Proper Mixing Procedure

djred2000

Member
I was wondering what the generally accepted practice is for mixing microphones on an analog board specifically for muting. I occasionally run sound for musicals at a local community theater and when the actors enter and exit the stage I just pot their mics all the way down or back up. When setting levels I do my best to set the level as close to the 0 mark on the fader as possible. Other people just press the mute button on the channel when actors enter and exit and don't mess with the faders much. I personally find it hard to keep track of what's going on if all the faders are up and just muted. The way I do it, only the faders that are up are live.
 
From what I've seen, all professional musical theater mixing is done with line-by-line fading. It's a pain in the butt to do, but the pros use VCAs and snapshots extensively to group ensemble mics per scene, which brings the fader count down significantly. The professional actors also know to follow the script religiously and not ad-lib or skip lines.

What I do for short-run community theater plays, with no VCA/snapshotting, is try to co-locate channels for actors that will be together a lot, and I use up to eight fingers (all but thumbs) for mixing. And if there is any doubt as to whether or not a particular actor will say something at the wrong time, I may leave a fader or two at half volume instead of pulling it down completely in between lines. There is a rule that attenuation below -6dB of a competing audio source will tend to not be heard by the audience ... so if I pull the fader down 20dB I still feel okay.

I have never muted an input channel as a mixing method. I have only muted channels when they are not being used at all for the performance.

-- John
 
A quick fade-up also gives you that split second to abort if something has gone bad on the channel. Better a missed line than a blast of noise.
Mute buttons spook me because I am at an age where I can remember some old analog boards where toggling the mute itself made noise.

As for the second part, I assume you mean that you adjust the pad so that you can set your channel fader so that the "normal" level will be with the fader at 0db ? Pretty normal as long as your mic preamp is within it's limits.
 
I'm a religious muter. That said, I often times will fade out a channel, hit the mute, and then return the fader to it's previous position to avoid cutting anything off. Same goes for unmuting, but in reverse. I'm doing a lot of channels at once, I use mute groups or VCAs to do the same concept, and I use a lot of snapshots and automation when I can.

That said, it's all about what works for you. The only thing you might have to worry about is cross talk. I recently discovered a few channels on my A&H GL2400 could be considered notorious for this when hot signals are played through them and the fader is down.

Again, however, there is no proper or best way, just ways that are better in certain circumstances. It just so happens to be that those circumstances are largely dependent on who is behind the console.
 
On my venue's LS9, (digital console) i'm a "Muter" on a little touring Mackie VLX analogue console i'm a "Fader" digital consoles have very clean mutes from my experiences especially if you are comfortably with in your Preamp range. On the Mackie, which is old and not in great condition i don't trust the mute buttons to be clean and/or not stick. With the LS9 if time permits i do use a lot of snapshots an scenes and just cue through a show light a lighting person would.

But to answer your question.
New consoles tend to work much better with the channel mute philosophy then older consoles, ("analogue" consoles")
And generally on analogue consoles or consoles that are not in tjhe "Pro" quality level you are probably safer and better off gaining to a given fader level, and using faders as your on/off.

Something to think about to that an "on/off/Mute" button will kill all signal from proceeding any further, so sometimes you might want to to keep number channels on, but not only mix them to monitors and not mails. In this instance Muting them will give you some grief. :)
 
Thanks for the replies. We are not using any AUX sends on the channels so we don't have to worry about the mute button taking out the pre-fade output. The actors are also not switching mics during the show. As far as setting the level, I bring the fader to the 0db position with the trim all the way down. While the actor is speaking, I raise the trim until the level is found and I can fine tune with the fader as close to the 0db mark as possible. This makes it easy for me to remember the levels when I bring them up each time the actor enters the stage.
 
That's sounds a bit backwards. How do you remember where each gain trim was previously? Normally gain trim is set at soundcheck, and rarely touched after that unless for minor adjustments. Bringing levels in and out of the mix should be done by faders and or mute buttons. Then when bringing the fader up to 0 you'll know you're at the exact same level as before because you did not touch the gain trim.
 
