Build Your Career As A Sound Engineer
Are you sick of reading the users manual every time you need to set up an SD5?
Me, too!
Good news. Last night I got Aleš real drunk and got him to spill the beans about setting up a session from scratch. There was a lot of gibberish in there, but I was able to edit it down into something comprehensible.
Let me know if this helps.
UPDATE: This article appears in The Ultimate Guide to Creative Mixing on the Digico SD5.
In this video I describe how I used snapshots to automate my mix on the Digico SD5 as head of sound for the Out Of This Word tour for the Ringling Bros. Circus. In it, I walk you step-by-step through how I implemented snapshot creation, global scope, recall scope, update selected, auto update, cross fades, add to group, relative groups, update group, recall times, and midi list for Qlab control.
One of the most important things you need to know how to do on the SD5 related to snapshots is how to update a snapshot in the middle of the show. There are many times, especially while building and refining a new show, when you’ll need to make a small edit. You need to continue mixing the show, though, so you don’t have time to create a new snapshot and type in a name for it or scroll back up to the correct shapshot and take care that you are updating the right elements.
The SD5 makes this very easy, but it’s also dangerous! Here’s my process:
The key here is not to move your finger away until you have disengaged auto-update. Why? Because every change you make while it is engaged will get written into the snapshot. So if you forget, which I did only once, and keeping firing snapshots and mixing the show, you will record all of those changes.
This happened to me more than once: You change a microphone position or another way affect the gain staging and all of a sudden 127 snapshots are wrong. Normally, you would have to go through and adjust these one-by-one or find a workaround, but the Update Group feature solves this pretty easily.
The most time-consuming process here is adding the snapshots to the group. You might think: Why don’t I just leave all of my snapshots in a group so that I can update them at any time? Well, you could do that, but then you put yourself in a dangerous position if you accidentally hit the Update Group button. The undo button does not fix this!
In Out Of This World, the first three acts were executed with the same timing every night. There were some fader movements that needed to happen at specific times so I automating them using Recall Time. As long as I fired the first snapshot at the right time, the mix would be in sync with the show. And if for some reason that didn’t work out, I would simply skip the timed snapshots and mix it manually.
The main benefit here was that I could convert the energy from my pre-show jitters into useful work, like double checking plugin settings and refreshing my memory of upcoming cues and changes. Another less import benefit was that I could fire the snapshot to take us into intermission and immediately run to the bathroom since I had the next set to fire by Recall Time. This is important when you need to beat an arena full of 20,000 people who are also rushing for the bathroom.
Welcome to sound design live. I’m Nathan Lively, and I just completed the fourth presentation of a live webinar training on sound system tuning. The webinars went well, but there just wasn’t enough time to answer all of the question that the 535 attendees sent in. But I don’t want to leave those people out in the cold so I have gathered all of the questions and I am going to answer all of here on the podcast.
I have organized them into sections, so today we are going to continue to part 2 and talk about how to prepare for sound system tuning so that we can be more efficient.
You see what I did there, Dennis? I answered your question with Joe’s question.
So what’s triage? Triage is the assignment of degrees of urgency to wounds or illnesses to decide the order of treatment of a large number of patients or casualties. And it’s a great word to describe our work because in the rooms that we work in, the sound system, our patient, is suffering from many wounds. Some of them life threatening, some of them not. Live sound is a war zone and we have to figure out which of our patient’s wounds to heal first, to keep it alive.
And of course, we have a limited amount of time, because the show will start whether we have healed all of the wound or not. So the way you save time in sound system tuning is by picking the biggest problems and attacking those first. And I don’t know what those are going to be in your room, but they most often lie with speaker position and aim.
My best time saving tip is to plan your microphone positions ahead of time. So if you can get the venue and system layout the night before, go through and mark all of your microphone positions with names OnAxA, XAB, VTopB, etc., and include what you want to do in each position, set level, delay, EQ, etc. If I don’t do this preparation step, it is almost guaranteed that I will take the right measurement in the wrong position, or vice versa, and have to go back and do it again, or worse, I won’t realize it until the show has started.
Another idea, if you have the resources, is to use multiple microphones. That will speed things up. Personally, I’m saving my money to get a multiple wireless microphone measurement rig. I got turned on to the Line6 XD-V75 and TX-3 microphone from Mel labs by one of my students, which seem like a good combo.
One other idea for you, Michael, if you can do some of the verification steps like polarity check ahead of time, at the shop, that will save you time.
This is really hard to answer because, as I mentioned in question seven, I can’t do triage on a hypothetical patient without knowing anything about it. But I do want to give you some place to start.
And of course, these are under ideal conditions where the system is already operational and I can just start optimizing. But half the time some technical problem will pop up and have to be taken care of because, if the system doesn’t play, none of this stuff matters.
