Making the M7CL sing

I’ve been blessed to spend the past 4 years or more mixing on some really great, really big desks.  When I first arrived at Kensington, we owned a Yamaha PM1D that, while I’d had previous experience on years before, I enjoyed learning inside and out.  Then 2 years later we changed directions and embraced the Digidesign (now Avid) platform.  Along the way I also was able to get my hands wet a bit with Yamaha M7CLs.

There are some tricks I’ve learned along the way from absorbing content from accomplished engineers in the field that made life mixing on the Venue and PM1D a bit easier and gave me better mixes.  A new challenge since coming to NLC.tv has been trying to find creative ways to get the same bang for the buck out of the M7CL.  I had already started exploring some of this process in my last few months at Kensington but I’m working now to flesh them out a bit more.  I’m going to share some of my favorite M7CL tips and tricks over the next few entries (there’s too much here for a single post).

Today we’re going to tackle parallel compression.  I’ve written on this before, as have others, but in its simplest form, parallel compression means double bussing a set of inputs to two different signal paths on the console.  In the first path, everything remains clean and unprocessed.  In the second path, a nice compressor is placed over the signals and they are compressed as a group, usually pretty hard with variable attack & release times depending on the song.  Then the clean and squashed signals are recombined before going to the stereo bus on the mixer, for me usually at a 2:1 ratio of clean to squashed.  This is especially magic for vocals and snare/toms for me.  Mixing the styles of music that I do, vocal intelligibility is normally one of the most important goals I’m fighting for and getting the vocal to sit nicely in its place with the rest of the band can be challenging.  Since I started implementing this parallel compression trick last fall, it has done wonders to the ease with which I can accomplish vocal consistency I really like.  It becomes an even more powerful tool the larger the vocal group becomes.  At Kensington it was normal to only have a single lead vocal and perhaps a BGV or two.  At NLC.tv, 5 to 6 vocals is the norm with sometimes as many as 7 or 8 on a given weekend.

Doing the parallel compression thing on an M7CL is really easy.  First, I like to set up 2 busses as fixed busses instead of variable so I don’t have to worry about making sure the sends to them are all at unity.  This can be accomplished under Bus Setup in the console setup menu.  Next, as long as I have enough mix busses available, I like to set up one buss for the clean group and just call this one VOX.  I unassign the VOX channels themselves from going straight to the L/R buss and instead route them to this VOX “subgroup”.  While I plan to keep the processing here as clean as I can, especially the larger the vocal group you have, it can be really handy to have a single place you can grab an EQ and deal with a problem area that effects all of the vocals during the heat of mixing.

Now I also route the VOX channels to the 2nd group that I call VOX Smash.  This group is setup just like the first one except on this one I engage the compressor on the buss, set to a 6:1 ratio with a medium attack and release.  The M7CL has an excellent feature called automatic delay compensation so even though the same channel is going through two signal paths with different processing times, they stay perfectly in sync so that when they are combined into the master L/R buss, they are still in phase with each other.

If you’ve never tried this concept before, I can’t suggest strongly enough that you do.  I always had the preconceived notion that a trick like this was only available to execute on larger desks but have been very pleased with the results I can achieve on our M7CLs and have started sharing the love with all of our engineers on this easy and effective mix technique.

The IEM missing piece

This one goes in the category of why didn’t I think of this sooner?

Last time I wrote about the improvements to our IEM system that have come from implementing a new transmitter/receiver combo and some additional ear piece options.  The problem with adding more wireless into an already congested environment is a higher probability of failure.  I’ve personally managed to avoid Murphy’s Law when it comes to IEM for a long time but I realize it was only a matter of time before someone’s system failed during a service and we’d be in big trouble.

Enter Sidefills.  A friend spent a couple services with me post-Christmas and suggested that what we were missing was a good holistic stereo mix that will fill in the missing pieces when someone pulls an ear out and also provide some added energy to the first couple rows that might be a tad light from the PA over their heads.  It seemed like a good idea.

In my past life as a monitor engineer, sidefills were an important part of getting a great onstage sound since I worked with lots of vocal groups who would often times only wear 1 ear anyway.  For some reason, I never even considered it at KCC since our old PA already put energy everywhere except where I really wanted it – the prospect of adding an additional full-range sound source was unappealing.  However, in the new reality of a controlled PA without a ton of stage spill, it seemed possible this might just work.

Work it does.  I have a set of EAW KF300s that have been unused for a while.  I put them on top of the subs, about 5 feet off the ground, pointed in towards center stage.  Stereo mix gives more clarity & separation than mono, add some EQ to smooth out the rough edges of the boxes themselves, and add a few milliseconds of delay to the PA so the clusters in the air are in relative time alignment with the sidefills and the result is really good.  I’m feeding these boxes from FOH subgroups so they essentially get a rebalanced FOH post fader mix.  The mix is split up to rhythm, band, and vocals.  Using this approach keeps them specifically music-focused – playback and speech mics stay out of them.

