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  • Writer's pictureMatthew Losey

Go with the Audio Signal Flow: Part 2 | How-to #5

Updated: May 2, 2023



Continued...

Once you get the audio into the digital snake it has to travel over an ethernet cable to get there. Go ahead and read the manual, many times there is an option for redundant cabling. This is very helpful because, unlike audio cables, ethernet cables have a tendency to spontaneously die. If even one of the eight wires in an ethernet cable stop working, the whole signal is lost. Best case, it results in intermittent disconnections. This can mean clicking and popping in the audio channels. At its worst, it can mean losing audio entirely until it’s fixed.

For this reason, I usually suggest running several ethernet cables at the same time. Ethernet spools are relatively cheap especially if you are running the cables by yourself. Then you can buy a termination kit on the internet for about $50 which includes female RJ45 jacks, a punch-down tool, and an ethernet cable tester. There are plenty of instructional videos for you to use on the interwebs. You can do it!

Processing

Once you get your audio over the ethernet line, it then goes into the board. Now the processing begins, which in itself has several stages, parts, and outputs. We will only go over the basics of each without getting too far into the weeds of preferential audio mixing. I’ll do my best to give you enough to be dangerous but also understand how mistakes can cause more dramatic issues down the line. At the same time, good decisions can help your mix be forgiving to the unpredictability of audio in a live environment.

Assuming you are able to patch the right input on the digital snake to your console input, you are ready to start mixing. Let’s pretend that we are working through a vocal channel. They are using a wired Shure Beta58a. It’s a dynamic mic with a uniform supercardioid pattern. This essentially means it’s good at rejecting audio that’s not right in front of it. That also means low risk of feedback and the ability to gain it up and down without concern for problems.

Speaking of gain, it is the first item on your channel that you can change. The best practice to find appropriate gain for a microphone is to gain it all the way down and press the pad button if there is one. The pad is a -20db button that slightly extends the dB range of the gain. Have the vocalist start singing and quickly bring up the gain knob until they are peaking at a -6db level on the meter and call it a day. This should take no more than 10 seconds. Do not make slow changes while doing this part of the soundcheck because it’s the easiest part. Don’t bring up the gain too much or you will clip. But if you are too low, you may have a hard time finding enough volume and have to compensate with the channel fader which ultimately gives you less control. Once you set up the gain, leave it alone unless it’s clipping because it will affect everyone’s in-ears as well as change your gate, compressors, and effect settings down the line.

The next thing that’s usually up is the gate and compressor. They can be difficult to use but I’ll just briefly talk about what they aim to do. The gate is primarily used to let the audience hear only the audio that the audience is meant to hear. It can take out noise and unwanted audio that the engineer doesn’t want to be highlighted. Using the gate appropriately will allow the mix as a whole to sound more cohesive and clear. But if used inappropriately, it can make the mix sound choppy and cut in and out seemingly at random.

After the gate cleans up unwanted sound (you can of course choose to bypass this) you usually have the compressor. The compressor hopes to achieve something similar to what the gate does. The compressor ignores certain sounds while reining in louder more unpredictable audio to make it more predictable. The popularly held belief is that a compressor makes audio less dynamic and that is sometimes the case. Take a vocalist for example. If they don’t have a compressor, their quiet singing may sound dramatically quieter than when they really get into it and start to get loud. When you amplify that through the speakers, it can be too dynamic and become distracting or unintelligible. So, we turn on the compressor and feed it the appropriate settings for the application.

The compressor also has the capacity to add dynamics to an otherwise typically non-dynamic instrument. If you have a bass guitar that is playing very predictably, maybe too predictably, you can use a compressor to emphasize the first few milliseconds of note hit and add some life in there. It can make drum hits sound more punctual and make acoustic guitars more snappy. It all depends on the skill of the audio engineer controlling them. But occasionally, by design adding compression brings the overall volume of the channel down. This means that you will have to make up volume in either the fader or in the “make-up gain knob” on the compressor. Don’t just guess, turn the compressor on and off until the channel sounds the same in volume. A quick reference is to turn the makeup gain up 1 decibel for every 2 decibels of gain reduction that you observe. Practically speaking, use your ears. If it sounds the same, it sounds the same.

