Equal Gain vs. Unity Fader Mixing Methodologies

[BCAP]

I had read a number of online articles on this topic recently and wondered if anyone had strong opinions one way or another for theatre FOH mixing - your preference?? And, are you using a digital board or analog board??

 

[jkowtko]

Equal gain? I've never heard of that one before. I assume you are referring to the input chanel gain/trim. I thought standard practice was to adjust the input channel gain based on the incoming signal strength to normalize the signal level across the channels. I didn't think there was a difference in adjusting input gain between digital or analog.

 

[Aaron Becker]

I remember being taught the unity fader method back when I first learned how to mix. I don't tend to use it, but I also don't like faders down in the -50/-40 range, because often times (even on digital boards) you find your range is compressed down there, a 1/4" move changes it 6db where up near unity a 1/4" move changes it 1.

 

[sdauditorium]

I've always done similar to Aaron. I use an LS9 (digital), and typically ensure that, in our case, all wireless transmitters/receivers have the same settings (sensitivity/AF out) first. As a disclaimer, all systems are equivalent Sennheisers with Countryman E6s to have some sort of standard signal leaving the systems. On the board, I'll tweak the HA per input depending on the strength of the performer so that the sound coming from the house system is normalized during regular dialogue. I try to keep "unity" for us at -5 which is the start of the range on an LS9 gives the most sensitive adjustments. That way, it makes it a bit easier to give a good approximate balance during large ensemble portions. Typically, we use around 24-27 mics so having a standard level makes things infinitely easier on the fly.

 

[Aaron Becker]

I do the same with wireless racks - the output of the receivers should be the same. I tend to tweak the packs themselves for each actor, which helps equalize the levels of the faders and gain settings. Sound FX tends to have the biggest varying of levels, but I tend to try and adjust most of that in the "recording" itself - so that again, the audio level coming in is nice and strong.

 

[TimMc]

Digital or analog makes no difference these days - what I commonly see is channel input gain (trim, level, whatever the regional nomenclature is) set too high.

Allow me to shout this:

IT IS NOT NECESSARY OR TYPICALLY DESIRABLE FOR RED LIGHTS TO FLASH *ANYWHERE* ON A CONSOLE, ESPECIALLY AT THE INPUT CHANNEL. IT IS NOT NECESSARY TO USE THE LAST *ONE* IN THE DIGITAL INVENTORY. DO NOT TRIM INPUT GAINS TO -0- DBFS.

 

[themuzicman]

All Faders at Unity, Gain or Trim to taste. This always sets you up for best resolution as you mix -- especially if you're line mixing and know you're pushing 2dB too hot on a DCA -- you can blindly grab the input fader and adjust just right to get your mixing marks back on track.

If your input meters across the board are too low but signals are coming in hot I'll go in and start trimming console output levels so I can gain up and get functional meters, I'll then make it up at the Speaker processors or aux sends.

I do the same with wireless racks - the output of the receivers should be the same. I tend to tweak the packs themselves for each actor, which helps equalize the levels of the faders and gain settings.

Yep, this is the right practice -- keep the output of your receivers wide open and you adjust the transmitter until your sound source can't clip the pack.

Sound FX tends to have the biggest varying of levels, but I tend to try and adjust most of that in the "recording" itself - so that again, the audio level coming in is nice and strong.

My SFX inputs into the console never change, they are line level inputs or digital inputs so my gain is all the way down and the fader is at unity. I'll heavily trim outputs in QLab's preference so my level settings on Cue's hit consistent places and are around 0 in the program as well.

 

[TimMc]

^^^ Yeah, what he said.

So much of our workflow and preferences come from experience that sometimes they're difficult to explain because we just do them automatically.

Back in the day of 8 bit digital audio it was pretty important to use as many ones as possible without running out... the digital noise floor was above the noise of the supporting analog circuitry. Twelve bits was getting very close; at 16 bits w/oversampling the digital noise floor is well below that of the analog chips.

That said, old ways die hard and it's worse if the operator is still thinking "dBVU" in a "DBFS" world. Another back in the day technique was to PFL/solo an input and adjust the input trim until the meter read -0- dB. Do that on current digital consoles and wonder why it sounds bad and the outputs seem really, really hot... (there's a story that goes with this but it's TL;DR)

The advice regarding wireless receivers is concise and absolutely spot on.

 

[techieman33]

I agree with the others. Though in high school we weren't allowed to touch the gain. Our teacher had some weird hangup on adjusting the gain. Didn't matter if it was a mic or a cd player, we couldn't touch it. And he would often come check to make sure we hadn't changed it.

 

[BCAP]

I've always done similar to Aaron. I use an LS9 (digital), and typically ensure that, in our case, all wireless transmitters/receivers have the same settings (sensitivity/AF out) first. As a disclaimer, all systems are equivalent Sennheisers with Countryman E6s to have some sort of standard signal leaving the systems. On the board, I'll tweak the HA per input depending on the strength of the performer so that the sound coming from the house system is normalized during regular dialogue. I try to keep "unity" for us at -5 which is the start of the range on an LS9 gives the most sensitive adjustments. That way, it makes it a bit easier to give a good approximate balance during large ensemble portions. Typically, we use around 24-27 mics so having a standard level makes things infinitely easier on the fly.

I do much the same procedure as you and Aaron. There are some situations I work in where I merge different groups of microphones and transmitter makes/models and then it's not as easy.

Agree with TimMc, no 0DBFS on the input gains. I just don't want to see that signal so low it's peaking at -30, that's all, especially if I'm sending that signal somewhere else - for recording or broadcast.

 

[TimMc]

I do much the same procedure as you and Aaron. There are some situations I work in where I merge different groups of microphones and transmitter makes/models and then it's not as easy.

Agree with TimMc, no 0DBFS on the input gains. I just don't want to see that signal so low it's peaking at -30, that's all, especially if I'm sending that signal somewhere else - for recording or broadcast.

Considering that on most digital mixers -18DBFS or thereabouts is the old *0VU* , peaking around that number should give a reasonably "analog" gain staging. It's a starting point, not a destination.