We still keep a set of wedges on the front row for lead vocal monitors.  The addition of the sidefills allows the vocal to literally be surrounded by themselves – just a touch of vocal in those front wedges pulls the singer’s image forward.  Needing less band in those front wedges reduces mud heard at FOH since the sidefills are actually working with and as a part of the PA rather than against it.  I find myself putting 100% vocal in them, 80 or 90% band, and 60% rhythm (since the acoustic kit on stage adds its own ambient sound).

If you mix on a primarily IEM stage as I am and haven’t revisited sidefills in a while, this experience enthusiastically suggests they’re worth a try.

Getting your mic positioning in order…

Microphone positioning and technique is largely a matter or personal tastes – usually whatever “sounds right” probably is right.  Nevertheless it’s a good idea to remind ourselves of some of the basics for getting there.

Following are a few tips that you might consider following when micing musical instruments for sound reinforcement.

• Try first to get the instrument to sound good acoustically before miking it.
• Use a mic with a frequency response that is limited to the frequency range of the instrument.
• To determine a good starting mic position, try closing one ear with your finger. Listen to the sound source with the other ear and move around until you find a spot that sounds good – put the mic there.  Remember, this may not be practical (or healthy) for extremely close placement near loud sources.
• Remember that the closer a mic is to a sound source, the louder the source is compared to reverberation or ambient noise.
• Place the mic only as close as necessary, keeping in mind proximity effect.
• When possible, use as few microphones as possible due to the Potential Acoustic Gain rule which tells us (among other things) that the volume level of a system must be turned down for every mic added in order to prevent feedback.
• If the sound from your loudspeakers is distorted even though you did not exceed a normal mixer level, the mic signal may be overloading your mixer’s input.  To correct this situation, use an in-line attenuator or pad to reduce the signal level from the microphone, or just back it away from the source some.
• More than anything, experiment and listen!

(Thanks to Shure for the tips.)

Ambience Mic Thoughts

I’ve had quite a few questions regarding ambience mic technique, what to use, etc.  We now use three sets of ambience/audience mics, each with a different purpose for capturing the room.  Inspired by some great discussion on the Digidesign User Forum, my implementation is…

Mics #1 are a set of Audio Technica 835 shotgun mics (purchased for this project) on the outside corners of our stage, mounted just under our side screens, pointed out into the room.  These are the primary audience pickup source because so much of the audience’s energy is directed towards the stage that these do a great job of capturing without adding too much PA to the mix.  They’re inexpensive and sound good for what I’m using them for.  You could certainly upgrade the make/model here to something more boutique but this serves our purposes well.

Mics #2 are a set of Crown PCCs placed on the lip of the stage at the 1/3 and 2/3 lines from left to right.  These combine with the shotguns to provide some presence and immediacy to the signal since another set of shotguns wouldn’t be acceptable aesthetically in these positions.

Mics #3 are a pair of Shure SM81s hung from the first catwalk, almost at the half way point of the room, also on the 1/3 and 2/3 lines to the stage.  These mics serve almost entirely the purpose of room ambience – making the mix sound like it is happening in the auditorium rather than happening in a dead studio.

The three sets of ambience mics serve dual purpose for us – ambience for IEM and ambience for the broadcast/record feed.  In the IEM world, we lean most heavily on the shotguns at the front of the room at the corners of the stage and supplement that with the PCCs on the front of the stage towards the center.  These mics are used mainly because they provide localization of what the artist is hearing in their ears.  For example, when someone to the artist’s right calls to the stage, we want everyone on stage to turn their head in that direction.  Likewise, if someone on the front row is singing their heart out, it’s cool for the artists to be able to sense that from those PCCs on the front of the stage.  That localization goes leaps and bounds towards breaking down the isolation.  Every artist is different with how much ambience is just right.

We may still look to add an additional set or two but I’m stuck because adding them will mean needing to deal with delay times between those mics and the rest currently in place.  Two of my current three sets are directly in line with the PA so there’s little issue there.  The delay induced by the third set certainly does some phase stuff to the mix but the added space it puts in the mix when the speaker is up is worth the compromise to me.  So we’ve dodged the bullet to this point and I’d like to avoid having to go there with more mics.  This is one of the only limitations of creating the broadcast mix inside the Venue rather than in an outboard mixer/processor.  There is no way to have the degree of time alignment control that would be needed in the broadcast feed without making other more significant sacrifices.

The shotguns are the most important piece of the puzzle for me because their pickup pattern is most effective at picking up the audience and rejecting PA and stage bleed.  The PCCs are next important to fill in the center stage imaging – think of the shotguns panned hard left and right and the PCCs panned at 9 o’clock and 3 o’clock or 10 and 2.  The hanging mics are almost halfway back in the room and that time delay adds depth to the ambience space. For IEMs, it can be distracting.  But for broadcast, it enhances the size of the room and makes it feel more real when you’re listening back.

