First time Sound OP

JackMVHS

Member
Okay, over the several years of theatre experience at school and community theatre I have learned the in's and out's of several lighting systems and their equipment, but haven't done a whole lot with sound. I am now one of two sound op's for the community theatre's production of Grease.

I want to know what methods YOU use to actually mix the sound. How do you maintain constant volumes across the board and make everything sound good together. I know some of the basics, but I want to hear about how you do it.

Also, do you do your live mix's by listening to the house system, a speaker in the booth, or headphones? If you use headphones, do you recommend a specific model?

Please correct me if I am confusing different terms or concepts above, I am not too skilled in the area of sound. To our school it has always been a finicky subject with too many problems, but I am hoping to fix that. Thank you!
 
I've been running sound for 10+ years myself, mostly church services, some experience with bands/musicals/theater. Where is your mix position located? Is it in the house, booth, etc.? It works best for me to listen to what I am mixing for. If I'm mixing for the house, I get the best results if I'm listening directly to the house. If I'm mixing for recording I'll use headphones or booth speakers, depending on ambient noise.
 
I'm assuming both booths having open windows to the house? What I generally try to do get an idea for the difference between the house and the booth. If you know the house is louder than the booth, you can mix accordingly. Also, if you have any rehearsals, that would be the time to set your mix and then go out into the house and listen, then go back and make adjustments as needed, and so on. The best thing is just to have a good idea of the differences between house sound a booth sound. You have to mix with those differences in mind.
 
Yup, that makes sense, and that's what I have been trying to do. If anyone has anymore tips about mixing feel free to post, it is most appreciated!

The community theatre has a basic analog board, but my school has a complex digital board, a Soundcraft 328xd. Almost all the features are unused. Does anyone know anything about this board? I know it has an FX processor, snapshots, etc. How do I use all this to my benifit? Fianally, can you use sub groups or mixes on this board? Thank you!
 
While specific issues or questions can be effectively addressed in a forum, the very general subject of 'how to mix' and how to use your particular console are almost always much more effectively addressed with a "hands on" approach. Perhaps you can get a qualified local Consultant or Contractor or operator, maybe from a nearby performance venue or church, to provide some training at your theatre. You can also typically learn a lot from reading the manual for your mixer, which can be found at Soundcraft - [Downloads] if you don't already have it, as well as from resources such as "The Soundcraft Guide to Mixing" also found on Soundcraft's web site, Soundcraft - [Support].
 
I remember learning how to mix, and slowly learning all the in's and out's of the board.

I don't think anybody here (or anywhere really) can sit you down and tell you when how and where to use every function of the board.

I find it easier to just learn as you go. If all you know how to do at first is "I push this fader up, mic gets louder", that's ok.

As you get more experienced, you'll learn what other outboard gear does, other dials on the board, etc etc.

Eventually, you'll be able to claim the "...yes I do know what every dial and button does, and yes IT TOOK A LONG TIME TO LEARN..." line.

Good luck!
 
Yeah, being a lighting guy myself, I found a few of these pointers useful when I first started to mix audio.
  • Don't throw faders up like you would on a lighting console... First lesson I learned.
  • Highs, Mids & Lows (The knobs in the middle) should be set to 12 o'clock if you don't know how to use them.
  • Lows are Bass. Bass is cool.
  • Aux Sends are you friend, they are the audio version of submasters, and submasters are awesome for busking.
  • Mute it if you aren't using it.
  • Feedback happens, it's not always your fault.
  • Standing in front of a speaker with a mic is bad, anyone who touches mics, including actors, should know this.
  • Laptops could be the cause of that buzz.
  • Dimmed power running next to audio lead can also cause that buzz.
  • If it isn't loud enough with the fader up to the top, slowly crank up the gain.
And that's about all I know of audio.
Hope some of these ideas helped. I can tell you how to set up Aux sends if you want.
Nick
 
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Start your mix with nothing, and add only what is needed. This will give you the cleanest mix and keep the volume under control.

When there's too little of something/someone, quite often the problem is actually that there is too much of something/someone else.

When the tone of something is not right, don't automatically boost EQ to fill in what's missing. The first action is to cut EQ to reduce the area that there's too much of. (With a typical vocal mic used up close, there will always be too much upper bass/low mid, usually centered around 200Hz.)

TP Audio's Mixing Primer
 
IMO,

Before you begin to even attempt to mix live music and vocals, you need to have some idea what the instruments are SUPPOSED to sound like, preferably in an acoustic environment. Without sound reinforcement.

