are only reaching half the full scale range, then you’ll only use half of the available bits.
Look it like hooking up a 1080p device to a 4k screen
Very informative 🔥
Everything's good except that instead of lowering the levels using subgroups (if the PA is overpowered), I would lower the volume as far as possible in the chain in order to keep an optimal level up to the Master Fader. I would ideally dim down the power amps with the benefit of reducing the self noise of the PA, otherwise the PA processor, or as a last resort lower the post master output on the console.
Thanks for this... really grateful, Pls how do you avoid digital clipping when gaining
what do you do if your mixer doesn't have sub groups?
How does it go from -12 being 22 bit depth to -0 at 21 bit? Timestamp 9:06 , wouldn’t the 0db result in a bit depth of 24bits?
Each bit represents 6db of dynamic range. When he says "hanging at zero" he means the "0" (unity) position on the fader/trim meters which is actually -18dbfs ...which equals 3 bits. Hard to tell without seeing his slide/graphics. A signal hitting 0dbfs (the digital ceiling) would in fact use up all 24 bits like you're saying.
So just to make sure I understand correctly: ideally, assuming we gain staged our inputs correctly, every fader would be set to 0dB, and we would be mixing with strictly subgroups?
I guess I'm confused, how do you make subtle balance adjustments if everything is ideally kept at 0db? I can' imagine making a sub-group for each individual channel as well as subgroups for groups of instruments, like an all drum group.
not at 0db but near it. earlier in the talk he mentions -/+ 10db being okay
@@me_fault I must have missed that. This makes much more sense now, thank you!
@@RonMikeGabe He does not make subgroups for each channel only the instrument groups. As a rule keeping the individual faders between +10 and -10, he calls this high resolution mixing. He uses the subgroups to adjust for the level differences between screaming guitar amps and thundering drums and comparatively low level vocals.
@@MatthewBostater Thanks for the info!
i know this is like a year later...but is there a way were i can get those notes on the screen just because i want to keep it like in a pdf or somethig
Are there more videos from this masterclass?
please which brand and model of wireless headset microphone did you use as you were teaching?
Now to go drop 30k on a new PA
So, when is the signal at line level exactly... when the RMS at the input gain meter is properly set or when in addition to that the channel fader is also at unity?
You'd start with the faders at unity (initially) to ensure full resolution of the signals. Then, gain stage your signals to reach the "line level" of +4dBu (0dBVU on a voltage meter) by driving gain into the signal with the preamp. That's the "sweet spot." This level is roughly equivalent to 0dBVU on an analog voltage meter or -18dBFS on a meter within a digital environment like a DAW. Your signal's average should hover around this spot on your meters. Some meters--like those in some DAWs--show RMS. In that case, this is also where you could stage the signals, especially if you are working with any analog hardware or plugins. For live sound, you'll be looking to get weaker signals like a "mic line" boosted up to the +4dBu target.
@@JeremyMedicinaVery helpful comment. Thank you.
This is fantastic information - I was puzzling between a DCA and a Subgroup to make this work correctly- I am having to run my Subgroup master faders quite low on the console (around -20 db) - Is this normal? Or is it perhaps that there is to much power in the PA itself, and this needs to be adjusted down?
You can just pull your master fader further down to push the subgroup masters further up to gain more control over them. Alternatively you can turn down your PA amps to gain more control over your master fader too.
@@danscu5278 Brilliant, Thank you! I have considered this and will work on this in my workplace. Using subgroups now in my mixing has been a game changer! I wish I knew this years ago
Funny a episode of running sound and it's hard to hear the guy talking..pex
Where are the class notes 4 this Scovil? Or PDF
Excellent! Thanks!
GREAT content! Really pooooor video/audio, get that keynote and that audiencemic 😂
Obviously couldn’t get the RUclips video audio level right!! lol 😂
Did you go to Berklee, 1983?
you mean audio group is the sub group?
So if 0dB is 24-bit, what is it when the drummer flams the snare and it +6dB?
I believe he was referring to 0 dBFS being 24 bits but dBU at 0 is 21 bits. He kinda jumped between scales in the moment you’re referring to.
7:33 dang
what about the difference in voltage
in Line level , Mic level , and Instrument level?
Someone correct me if I'm wrong but...
Line level = ~1V (about 1V of signal)
Inst Level = (basically anywhere in between)
Mic level = mV (milliVolts of signal)
Line level = around 1 volts + low to medium impedance (from several ohms to about 1-2k)
Mic level = below 50mV + low impedance (most are less than 300 ohms)
Instrument level = high impedance(10k to 1M), the voltage varies - Passive single-coil guitars produces weaker signals, while active-pickup metal guitars and electro-acoustic guitars can pump out voltages close to line level
Forgive my ignorance here, but what is in fact the difference between a VCA and a "master subgroup" that he describes near the end? Are they just the same thing, but one is pre and the other is post fader or something?