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That's sounds a bit backwards. How do you remember where each gain trim was previously? Normally gain trim is set at soundcheck, and rarely touched after that unless for minor adjustments. Bringing levels in and out of the mix should be done by faders and or mute buttons. Then when bringing the fader up to 0 you'll know you're at the exact same level as before because you did not touch the gain trim.

He said exactly that...

Also, many engineers will set gain with the fader down and dial it up until it meters at or near zero, then bring the channel into the house until it sounds good... then throw the mute. They care more about how hot the pre-amp burns then where the fader ends up. Midas is especially known for how their pre-amps sound when over driven like this. Recording engineers do the same thing. I can't tell you how many engineers, especially the jazz guys, who walk in, spin every gain to 2 o'clock, then start pushing faders up. If they find everything is sitting low on the console they will halve their output faders. This is all for music, not theatre, but you will see both ways in the real world. When mixing on VCA's many engineers will do the same thing. It is more about gain structure vs fader position. Of course if the gain structure is set correctly past the console you don't have to deal with this as much. Then again, the gain structure out PA is set for is to output as much sound as possible so when doing quieter events we tend to drop our output faders to compensate.
 
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He said exactly that...

I got confused when he said he pot's things up and down.... thought he was refering to the gain pots, but I see now he's refering to the fader pots.

when the actors enter and exit the stage I just pot their mics all the way down or back up.

He's doing it the right way, I do that too to avoid the hashness of muting buttons.

I don't know why you'd want to overdrive your gain trims and keep your faders down below unity though, that sounds more like a 1960's Beatles recording effect rather than proper gain structure. Today there are better ways to add distortion.
 
It depends on the show for me. If I know it well I'll use mute buttons. If I'm not as comfortable with entrances/exits or have too many things to do at once then I'll use faders. And I've never tried to mix line by line.
 
He said exactly that...

Also, many engineers will set gain with the fader down and dial it up until it meters at or near zero, then bring the channel into the house until it sounds good... then throw the mute. They care more about how hot the pre-amp burns then where the fader ends up. Midas is especially known for how their pre-amps sound when over driven like this. Recording engineers do the same thing. I can't tell you how many engineers, especially the jazz guys, who walk in, spin every gain to 2 o'clock, then start pushing faders up. If they find everything is sitting low on the console they will halve their output faders. This is all for music, not theatre, but you will see both ways in the real world. When mixing on VCA's many engineers will do the same thing. It is more about gain structure vs fader position. Of course if the gain structure is set correctly past the console you don't have to deal with this as much. Then again, the gain structure out PA is set for is to output as much sound as possible so when doing quieter events we tend to drop our output faders to compensate.

For musical theater I set my system gain as so: Input Faders at Unity, vox VCA's at -10dB, Output Faders at 0dB.

On initial system setup, I set outputs with music and pink and I'll gain the Amplifiers up to where they need to be, or if they are passive I'll drop the gain in the system processor keeping the console as clean as possible and doing the EQ and Gain work on the back end. On the input side I set mic pack gains all the same to start and I change gains as they meter on the receiver. I set the input fader to 0dB, throw the vocal inputs DCA to -10dB and I have the actor talk on stage, from there I gain up the channel until it is at an appropriate level. I don't use meters for anything other than to see clipping. I like to throw -10dB on the DCA for vocals, throw to -5dB for singing and unity when I really need to push. If I need to throw hotter than +3 I start looking at adding a compressor into the channel but I like to start with no to very little compression to maintain the natural dynamic range of the musical.
 
(This statement does not apply to the people on this thread, who have a good understanding of sound.)
I am always amazed at the confusion between mic preamp gain and channel volume. Several times, I have been on "service calls" where problems were resolved by changing gain at various places. The preamp gain (or "pad") is there to optimize the gain of the mic preamp. Too much gain and you will have clipping/distortion. Too little gain and your system may appear noisy as you are having to crank the gain somewhere else. In a gain optimized system, the largest amount of gain should occur in the mic preamp. If you achieving that gain elsewhere, you are also adding gain to any noise produce by any of the electronics or cables that come before the gain jump. Working backwards, power amps should have just enough gain to achieve clipping before the signal clips from the prior device. The same applies to each piece as you work your way back. Although digital equipment is dead quiet compared to the old analog junk I used to deal with, this practice still has great advantages, especially if there are long cable runs. In those cases, any noise picked up on a cable run between processing, crossovers, etc, will be minimized. Needles to say, the best way to achieve this occurs if each idem in the chain has a meter. In other words, if the board meter is up at 0db, then the signal meters on any equipment up to and including the power amps should also read 0, each piece then having about the same headroom above that point. This also tends to minimize noise from accidental hot-patches.
To this day I find myself frustrated by people who believe that if a power amp is only turned up "half way" that it will only produce half it's rated output! (last argument, 48 hours ago.) All gain controls throughout the system should be considered "calibration points" with the exception of the board faders and masters, which provide the user input.
/rant
 