Vocals, on the other hand are something you should have already have a grip on, beginning with hearing your mother cooing at you as you breast fed.

Musical instruments are more nuanced and DO require an ear that is "tuned".

It ain't about numbers on the sliders, presets or pre-programmed effects being used by the numbers or some formulae.

It is music. If you have any musical tendencies, it will just come to you. Feel it.

If you ain't got it, you won't be the only one to figure it out, real quick.
 
Aux Sends arn't like sub-masters at all... Aux sends send only parts of the feed out a seperate line. For instance... a moniter mix. They dont' want drums in the moniter, so you set up an aux with everything except the drums, and then run the "Aux Send" on the back of the board to the moniter. The "Groups" are like submasters. (Typically the faders to the immediate left of the L-R Master). Channels are assigned to a group by the buttons right next to the channel fader (and sometime the pan nob), and then the sub-master fader controls the overall volume of those channels. Just like lights... (I do some lighting myself, but I really only use a crappy little Elation 16-ch Stage Setter. no sub masters ;)

I agree with CenterSpot in that once you figure out that the faders make it louder and softer, it's all about what you hear and feel. Alot of sound ops are musicians, and it definantly helps for a good mix.

One other hint I can offer is during sound checks, set all the faders to ±0dB, then set the gains so each channel is at a comfortable volume on its own. That way, you don't have faders all over the place, and they're somewhere around 0.

(Read this article - it replaces all 4 professional classes I've attended)
GAIN WITHOUT PAIN
Good luck!
 
It ain't about numbers on the sliders, presets or pre-programmed effects being used by the numbers or some formulae.

It is music. If you have any musical tendencies, it will just come to you. Feel it.
Unless you understand the technical aspects of how a mixer works and well beyond that to how the rest of the system and the environment potentially affect what you hear, you may know what you want but can struggle to get it. I have seen very good musicians fight to even get a decent sound when mixing, watching one struggle with feedback throughout a show that anyone who understood the basic physics behind what was happening could have addressed in a couple of seconds. They obviously knew what they wanted and that something was wrong, but were simply unable to apply the tools available to resolve these with the result of a terribly unmusical, and physically painful, mix. I believe that good mixing is not about being just a good artist or just a good scientist, but rather about also being able to work well in the areas where the art and science involved interact and, most importantly, being a good listener to not just the sound from the system but also to the input of others.
 
Thanks for the advice guys! I think I have learned most of the basics now through research, talking with a few sound experts, and most importantly experience. I have figured out how to use groups and auxiliary sends correctly now on our digital board. I had to look up the manual, but figured it out in the end.

I have 2 more questions, and I'm not sure if I should start a new thread for them or not... I will list them below, if you think a new thread is necessary, we can start a new thread. Thanks again!

Questions:

1. Our system has a constant loud static sound. I have ruled out the sound board. We have 3 main amps, 1 for the L&R subs, 1 for the L&R "Horns", and 1 for the center cluster. We have 2 more amps that power the stage monitors. When the L&R horn amp is shut off most of the static disappears. So it could be the amp, the speaker, connections, or wires. Anyone have an idea on how to fix this?

2. I am (hopefully) attaching a picture of our stage, you can see the 3 sets of speakers hanging from the ceiling. As you can see the speakers are BEHIND the apron causing LOTS of feedback issues. Anyone have ideas on how to get around this??

 
"One other hint I can offer is during sound checks, set all the faders to ±0dB, then set the gains so each channel is at a comfortable volume on its own. That way, you don't have faders all over the place, and they're somewhere around 0."

This has been debated, and the one glaring problem with this approach is that not only is your gain staging incorrect and reducing your s/n ratio, it also screws up the gain being sent to aux sends.

In fact, I'll go one further. Rather than using the trim pot to make all the faders look pretty at 0db, I'd rather have my input gain trimmed to 0db, and then my fader positions will show me the ralative loudness of all the sources. I can then make loudness adjustments at the source.

The other disadvantage of using trim to line-up faders is that your channel level indicators (peak light, 3 segment vu, ect) are now useless for monitoring input conditions.

No matter how you slice and dice it, no audio engineer (and I mean degreed EE with a specialty in audio) would configure a console in the way you describe.