Main difference between a VCA (called DCA on a Digital Console) is that on a VCA when you move the fader it is the same as adjusting all the Faders individually in that VCA group where with a Control Group you are simply just adjusting the master volume that's being sent to that Control Group.
My problem is that many digital consoles only have DCA's and not Control Groups so you don't have the option. Unless a Matrix can do the same job?
@jacomeintjes9709 I have yet to see a digital console that doesn't have groups options available. Which console have you used that doesn't have this feature?
@@conorm2524 The Yamaha TF series mixer does not have subgroups. The closest thing you could do on those is send your channels to an Aux Send, and then use that Aux as a subgroup, and feed all your different Aux sends into a Matrix for output to the amps. But if you use your Aux Sends as subgroups like that, then you lose the ability to use them for more important things like stage monitors.
@garrisonaw So you can still use them as sub groups. Cool.
Still waiting on the other guy to explain how "many digital consoles only have DCA's and not Control Groups"...
Subgroups are buses (sub-mixes) in which audio signals are actually merged and then regulated by a single fader, so it not only provides a collective volume control for a selection of channels but also makes it possible to apply public processing to them - For example, applying a general EQ for all lavalier mics of same brand to calibrate their frequency response and to alleviate feedback problems, or applying a general compression for all mics recording sound from a drum set. However signal from a subgroup cannot be routed to auxes - It can only go to the main stereo bus or to matrices. This could cause a problem if aux-fed signals are used.
VCA's are batch-operation remote controls which turn down the volume of their member channels proportionally under command from a single fader. Signals from member channels of a VCA are still separated and never premixed, so all aux-fed things like FX's or monitors would work fine. However VCA doesn't allow the signals to go through the same process like what a subgroup does.
So I have a question. Ive been doing sound on call at a bar, and I've done about three gigs. The system hasn't been tuned yet, and though I want to get in, and do that, for one, I'm not the main FOH guy there, and also, they don't have much gear, or mics/stands etc, so I'm already doing a lot to bring a lot of my own mics, compressors etc.
Here's my question though. I totally understand gain staging etc, and once I go through sound check, and get all my levels, and I'm pretty good at getting the most out of the Pre's in the cheaper mixer they have, but anyway, the biggest struggle is, with bands that play loud, I definitely feel like I'd like the drums to cut more, mainly the kick/toms, and both are all individually mic'd) the issue is definitely always that, and vocals. I can basically have almost no guitar, and bass going on in the PA mix, and vocals are still drowned out, but not by a lot, I just always wish I had like literally 5-10 more DB's that I cant seem to get.
Now, I noticed the PA's (just two) , and 2 floor subs, are set half way, and Iv'e never touched them, because they are basically taped over, I guess by the main engineer, I havent asked, but I know the bar owner mentioned random people have messed with them before, so maybe he taped them. At any rate, they are pretty good size speakers, and the subs hit pretty good, and they are cheaper, but not to horrible, like the old Yorkville Elite classics lol. Think I could just turn them up to maybe like 3 quarters, and that might help with the issue, because to give an idea, I'll pretty much have my main mix faders at unity, the Kick a bit higher than unity, the snare a bit lower, toms about unity, guitars, and bass usually always less, except when I need to boost for solos or some automation, but vocals are almost close to maxed on the faders, and just verging on feedback lol. I do a good job at avoiding feedback for sure, but just saying it's right on the edge, but I still need like 5-10db's ideally. Also the meters on the board are always close to maxed, so I thought maybe turning up the PA speakers would maybe help in some way? Like maybe I could have most stuff a bit lower on the board to get the volume I was getting with the PA's where they are now, and I could push the vocals abit more maybe?
I think you are exactly right, the PA needs to be louder to make more room for the vocals. However, if it is a separate it could be the amp is already at maximum volume for the specification of the speakers.
Another consideration is whether the overall sound is too loud. Could the guitar and bass amps on stage be turned down a bit? Sometimes guitar players are more willing to do this is they have more guitar in their monitor.
Also, consider that the vocals mics will also pick up the guitar amps. So it may be worth considering ways to reduce that.
@@me_fault Ya okay.
So, they are just powered speakers, but pretty sure they aren't close to maxed, but maybe a precaution where it's taped, because there's no limiter either.
That said, I totally always try to get the guitarists to turn down, and then I give them some guitar in the monitors like a normal person, and those sets go excellent, but 70% of the bands are just hard headed, and think they need to have their Marshall on 7-8, and I even tell them I can give them whatever guitars they need in their monitors(theres 4 monitors) Stage L/C/R/Drums, but they just don't listen, or if they do, 2 songs in it's back up. I try to tell them the PA's are in front of you so if you turn down, it may seem quieter than you're used to on stage, but think of the monitors as your own world, and as long as you guys can hear what you need, don't worry about volume, because I can crank it more through the PA's, but it will just be a bit quieter on stage than your used to, but just as loud out front. It's like bands arent used to proper sound, so they think monitors are just for vocals or something. With those bands, it's actually so loud that I can hardly use the guitar mics. It is a smaller venue, but still, on the bands that do keep it more chill, it always sounds so much better, and then drums/vocals aren't an issue.