(This statement does not apply to the people on this thread, who have a good understanding of sound.)
I am always amazed at the confusion between mic preamp gain and channel volume. Several times, I have been on "service calls" where problems were resolved by changing gain at various places. The preamp gain (or "pad") is there to optimize the gain of the mic preamp. Too much gain and you will have clipping/distortion. Too little gain and your system may appear noisy as you are having to crank the gain somewhere else. In a gain optimized system, the largest amount of gain should occur in the mic preamp. If you achieving that gain elsewhere, you are also adding gain to any noise produce by any of the electronics or cables that come before the gain jump. Working backwards, power amps should have just enough gain to achieve clipping before the signal clips from the prior device. The same applies to each piece as you work your way back. Although digital equipment is dead quiet compared to the old analog junk I used to deal with, this practice still has great advantages, especially if there are long cable runs. In those cases, any noise picked up on a cable run between processing, crossovers, etc, will be minimized. Needles to say, the best way to achieve this occurs if each idem in the chain has a meter. In other words, if the board meter is up at 0db, then the signal meters on any equipment up to and including the power amps should also read 0, each piece then having about the same headroom above that point. This also tends to minimize noise from accidental hot-patches.
To this day I find myself frustrated by people who believe that if a power amp is only turned up "half way" that it will only produce half it's rated output! (last argument, 48 hours ago.) All gain controls throughout the system should be considered "calibration points" with the exception of the board faders and masters, which provide the user input.
/rant

Let me see if I understand correctly. Lets say I am setting the level for an iPod connected to a stereo input on an analog board. I would have the channel fader all the way down then solo the channel and raise the trim until the meter was showing signal at the 0db mark. I would then set the main output faders to the 0db mark and the output knobs on the power amp would be set to 0db as well. I would then raise the channel fader to the proper volume.
 
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Let me see if I understand correctly. Lets say I am setting the level for an iPod connected to a stereo input on an analog board. I would have the channel fader all the way down then solo the channel and raise the trim until the meter was showing signal at the 0db mark. I would then set the main output faders to the 0db mark and the output knobs on the power amp would be set to 0db as well. I would then raise the channel fader to the proper volume.
Well, the stereo line input actually bypasses the mic preamp, so the trim/pad/input gain will have no affect on the signal.
It's not about hitting 0 on the meter, that is only when you first set up or recalibrate the system so that all your hardware is happy. (Usually done with a noise generator as compared to program material.)
If you had the chance, you would want to pre-test the iPod and set it's output level so that you have a comfortable level coming out of the system when you fader is near the 0 mark in it's travel, as well as check for any overload distortion. What you don't want is to find out you need your channel fader almost all the way up or all the way down to achieve that level you want at the time of the show.

As far as system calibration goes, systems vary a lot as far as their structure, but to keep the concept real simple, lets say we just have a board and a power amp. (no crossovers etc.) You want the output level meters on the board to accurately represent what the system is doing. You want to know that when your amp is about to clip that your board is telling you this is about to happen. There are exceptions to this. For instance, in theater, the working level of your power amp system may be so low that using this setup would render the board meters useless as they would never show any level. In that case, you would decrease the gain on your power amps so that the board meters are at least providing you useful data.

The key element to my original post was that it is critical that we don't accidentally have a weak link in the chain that goes into clipping during normal use and at the same time provide enough front end gain that we don't amplify the background noise of everything else in the system.
 