Here's an analogy:

When you record a source, don't you try to get your record levels up near 0db. Why not just record it at a lower level, and then crank it up during playback with the volume knob? The reason is signal/noise ratio. We always try to get the signal at the appropriate level as early in the chain as possible. In the case of live audio, this occurs at the input gain/trim knob of the console. The fader levels then depict the relative volume of all the sources as they exist on stage. This also helps when established appropriate monitor levels. For instance, if I've adjusted all my inputs so I'm getting 0db thru the rest of the channel strip (peak light just barely hitting), and I see that my guitar mic fader is way down while getting a good mix to the house, I can instantly determine that the guitar is fricking loud on stage, and most likely needs little in the way of monitor send.

The other issue is if I am sending either channel inserts thru outboard comps, or via aux. sends to outboard effects processing. In the case of channel insert compressors, without the input trim being set correctly, my threshold value on the comp is now meaningless. The same holds true for outboard effects processing.

Bottom line:
Each stage in the signal path is designed to be operated at an ideal signal level. Hence the reason that most all controls associated with gain are calibrated and marked as db. While all have leeway in regards to make-up gain, when operated outside their ideal level, artifacts such as noise, clipping, and dithering distortion are unavoidable.
 
When you say you are hearing a constant loud static sound, is that hiss (noise like wind in trees), or is it crackling? If it's hiss, probably your power amps are turned up too high. Turn down the power amps, get your mixer levels to peak around 0, then gradually turn up the power amps until the sound is loud enough. That will result in better gain staging. Don't just turn up the power amps to maximum.

Re speaker location, can the speakers be re-hung closer to the audience, behind the mics?

Bruce Bartlett
Bartlett Microphones in Elkhart, Indiana - home page

Questions:

1. Our system has a constant loud static sound. I have ruled out the sound board. We have 3 main amps, 1 for the L&R subs, 1 for the L&R "Horns", and 1 for the center cluster. We have 2 more amps that power the stage monitors. When the L&R horn amp is shut off most of the static disappears. So it could be the amp, the speaker, connections, or wires. Anyone have an idea on how to fix this?

2. I am (hopefully) attaching a picture of our stage, you can see the 3 sets of speakers hanging from the ceiling. As you can see the speakers are BEHIND the apron causing LOTS of feedback issues. Anyone have ideas on how to get around this??

 
"Also, do you do your live mix's by listening to the house system, a speaker in the booth, or headphones? If you use headphones, do you recommend a specific model?"

I place two condensor mics in the house in a position I think will approximate the audience's listening position. I then route those inputs to two channels and not assign them to any output, then hit the solo button and monitor on headphones. That way, my headphones are giving me a pretty good idea of what the audience hears. I started doing this when I worked in control rooms which were isolated from the house sound, and have adopted this for all my theatre work, whether there is a window to open or not.

This also gives me quick access to solo individual mics and tweak compression settings, eq, etc.
 
This has been debated, and the one glaring problem with this approach is that not only is your gain staging incorrect and reducing your s/n ratio, it also screws up the gain being sent to aux sends.

In fact, I'll go one further. Rather than using the trim pot to make all the faders look pretty at 0db, I'd rather have my input gain trimmed to 0db, and then my fader positions will show me the ralative loudness of all the sources. I can then make loudness adjustments at the source.

The other disadvantage of using trim to line-up faders is that your channel level indicators (peak light, 3 segment vu, ect) are now useless for monitoring input conditions.

No matter how you slice and dice it, no audio engineer (and I mean degreed EE with a specialty in audio) would configure a console in the way you describe.
Could you please explain the comment "not only is your gain staging incorrect and reducing your s/n ratio, it also screws up the gain being sent to aux sends."? Consider the following example.

Let's say you have a mic signal hitting the preamp with a level of -40dBu, the noise floor is -100dBu and with the trim set for 0 you have the fader at -10 to get the desired level. Using the 0 signal level trim method you would thus add 40dB of gain at the preamp to bring the signal levels out of the preamp to 0dBu which also raises the noise level to -60dBu, then attenuate those levels with the fader by 10dB so that the signals post fader are -10dBu and -70dBu respectively. With the fader at 0 approach you add 10dB less gain at the preamp but then apply no attenuation, thus the signal and noise levels are -10dBu and -70dBu out of the preamp and after the fader. So both approaches seem to result in a -10dBu signal level and 60dB of S/N after the fader, however the second approach with the fader at 0 seems to result in the same 60dB S/N (0/-60 versus -10/-70) but 10dB more headroom (a -10dBu signal level versus 0dBu signal level) between the preamp and fader. At higher input signal levels the trim at 0 approach may indeed provide greater S/N, but that seems to not be relevant to mics while as long as having the preamp output at 0dBu results in the fader being below 0 it seems that will always result in having less headroom between the preamp output and fader. So why is a method that provides greater S/N only for higher line level signals where the S/N ratio is typically higher to start with and less headroom through part of the signal chain for most signals inherently better?