Im just going to crank the PA's to like 3 quarters next time, and just not say anything. Then ill just be a bit more conservative on my master fader to start. Also, It's hard sometimes to argue with guitarists, especially when it's like a veteran band thats so used to having those volumes, so I usually ask nice before, if it's to loud, I might ask once, but after that I let it go. If they wanna yell at me they cant hear there vox all night, and same with the crowd, I just say, go talk to the guitarist, and tell him to turn his Amp down hah.
I don't quite understand the last part, or I should say I don't see a scenario when using a VCA to turn an input down 20db is a problem for aux levels, If you do foh and monitors from one console, you most likely send your inputs to auxes in pre fader and if you are doing only monitors for example why not just use digital trim ?
I think part of it would have to do with any post-fader sends that would be affected such as FX or subgroup processing.
@@christianmartinez1 Oh yeah I see but what was shown on the screen when he was saying that is confusing.
@@cocofocan what timestamp?
Scovill is not talking about Aux Pre problems for "Monitors from FOH" scenario, but for AuxPost sends feeding the parralell compression processing (ie. Parallel Drum Bus) that will be disrupted once you lower the VCA on the drumkit (AuxPost sends to DrumSquash compressor will be lower, under the threshold on the comp). Hope that all is clarified now 👋
@@thebuzge it is now, thanks !
i can not believe no one there could answer why they need a mic pre...
Everyone too nervous to answer.
I’m more concerned with the participants who don’t know what the sole purpose of a preamp is….. that’s like “college audio 101” not a masterclass.
Lol I thought the same thing.. I am super new, so I appreciated it but yea, very basic.. Where do you get that basic information?? I don't have a college near by that offer's any of this stuff.. Idk where to start other than YT.. I am finding it difficult to find direct information tho.. Lots of sales stuff.
@@rhinoskin7550it doesn’t help that RUclips is FULL of confidently-wrong misinformation. You can’t go wrong with Scovill tho! He’s one of there greatest.
When I was coming up in the early 2000s, the Yamaha Sound Reinforcement Handbook was a great manual for proper audio fundamentals, live and studio. But it might be somewhat dated now
I’d say most people know, they just need to be reminded… because they’re thinking too much about the tiny details and sometimes forget the main purpose of things.
@@rhinoskin7550my textbook for audio and live sound is the yamaha sound reinforcement handbook, 2nd edition. Try starting there. It’s a difficult subject to learn without the hands on practice of working with a sound reinforcement system but you can definitely learn the theory and practice via reading and vicariously through RUclips. At some point, though, you’ll need to find a venue or school or something that can give you hands on practice
@rhinoskin7550 Bobby Osinski's books are Belmont U's curriculum.
Too many questions. just tell them
Get your gain structure right and you look like a wizard. And you can almost put your feet up and let the gig run itself. Don't though...
Isn’t this the guy who used to travel around on behalf of avid, claiming you had to gain as much as possible to «preserve all the bits». Never lost respect for anyone that quickly in my life.
So what's your explanation to why he's wrong?
Yeah that’s for digital recording using audio interfaces. Live mixing is different
This is flawed teaching. Subgroups or sub-masters should not be used for the presented reasons. Also, too much PA should be dealt with using the matrix feature, so your matrices become control room volume.
Why not use subgroups to set the volume in the room? How do you do it? Any better way to do it? You know he sends the subgroups to a matrix for the PA?
And how do you feed your matrices?
@@Rythym4god Main stereo buss feeds all Matrixes.
this guy has no idea about audio. "optimized for Line level" lol. has no idea about bit depth haha
Even if you consider he won't be working at 32 bit fp?
I assure you Robert Scovill knows that he is talking about. He has forgotten more about audio than you will ever know. Not only does his artistic/mixing skills surpass most FOH engineers, I think you would be hard pressed to find a more "technically minded" person than Robert. Did you know that Robert is credited as the first person to use virtual sound checks and live multi-track workflows? (Something he started using back in the "analog days" while working with Tom Petty). You should really use Google to learn about the people you are criticizing before you post.....
@@BrianSimmonsNo seriously. Though he has had a succesful career. His understanding of the basics of audio, and especially digital audio is deeply flawed. Fortunately/unfortunately, there is enough headroom and leeway in modern gear to let him keep his job;-)
Ummm…. Who gonna tell him? This is also basics of audio. Regardless of bit depth… all mixers are designed to operate at line level……
@@Seanalbertt I thought they were designed to operate with digital audio. Could you look in the back of yours and tell me what inputs do you have? Also could you point me to the part where the mixer "operates" with line level?