Unity Mixing Followup

I ran across some cool conversations happening on the Sound on Sound forum a few weeks ago in reference to unity mixing.  If you’ve missed the various things I’ve written about the topic, there are links on the homepage of this blog to the articles.

I’ve also seen another forum where one poster blasted me and called me a moron for approaching this topic.  It has been fun to see how literally some people take this whole thing and think I’m somehow saying you should literally mix your show from the gain knobs or advocate poor gain structure in your console or the rest of the system.  QUITE the opposite.  I run my console really hot because we all think the Digidesign mic pre sounds better when its hit pretty hard and the desk as a whole sounds great when you get up and go with it.  I think some are missing the point of my suggested unity mixing completely, so let’s dive into this just a bit to hopefully clear it up.

The whole point of this discussion is to make the inputs well mixed while setting gain, rather than just blindly going channel by channel and setting each to maximum individual gain.  It doesn’t have to be perfect but just get things in the neighborhood so you don’t have to run some faders -30 or -40 dB below others.  This makes mixing monitors from the same desk (which we do at every Kensington campus) MUCH easier.  In this configuration, the inputs are well mixed, so that the sends to the monitor mixes and FOH faders are (more or less) at the same place across the board, making adjustments for the band a lot easier.  If I know the band well, I can often dial in their monitor mixes from the FOH board ahead of soundcheck and nail it with few, if any, changes needed.  We’re still running the inputs as a whole as hot as they should be through the desk so let’s not get into the whole topic of maximizing digital bits in the desk and the like.

Another example… imagine trying to adjust a monitor send, pre-fader of course:

A) with a very hot input, therefore with the FOH fader pulled down a lot, and
B) with a low input setting, with the fader pushed nearly all the way up . . .

‘A’ will be very sensitive to tiny adjustments at 9 o’clock and
‘B’ will need large movements at 4 o’clock.

With a mess like this all over a big mix, the experience is not nice trying to keep the band happy with their wedges and that is what gets so many of our engineers in trouble!

Second, if you end up with a difference, for whatever reason, where one fader is sitting at -30 or -40 dB most of the time, you have a MUCH harder time mixing due to the same amount of travel for a 3dB difference at or near unity, now gives you a 10-20 db difference with every move (Or whatever it works out to on the board, but a LARGE difference).  This makes fine tuning a mix really difficult.

That’s it.  Does this help make the concept clearer at all?

Inside IEM Mixes: Lead Vocal

We finally made it to the last and likely most important mix in the IEM series…lead vocal/worship leader.  In this particular example our lead vocal is also playing acoustic, so not surprisingly you’ll hear those two inputs most focused in the mix.

As with some of the other mixes we’ve discussed, our vocalists seem to largely prefer their voices dry…if they want to wetten things up, adding more of the extras channel does the trick.  This works well in our particular setup since the vocals are fed to the IEM system direct off of board channels – adding verb would increase the complexity of routing.

I strongly encourage everyone who wears ears to keep both sides in all the time.  My primary reasoning is that your brain does a cool thing when it receives similar input from both ears in summing that source 6 dB internally.  So, if you only wear one ear, you’re going to have to turn up your IEM mix approximately 6 dB in order for it to feel as loud as both ears in.  Obviously you have to be really careful with this because an untrained artist could easily cause themselves hearing damage night after night of crazy levels in their IEM.

I say all that to say, the mix an artist wants changes pretty dramatically depending on whether they are wearing one or two ears.  These mixes you’re going to listen to today are one ear in mixes, so you’ll find that the acoustic, vocal, and click are WAY more out front then in the other mixes we’ve analyzed since the artist is getting a lot of their sound ambiently through their other ear not wearing the IEM.

Our philosophy at Kensington regarding ears has been to do everything possible to gain band acceptance.  By doing that, the stage volume lowers significantly and vocals have a much easier time just hearing themselves through a wedge.  I’ve mentioned that we have 6 channels of wired ears and 1 wireless transmitter/receiver system.  Because of this wireless limitation, we haven’t pushed very many vocalists to jump to ears.  This isn’t necessarily a bad thing since vocalists often have a much harder time adjusting to performing through IEM systems then instrumentalists because it can effect their pitch center, how loudly they perceive their head voice, etc.  So we’ve had good success getting musicians on ears and, by doing that, getting enough sonic space on stage for the vocalists to be able to hear themselves comfortably through wedges and not have to deal with their transition yet.

I’ve heard from another large church that they have the policy that a new vocalist must sing BGV on ears for at least 6 months in order to get used to them before they would ever ask them to sing lead VOX.  I can certainly see the wisdom in that approach.

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We’ll wrap this whole thing up next time with the live mixes of the 2 songs we’ve been using for test material so you can hear overall context.  See you then!