We finally get to the most fun part of the chain. The equalization a.k.a EQ. EQ is comprised of a few specific tools in order to shape the full spectrum sound of the sound. First, you have the high pass filter which lets you pass all audio over a specified band. This prevents ‘plosives and booming sounds from being too noticeable. Next, you have the low-pass filter which is not on all boards. It does the opposite of the high-pass filter and lets all audio below the specified band pass through. This keeps a lot of unwanted high-frequency noise or ringing out of the mix. Lastly, you have the parametric EQ bands. With this powerful tool, you can completely overhaul a channel. If you aren’t in a live scenario, you can quickly boost a frequency, sweep side to side to hear the worst parts, and back them off, one by one. This takes all the ugly away and gives you space to work with. Then you can start doing more creative things to the EQ after your ducks are in a row cleaning up the audio. Occasionally, you get a makeup gain in the EQ because EQing has a tendency to reduce the overall gain. This can bring your volume back to acceptable levels but this is a rare option among live console boards.

Before the well know fader control, you usually have the panning function. This takes your input signal and balances it between two or sometimes even more outputs. Most popularly, it is just a left and right speaker. When mixing stereo inputs, it’s very important to “hard pan” which simply means panning it 100% to the left and 100% to the right for the corresponding left and right channels. This prevents a lot of muddiness that the instrument is typically already accounting for in its stereo functions. The stereo inputs we are referring to are backing tracks, background music channels, pianos, keyboards, and sometimes even guitars. If you go 50% and 50% on your stereo inputs you tend to get a lot of boominess in the mids which is where our ears are most sensitive to gross sounds.

Following these rules on stereo inputs is a quick way to get to the end of a good mix, but mono inputs are completely subject to musical taste. Examples of mono inputs are guitars, vocals, bass guitar, drum mics etc. The main consideration here is whether or not your audience can hear all of the speakers. If everyone in the room can hear both speaker channels (left and right) you can mix like you would a professional recording which is usually not the case. If not, you have to be aware that some people may never hear a rhythm guitar that is hard-panned to one side. The same goes for drums. If you like to dramatically pan drum toms, I’m right there with you. But I hate hearing a couple of toms well and feeling like the rest disappear into the mix. Because of that, I usually gently pan them side to side while live mixing so I can still hear some space in the mix. Panning typically allows space for similar inputs to be heard simultaneously and determine general positioning on stage but since most audience members don’t ever hear a full stereo image, this just ends up causing additional problems you didn’t intend. The most popular configuration for live setups is stereo but the most effective mix tends to be a hybrid of mono mixing with aspects of stereo when necessary.

Processing Peripherals

I’m not sure if any audio effects and aux sends are considered part of the processing but I’m creating a subgroup (audio joke) called processing peripherals. When it comes to auxiliary sends, including effects and monitoring sends, there is this thing called pre-fader and post-fader. There are actually a couple more in between on more expensive boards but I’ll leave that alone for this post.

When it comes to pre-fader vs post-fader it’s all about control. The less you have to do, the better. The more you can do with a single fader, the better off your control will be as long as you have everything set up correctly. Here is the rule of thumb: Pre-fader is for monitor mixes and post-fader is for effects. But as with any rule, rules are meant to be broken.

Practically speaking, if you are mixing a few vocal channels and you wish to have a delay or reverb added, you don’t want to have to constantly manually change the sends. If you had mistakenly set channel sends of the vocal to pre-fader, there is the possibility of the input being turned all the way down, while the FOH feed still hears the affected signal of whatever the mic hears. In FOH it could echo a fart, it could echo a mic drop, a mic smack, general noise, or some other unwanted amplification.