So Lou, I have a special deal for you. Just put a crisp $100 bill in an envelope and send it to Nathan Lively, Minneapolis, MN, and I guarantee that you will get consistent results from this day forward.
Just kidding Lou. Of course, when you get really good at using all of the tools you have available to you, your ears, eyes, audio analyzer, etc., you can better predict the most important factors that will influence consistency. I wish I could tell you that if you match tonight’s frequency response with last night’s frequency response that it will sound exactly the same, but that would be a lie. There are other factors like room reflections that may actually require you to use a different target frequency response to achieve consistent results.
The only way I know to guarantee consistent results is to give everyone headphones and make sure all of your instruments go direct.
But there is hope. I think that if you not only train yourself to correlate what you see on the audio analyzer with what you hear in the room, but are also diligent about taking notes every day on what you learn and then reviewing those notes on a regular basis, your results will continue to improve and become more and more consistent.
Carol, I have to admit to you that I still get nervous before every gig. There’s a little voice in my head that say, “You’re an imposter. You have no idea what you’re doing.” So the downside is that there is some stress related to each event i work on, but the upside is that that red flag motivates me to prepare better. So I find out as much as I can about the equipment, room, and production I’ll be working on ahead of time. I actually have a needs assessment checklist that I go through to make sure I haven’t forgotten anything.
And of course, this might not work out. It is fairly common that I either won’t be able to find out anything, or all of the information I got will turn out to be wrong or changed at the last minute. But I still feel more confident, knowing I did everything I could to prepare myself. And I know this all may sound kind of obvious, but I can’t tell you the number of times that I have arrived on site and been the only one who has bothered to look up the coverage angle and power rating for the speakers we are using.
So Seun, there’s nothing special about speakers being in a line or not, but I’m guessing that you are asking about working with constant beam width versus proportional beam width speakers, the modern line array element. And the main difference that comes to mind is that with proportional beam width speakers in a coupled array, you create the shape you want with the splay angles between elements, but that really only affects the high end. The low end doesn’t really care if you adjust the splay angles. It simply plays out out of the mid point of the array. Tapering the amplitude won’t help you like it will with constant beam width speakers. The only way to get around that is with all-pass filters to help you slightly steer the low end up, which is beyond me. I’m still trying to master the basics.
So Kyle, you’ve got it right…but exactly backwards. The first parameters you should look at are polarity, level, and time. So if we have to sacrifice someone, I say we sacrifice EQ.
It doesn’t matter how good a monitor engineer you are. If you have severe halitosis you will not be asked back.
Things that I learned/re-learned from watching Mixing Monitors from FOH on Soulsound.
I’ve been a member of Soulsound for 2 years and 7 months. Their videos are great quality and watching them always makes me want to go back and redo all of mine.
I learned a lot from watching this masterclass with Justin Grealy. He is very thorough and it really motivated me to up my game and deliver better service to the clients I work with. Instead of minimal effort to get the job done, prepare for every possibility and make every comfortable.
The only thing missing from this class is practical work with a band on stage, which would make a great part 2 (hint, hint).
If you have an analog console, Y-split at least the vocal channels.
I’ve done this technique a couple of times. My first question is always: What amI going to do differently to the monitor split compared to the FOH split?
Answer: EQ, dynamics, effects
One of the biggest problems I have in small concert venues is that the mix in the room can be ruined by the LF from loud stage monitors. At first you think, “Who cares, crank it up till their ears bleed if they want it.” But remember, every speaker is omni directional at some point, not to mention reflections from the back wall. So a loud stage monitor is going to give you lots of LF into the front rows as well as the stage. One way to help mitigate that is to raise the HPF on the vocal channel to something insanely high, like, as high as you can get away with. I’m talking 350Hz! And you can get away with that by splitting the vocal channels.
When it comes to dynamics, it’s likely that you don’t want as much compression on your vocals in the monitor mix as you do in the FOH mix. In most cases, on small, loud stages, you want almost no compression so that the vocalist can hear their raw performance to improve their microphone technique. In the same way that an actor learns to “find their light,” a singer must learn to find the pocket for a consistent performance.
The more you compress it, the more you affect the way the musician “actually” plans the instrument. It’s really easy to make people over play.
A gate set correctly can also be a huge advantage in the FOH mix, but tricky in the monitor mix. Vocalists will complain at the first hint of a problem, and we could get into the psychology of that, but what’s important here is that you don’t want to give them any reason to complain. It’s like when you’re on a Skype call and it mutes everyone else while you’re talking. You’d think silence would be nice, but it’s unsettling.
You will rarely meet an analog desk these days.