As far as screwing up the gain to the aux sends, I believe that the additional headroom between the preamp and channel fader can actually help with not screwing it up as you are typically combining multiple signals onto a single bus, meaning that if you start with all of the preamp outputs at 0dBu then you usually have a higher total level, and thus less headroom, than expected on any pre-fade send buses. And on many consoles there is no way of monitoring those levels except perhaps to look at the output level and factor in the master send level setting. Having a bit more headroom on an aux bus does not seem to be a bad or even undesired thing.

So I'm not clear how setting the trim to provide a 0dBu level actually provides better gain structure or better S/N or how it is problematic for aux sends. Maybe you can explain it.

A less technical aspect is that if having the preamp output at 0 results in the faders being quite low then you are into a non-linear part of the fader's operation where small movements make large differences, making it more difficult to have fine control, as well as having less attenuation available with the fader all the way down. And I find that having a consistent starting point for the faders really helps when you have to move from one mixer to another, such as a BE using different house systems, as trying to mix the same thing as always but with significantly different fader settings can throw you off. So can having to remember where faders were set to bring them back up after you have brought them all the way down.

I also do not understand why the indicators would be rendered "useless" by having the trims set lower. They are simply monitoring level; the peak light still lights when the maximum input level is reached, an XdBu level signal is still reflected as an XdBu level signal, etc. Nothing has changed the indicators' functionality or use.

I am not advocating simply lining up all the faders at 0 but rather setting them at levels that make sense for operation. If you have an input that you know needs 20dB more level for a solo, then start with the fader 20dB down from the nominal starting point. However, don't get so tied into setting all the preamps for 0 that you find yourself not having enough useful operating range on a fader or that you are clipping an aux bus. The reality is that you may get the best result with some combination of or intermediate approach between the two extremes.
 
Could you please explain the comment "not only is your gain staging incorrect and reducing your s/n ratio, it also screws up the gain being sent to aux sends."? Consider the following example.

Let's say you have a mic signal hitting the preamp with a level of -40dBu, the noise floor is -100dBu and with the trim set for 0 you have the fader at -10 to get the desired level. Using the 0 signal level trim method you would thus add 40dB of gain at the preamp to bring the signal levels out of the preamp to 0dBu which also raises the noise level to -60dBu, then attenuate those levels with the fader by 10dB so that the signals post fader are -10dBu and -70dBu respectively. With the fader at 0 approach you add 10dB less gain at the preamp but then apply no attenuation, thus the signal and noise levels are -10dBu and -70dBu out of the preamp and after the fader. So both approaches seem to result in a -10dBu signal level and 60dB of S/N after the fader, however the second approach with the fader at 0 seems to result in the same 60dB S/N (0/-60 versus -10/-70) but 10dB more headroom (a -10dBu signal level versus 0dBu signal level) between the preamp and fader. At higher input signal levels the trim at 0 approach may indeed provide greater S/N, but that seems to not be relevant to mics while as long as having the preamp output at 0dBu results in the fader being below 0 it seems that will always result in having less headroom between the preamp output and fader. So why is a method that provides greater S/N only for higher line level signals where the S/N ratio is typically higher to start with and less headroom through part of the signal chain for most signals inherently better?

As far as screwing up the gain to the aux sends, I believe that the additional headroom between the preamp and channel fader can actually help with not screwing it up as you are typically combining multiple signals onto a single bus, meaning that if you start with all of the preamp outputs at 0dBu then you usually have a higher total level, and thus less headroom, than expected on any pre-fade send buses. And on many consoles there is no way of monitoring those levels except perhaps to look at the output level and factor in the master send level setting. Having a bit more headroom on an aux bus does not seem to be a bad or even undesired thing.

So I'm not clear how setting the trim to provide a 0dBu level actually provides better gain structure or better S/N or how it is problematic for aux sends. Maybe you can explain it.

A less technical aspect is that if having the preamp output at 0 results in the faders being quite low then you are into a non-linear part of the fader's operation where small movements make large differences, making it more difficult to have fine control, as well as having less attenuation available with the fader all the way down. And I find that having a consistent starting point for the faders really helps when you have to move from one mixer to another, such as a BE using different house systems, as trying to mix the same thing as always but with significantly different fader settings can throw you off. So can having to remember where faders were set to bring them back up after you have brought them all the way down.