At the same time, if that vocal channel needs to be turned up, the effect volume for that channel needs to change as well. If the effect volume doesn’t change with it, it will by contrast sound like there is no reverb or delay because the difference in volume is too great. So to solve this problem, set the FX sends to post-fader. This allows the auxiliary send to be changed or adjusted in accordance with the fader position i.e. less work for you, the engineer.

The opposite is pre-fader. There are times when you do not want the position of the main channel fader to determine the volume of the auxiliary send. If I need to hear my instrument or vocal channel at the same volume no matter what in my monitor mix, I want my mix to be completely separate. So “pre”-fader position, is sending the audio “before” consideration of the main mix fader position.

Here’s the rule-breaking part. If you have an input that tends left on all the time, you want that one to be post-fader in a monitor mix. The most popular example is the pastor or announcement microphone in a church. The pastor might leave their wireless pack on all the time which means it’s constantly receiving an input signal. It could contain, the pastor singing, the pastor talking to their spouse, going to the bathroom etc. This signal wouldn’t be heard in FOH because the main FOH fader is down but it would be heard in all the monitor mixes. Not ideal. To avoid this, set infrequently used channels like computers, pastors, and announcement mics to post-fader. Again, this makes the engineer’s life easier because they don’t have to adjust several monitor mixes when they already have to mix FOH. This way it will happen simultaneously so long as long as the fader is set at a position above (-)∞.

When comes to effects, you are going to deal with something called parallel processing. Some soundboards actually have parallel processing knobs on every channel or each effect. That would allow you to potentially have a compressor effect set to 50%. Sometimes called the wet/dry mix, it allows you to reduce the total effect of the effect by however much you choose. With aux feeds that are fed back into the same mix, it is called parallel because their signal path is going side-by-side down the line. You are also changing the wet/dry mix by changing the volume of the feed returning into the mix as well as the individual send of the channel to the effect send. When setting up the effect channels there is sometimes a wet/dry mix knob as well. When feeding the aux returns back into the main FOH channels, always set it to 100% wet. This will keep your channels from doubling in volume and only affect the audible FX.

When attempting to make your engineering life as easy as possible, set all your individual effects sends to unity for any channels you want to effect. You can then use one fader to affect the most common FX returns (reveb and delay) and decide when and when not to apply the effect. This gives you a quick and fast way to turn off the effect when a person is speaking and back on when they get back into singing. All this can be done with one fader.

Final Mixdown

The final set of the mix-down is the main sends to the speakers. This could be in mono, stereo, LRC (Left right center), LRM (Left right mono), or dual stereo mixing for live audio. This would take all the information that you have been processing for all channels, combine it to 1-4 channels, then output it to where ever you have patched. This would then go to an amplifier and finally a speaker or a powered speaker where both are integrated together.

Some amplifiers accept digital audio inputs which I always prefer. This means that you can keep your noise floor down as low as possible and keep your AD and DA conversions as minimal as possible as we discussed earlier. ON more complicated systems, you may have a few output from your audio board and it will then go into what is called a DA or distribution amplifier. A distribution amplifier/processing unit or whatever your integrator calls it intelligently sends different parts of the audio signal to different places. It also has the capacity to balance speakers that are designed for different frequency bands as well as time delay speakers that are further away from each other. The final and probably most important job for a DA is to protect a speaker and its amps. This prevents an audio signal from being processed through the speakers that might be harmful or damaging and cause it to “blow” or becomes distorted.

Conclusion

This hopefully means you have a great-sounding mix that is evenly distributed across an auditorium event space. If this was too long and you decided not to read it here’s the conclusion.

  • Keep the analog to digital conversions to a minimum

  • Pre-fader for monitor mixes

  • Post-fader for effects

  • Panning is incredibly important but don’t go overboard because not everyone can hear both speakers

  • Plan your mix so you can do less work and make it sound better

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