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Well, the stereo line input actually bypasses the mic preamp, so the trim/pad/input gain will have no affect on the signal.
It's not about hitting 0 on the meter, that is only when you first set up or recalibrate the system so that all your hardware is happy. (Usually done with a noise generator as compared to program material.)
If you had the chance, you would want to pre-test the iPod and set it's output level so that you have a comfortable level coming out of the system when you fader is near the 0 mark in it's travel, as well as check for any overload distortion. What you don't want is to find out you need your channel fader almost all the way up or all the way down to achieve that level you want at the time of the show.

As far as system calibration goes, systems vary a lot as far as their structure, but to keep the concept real simple, lets say we just have a board and a power amp. (no crossovers etc.) You want the output level meters on the board to accurately represent what the system is doing. You want to know that when your amp is about to clip that your board is telling you this is about to happen. There are exceptions to this. For instance, in theater, the working level of your power amp system may be so low that using this setup would render the board meters useless as they would never show any level. In that case, you would decrease the gain on your power amps so that the board meters are at least providing you useful data.

The key element to my original post was that it is critical that we don't accidentally have a weak link in the chain that goes into clipping during normal use and at the same time provide enough front end gain that we don't amplify the background noise of everything else in the system.

I get it now. We do have our power amp only turned up slightly to about the 9 o'clock position so that we can see the signal on the mixer.
 
While we are talking gain structure, here is an interesting way to do it easily: http://digital.livesoundint.com/publication/?i=223280&p=20

Basically, you have a calibrated device that puts out a known level. You put that in front of your favorite mic. Gain it up until you are at or near 0 on your channel stripe. Bring the channel fader and main faders to 0. You can then go through your entire system and get a 0 reading or correct volume everywhere. At that point your entire system is lined up correctly. In my work I usually find systems have way more power then they need and the consoles can not work properly.
 
Well, the stereo line input actually bypasses the mic preamp, so the trim/pad/input gain will have no affect on the signal.
It's not about hitting 0 on the meter, that is only when you first set up or recalibrate the system so that all your hardware is happy. (Usually done with a noise generator as compared to program material.)
If you had the chance, you would want to pre-test the iPod and set it's output level so that you have a comfortable level coming out of the system when you fader is near the 0 mark in it's travel, as well as check for any overload distortion. What you don't want is to find out you need your channel fader almost all the way up or all the way down to achieve that level you want at the time of the show.

As far as system calibration goes, systems vary a lot as far as their structure, but to keep the concept real simple, lets say we just have a board and a power amp. (no crossovers etc.) You want the output level meters on the board to accurately represent what the system is doing. You want to know that when your amp is about to clip that your board is telling you this is about to happen. There are exceptions to this. For instance, in theater, the working level of your power amp system may be so low that using this setup would render the board meters useless as they would never show any level. In that case, you would decrease the gain on your power amps so that the board meters are at least providing you useful data.

The key element to my original post was that it is critical that we don't accidentally have a weak link in the chain that goes into clipping during normal use and at the same time provide enough front end gain that we don't amplify the background noise of everything else in the system.

JD, I'm nitpicking, but there are two things that have irked me while reading your otherwise educational rant.

First, a "pad" is a very particularly functioning tool that should not be confused with a preamp or trim. Pads are level reduction devices that have no level adjustment functionality and are simply in or out. These are highly useful on very hot (loud) sources which would otherwise overdrive the pre-amp or or to keep an input below the pre-amp's optimal point of sonic performance.

Second, the comment about the line input bypassing the preamp is not necessarily true on many boards, and more importantly, is still affected by the same pot. For example, on the A&H GL2400-32 I use (like frickin everywhere), when you want to use a line source on a given channel, the "pad" button opens the switch to the line input and reduced the signal level into the board by 20dB. This effectively puts it at the proper voltage to be gained to unity at the preamp (minus 4dB from pro gear and plus 10dB from most consumer gear), which is actually marked around the pot.

Back to the thread as a whole, gain staging is very important, and should be maintained until the absolute last level control in the system you have access to. If it's the amps, it's easy. In my case, I usually don't have access to the amp/system processors, so I either just pull my mains fader(s) down or route them to matrices so I can see the level as it should be seen on my meters relative to my target SPL.
 

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