I also do not understand why the indicators would be rendered "useless" by having the trims set lower. They are simply monitoring level; the peak light still lights when the maximum input level is reached, an XdBu level signal is still reflected as an XdBu level signal, etc. Nothing has changed the indicators' functionality or use.

I am not advocating simply lining up all the faders at 0 but rather setting them at levels that make sense for operation. If you have an input that you know needs 20dB more level for a solo, then start with the fader 20dB down from the nominal starting point. However, don't get so tied into setting all the preamps for 0 that you find yourself not having enough useful operating range on a fader or that you are clipping an aux bus. The reality is that you may get the best result with some combination of or intermediate approach between the two extremes.

Wow, this was very well thought out and explained. Let's take the first part of the discussion regarding input trim. I would be fair to assume that any op-amp is not linear in it's s/n ratio throughout it's amplification range, and most all board pre-amps I have worked with find their lowest noise floor when operating at 0db to the fader. Pro sound 101 teaches us to adjust trim until the input is barely triggering the channel peak light (or hitting near 0 in a solo vu). This is the ONLY way I have of knowing the level state of the signal thru the rest of the channel. And no one can argue that the earlier in the signal chain the voltage is raised above ambient noise, the better. Again, if this were not the case, we could operate at any low level we wanted and make up gain just before the amplifiers. I know of no one who would take that approach.

Regarding aux sends (and much more importantly channel inserts). Why would outboard gear bother to specify their desired input levels if it really didn't matter. The answer is again, the non-linear nature of s/n ratios, and especially when using channel insert comps, the desire to know the signal condition entering the comp trigger. Last I looked, nearly all outboard comps have their threshold control calibrated in db, meaningless if my input signal is operating below that level, which it will since the channel send is taken after the input pre.

I do agree that with aux sends to outboard effects, there is a cumulative effect on level, but rarely have I even driven an channel aux send at 1:1 to the aux bus.

See, the math regarding s/n can only be applied when amps (and by that I also mean preamps) are operated at their optimum level.

When operating at the level the speficied by the manufacturer, I would get the most in regards to s/n ratio and it's relationship to headroom. And regarding headroom, I can't think of one occurance in live work where I've operating a rig requiring more that 60db of dynamic range and the resulting required headroom. What with comps and speaker processor limiting, I'd much rather have the noise floor down, then pursue headroom I don't need.

Good discussion, and forced me to analyse why I do things the way I do.
 
There's another reason to "zero" the faders and adjust trim pots accordingly. Speed of execution.

If you are really mixing a play live then the only mic, or mics, at full volume are the people on stage who are ACTIVELY talking or singing. If Romeo is talking to Juliet, then his mic is at unity while hers is at maybe -10, with the engineers fingers on the faders waiting to bring her mic up for her line and his mic down. This way you reduce phasing and comb filtering when people are speaking or singing near (or at) each other.

Now if you think about having 5 or 7 or however many people on stage and you're trying to do this type of live mixing...well, it's a whole bunch easier to only have one level you bring people up to for their line, as opposed to a different "full volume" level for each and every actor.

BTW, I realize this type of very active mixing is not for everyone and it's not how everyone does it. But a lot of people do. And regardless, I think that having the mics all zeroed is not about making the faders "pretty" but making for a more intuitive and efficient, fast-reactive, mix. This goes for voice and music.
 
Frankly, I never let making my job as live sound mixer easier take precedence over proper engineering principles.

I mix with my ears, not my eyes, and never rely on what the fader positions look like to tell me what I'm hearing.

All of this is not a question of how to get the faders in a position that makes mixing easier. It's about staging gain thru the channel strip, and all the talk about ways to make mixing easier and comb filtering (which is totally impossible to control with actors moving in and out of other actors/mics sound fields) does nothing to address the proper way to set levels thru the signal path.

Ever watch a recording engineer work? Once each track is layed, ensuring that the signal level is proper through each stage before arriving at the tape preamps or a/d converters, during playback the mix is accomplishing with faders representing the relative loudness of the desired mix. During playback, the engineer doesn't adjust trim so that all the faders line up based on the mix he wants, but rather, has all the signals hitting the console at 0db, and then the faders are adjusted to the desired mix, and the engineer doesn't give a hoot where the faders reside.

if you've got a way to trim the signal just prior to the fader, the approach of lining up faders may have merit, but as long as you're using channel input trim to reduce gain below optimum levels for the sake of making your job easier, then you're no longer talking engineering, but rather, sacrificing proper operation for the sake of comfort and ease of use.
 

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