This is an interesting difference between the US and UK markets. In the US, analog mixing consoles still hold strong, especially in small rooms and hotel AV. They are cheaper and easier to train people on. Which makes me wonder, why have analog desks disappeared from the UK?
Grealy suggests duplicating all inputs for a separate monitor mix for more control, which is ideal, but I never do that. There’s just not enough time to do both jobs comprehensively. I can imagine a situation where maybe you have some of the controls linked between layers or quickly copy and paste your settings, but I can’t imagine anyone having the patience for me to track down technical problems, setup the FOH input channels, and then setup the monitors channels for three different bands. I usually barely have enough time to do one job.
So time management is the main consideration, but hearing is another. Most venues where I’m mixing monitors from FOH are not going to include a stage monitor for me at FOH to listen to my changes. That means I’ll need to listen from across the room, on headphones, or do a lot of guessing. I might be making more problems than solutions.
In the video, Justin inserts graphic EQs in his monitor sends. I never do this anymore since most desks also have a parametric EQ on every output.
There are only two good reasons to use a graphic EQ: 1) ear training and 2) the war zone that is monitor world. But aren’t we in monitor land? Yes, but the reason we might want them in a war zone is the simple speed of grabbing a fader and moving it. That doesn’t work on a digital mixing board.
Check out the graphic EQ setting in the image below. It looks pretty minor. A little shaping. Maybe removing some feedback.
Then look at the shape that it’s actually creating in the measurement below. Yikes!
One thing I’ve never understood is the doubling up of stage monitors for the main vocalist. I mean, I’ve asked people about this. When I asked Michelle Sabolchick Pettinato about why Eddie Vedder has 8 stage monitors she said, “SPL and coverage,” which I still don’t quite understand. In all cases, why not just turn one speaker up louder or get a louder speaker?
Maybe someday I’ll work on a show like that and finally understand it.
The way Grealy uses polarity and phase interchangeably is confusing. Grealy says he is going to “check that all of the monitors are in-phase with each other.” He then proceeds to verify polarity. It’s important to remember that phase shift is frequency dependent and caused by delay in the signal. Polarity is frequency independent and has no time offset. So if we are checking phase we are going to need a delay line or physical offset and if we are verifying polarity we just need a polarity inversion switch or cable. So when Grealy says that on tour he found an out of phase cable, he basically discovered a time machine. That being said, I do appreciate his quick and dirty method of listening to stage monitors in pairs. Nice!
Grealy’s explanation of pink vs white noise is a little confusing. Although white noise is statistically equal level over the frequency range, we perceive it as spectrally tilted toward the high frequencies because our ears respond on a logarithmic basis and are more sensitive in the high end. Pink noise is filtered white noise, attenuated at 3 dB/octave, to sound more even to our ears.
Some people cheat and use an analyzer. I use these [pulling on ear], because I find that they work rather better.
You didn’t think I was going to let Grealy bash audio analyzers and get away with it did you?
Grealy goes on to say that an audio analyzer can mislead you, but fails to demonstrate how, which seems like fear mongering. There is enough misunderstanding about audio analyzers as it is without Grealy turning it into a pissing contest. An audio analyzer is a tool to supplement the two audio analyzers you already have attached to your head, not to replace it.
Grealy says that he is a big fan of decoupling the stage monitor from the stage to get it closer to the performer’s head. Somewhere along the line I learned that stage monitors are tuned for half-space loading, but I’ve never measured the effects of the change. My question is: Does removing the stage monitor from the stage require EQ compensation in the LF? My guess is yes, but I’m not sure how much.
I would test this by comparing measurements of a stage monitor on the stage and 2 ft up on a stand.
I like Grealy’s suggestion to keep two monitors as close as possible if you want LF coupling, but I wonder where he came up with 30cm? Two LF drivers spaced 30cm apart would give you coupling up to 755 Hz since the wavelength of 755 Hz is 45cm long and 2/3 of that is 30cm. But the fundamental range of a 4-string bass goes from about 40Hz to 400Hz. So if all you need to do is support the fundamental range, then you can get away with a wider displacement, up to 58cm. Maybe Grealy is counting the overtones, but that would go way past 755 Hz.
I would test this by pulling stage monitors apart until the bass player complained.
Are you a person that listens to business books at 1.3x speed? I am. Grealy is very well spoken, but there are enough pauses and “uhs” that I installed the Vimeo repeat & speed extension for Chrome so I wouldn’t get distracted.
In this episode of the Sound Design Live podcast, I want to share with you an interview that I did with Nate Schneider at AV Shop Talk. We talk about the technical challenges I faced while touring with the Ringling Brothers Circus, how I got my first jobs in the seven different cities I’ve lived in, and the advice I would give to 23 year-old me.
Show notes:
