Improvise. Adapter. Overcome.

all Western music that in some way is an offshoot of blues (so basically everything) builds on a few modes, scales and scale steps. we can go fancy with genre specific scales or scale types or modes, and accord for blue notes etc (which we should and could) but basically, we can get very very far across genres with a few notes/steps from a few scales. what we do for a basic setup is: we take a few scales that are very common for the genre(s) we're working in, and the scale steps/root notes/... that a bass line would definitely have. maybe just a few major/minor scales and their respective steps I, IV, V, VI, maybe a blue note or two as well we pick a colour scheme for each scale, whatever might work. like, let's say A minor is now greens/blues, just because. so step I in A minor would give us light cue "greens/blues 1", step IV would give us light cue "greens/blues 2", step V would give us light cue "greens/blues 3" etc. we set a MIDI input for our colour cues such that any MIDI A ( as in, all the As in the tonal range of your bass) will trigger "colour cue greens/blues 1". if you do this for more than one scale you'll need to check for repeating notes across different scales, and adjust your light show accordingly. so we have our show output and we have cued it to our MIDI scale steps. and we have our bass player. what about the middle part? we feed a DI bass signal to a Reaper track. we add the (free, native) REAtune plugin to said track that will translate the incoming audio to MIDI notes, and let it write its MIDI to another track. we then feed the resulting MIDI translation (so, the Reaper MIDI track out) to our DMX controller and to all the channels in it that need it, via whatever way we have to connect the two. alright, step 2: strokes determine effect speed/onset the idea here is: the faster the strokes come, the faster and more dramatic the light show - on cue with the bass strokes.

so at least for the basic setup, you'd couple a few position/dimmer effects that you want to be determined by strokes with a MIDI note as a trigger - so they'll start as soon as a stroke happens, and restart at the next one, or ring/fade out more or less naturally if the strokes are slower.

(think "fast playing -> lights come on and off almost strobe-like, movingheads waving about wildly and in chopped movements" vs "slow strokes -> lights coming on and off more slowly, heads moving less abruptly".) (also, ideally you'd pick MIDI triggers that aren't already occupied by colour cues. unless you specifically want whatever that does. try it. I haven't so far.)

right. we have our strokes-determined light cues set up. we have our MIDI triggers specified. how do we get our input? back to the Reaper bass track: we put another free, native plugin on it. this time it's REAgate, which we set to "gate triggers MIDI note". now whenever the gate opens or closes, a MIDI signal will be sent. we (of course) route the MIDI output from the gate into our DMX controller and to the respective channels/cues, and make sure the triggered note is the one we actually want to use.

then we tune (or set) our gate to a comfortable value that Makes Lights Good. which just means: sit at your desk for a bit and try out where you want it to be for the effects to be triggered in a way you like. depends on the way your bassist is playing, on general signal strength, whatever.

Pretty normal rush hour sounds, yeah? But this thing sounded something like this:

HELLOOOOO??? You gotta be on crack or just straight up deaf dawg.

Anyway, we installed extra wireless, relabeled and patched the Dante network, built a console file from the ground up to keep it in line with the other venues, and it felt like a bunch of other stuff.

So I can’t sleep because sick and headaches so imma babble a bit. Specifically about the filters in this sucker.

To the point of 140hz being ambitious for an HPF I’ll say this is about where I would start with a good chunk of these because mixing dialogue has to focus on intelligibility, so reducing the range of fundamentals in the human voice does a lot for that. We’re working with lavs, which live off axis and right against the resonant cavity of the chest. In addition, these boxes are very flat and very balanced, so running an additional HPF for the house buss makes sense when you hear it.

The LPF is lower than I would set, but that’s also dependent on me utilizing a de-esser on my channel for sibilance management. Generally I run mine closer to about 8khz so we don’t lose the intelligiblity of that high end. But more importantly we get to talk about my fav lil fact of cardioid mics: the presence boost.

So when we talk about proximity effect in mics it’s easy to think of it as a singularly occurring phenomenon; speaking right on the grille of a 58 sounds bassier than holding it back. There’s proximity effect and then there’s ‘not’. Except proximity effect is a continuous effect, you will get varying degrees of low end build up and loss at all positions. It’s a spectrum I guess. So to combat this a lot of directional (cardioid, super-cardioid, etc) mics will design a ‘presence boost’ in the capsule to offset proximity effect. It’s part of why a 57/58 works in a lot of settings. 2-5khz, maybe up to 8khz and then there’s the whole condenser vs dynamic blah-blah-blah.

Lavalier mics in live production are often cardioid or super-cardioid (shout out to my hated wl184s, you sound like wet cardboard) and so they too have a presence boost built into them. And since the microphone is designed to sit off axis against the resonant cavity of the chest they are often a brighter capsule period. So these capsules are Shure UL4 lavs which are noticeably brighter and deeper than their aforementioned counterpart and almost always requires an LPF somewhere in the chain; generally on a buss for processing workflow reasons. Oftentimes with Wl184s I didn’t need to add a LPF because the presence boost was good enough and again they sounded like wet cardboard and I wound up hacking out the midrange to cope.

Soooooo what June? You’ve babbled about eq and filters and hating this person’s bad ring at midnight on a Friday with a killer headache for what?? To affirm that at minimum the filters here made sense and were a decent call for the capsules and P.A that they had. It’s super easy to roast this (and we should) but audio is of course a “sounds good is good” game just like music. The technology itself impacts the choices we make and nothing can be evaluated independent of the other. For another long winded example, lemme just ramble about omnis for a minute.

I fucking love omnidirectional mics. And if directional mics experience proximity effect as a continuous phenomenon because they are directional then omnidirectional mics do not. There’s a whole article from DPA I’ll link later about it. But basically, omnis don’t suffer from low end loss as severely and are more balanced as they run off-axis. It’s why theater and broadcast and film/tv utilize omnis more: consistent tonality, flexible in mounting choices, and they just sound good. Those same omnis also, generally, do not have a presence boost built in because they aren’t combating proximity effect to the same degree.

We’ve got some DPA 4060 omni lavs that sound fucking delicious because their low end is smooth and consistent and unless we have the presence capsule added (an interchangeable piece to brighten up the capsule) the high end is very nicely balanced. If I were ringing out that capsule I’d run my HPF higher, probably 180hz with a reduced slope of -12dB/octave instead of -18dB/octave to adjust. But that’s because I know it and the ring out itself would be drastically different for the capsule type. But omnis in corporate with untrained speakers are fucking haaaaard. If it’s my venue then I’m happy to do it, but I would never set up another A1 with omnis and tell them to cope.

Okay, full circle now. The filters more or less make sense, they are a design choice, but having spent time in the room, redone the ring out, and worked with these capsule specifically I can give a pass on these choices. The other bands though?? Jail. Jail for audio.

these 3 are my core list, might fuck around and add some sisters of mercy bc when it comes to female vocals you can hardly do better than ofra haza

and then there*s always iamx and hozier

Since it's all corporate audio over here (except for the musical performances that come through occasionally) there's not really a lot to consider. But sometimes you get told: 'Hey, there's 4 people and a phone call and we're gonna do a bunch of redundancies and link this into comms to set up IFBs and take a good long while to test every single thing' and you decide to do your job.

So we start with sources. This show was pretty straight forward: booth and backup handheld mics, a primary and redundant call, primary and redundant playback, royalty free music. Each mic got the same suite of tools: DANSE, dynamic EQ, and a spot in the Dugan. The booth mics were Shure MX418 goosenecks and were pretty bright, so they get some good ol' hi-pass and lo-pass filtering to match with the KSM8 capsules on the backup handhelds. Basic comp and de-essing on the built in channel dynamics.

Since we were splitting the Axient wireless gain structure for those were set via digital gain, but all other inputs fed an analog split on the floor going into two separate RIO3224D stage boxes for each console. Monitors drove phantom for this. Playback devices ran synchronized QLAB playback over some UA Volts into the split. Calls came in to the Riedel comms system and then looped out into the Dante network for IFB workflows.

DANSE in the Yamaha Rivage ecosystem is noise suppression based on the Cedar DNS hardware and plugins. It is really solid for gating various frequencies, cleaning up resonances, hissing electronics, or HVAC. Dynamic EQ is essentially a reactive EQ that makes a cut when a threshold is passed. I most often use DynEQ to manage proximity and plosive effects in directional mics: low end build up is like the most noticeable changing quality to me in a microphone. Mic technique is obviously not a huge priority in corporate and thus a tool like this really helps to maintain a consistent tonality. It also works well for managing sibilance depending on insert position or just catching minor tonal adjustments. Each band has it's own threshold, boost/cut range, q width for the frequency, and envelope.

The Dugan automixer is pretty classic. Every digital console has their own version. It's a gain sharing device to ensure that when multiple microphones are open and talking the overall output gain doesn't change. If you do corporate, you'll use the automixer. This is also important because we feed our calls into the automixer with a lower priority so that the microphones can interrupt them easier. FWIW you can also split up the Dugan so that on more complex shows you can assign certain channels to certain groups so that they don't all effect each other equally.

Each source feeds it's own relevant mix type where we apply some buss compression and EQ because passing it to the stereo which then folds down to our output matrices.

Mics in the booths -> BoothMic mix

Incoming calls -> Calls mix

Playback sources -> Media L/R mix

Backup HH -> HH mix

Music -> Music L/R mix

In a Yamaha ecosystem I really like the BussComp369, modeled after the Neve 33609 compressor. The auto-release functions are really smooth, the input adjust proportionally adjusts the output gain so you can drive into the comp without big jumps in signal, a higher threshold limiter and it's ready to roll out. These are the 'nice' compressor on the older QL and CL series too and I'm still a sucker. The other comp I like on media sources is the FET Limiter, which is a perma-mounted 1176 comp. It reacts to things nicely, adds a bit of saturation without getting super distorted and is super fast for catching transients like I expect in media of varying quality.

We also add some mixes for mix-minus which are essentially your program (main) mix minus the source they are sending to. If you have a call coming in then the mix you send that call wouldn't contain itself. These are pretty easy to build on most digital consoles with a lot of them having dedicated mix minus buss architecture built in now, but in analog mixers or some other digital ones you can manually build these. You'll see on the right side that mixes 13-16 are set to 'mix minus' and their 'n-1 owner' is the channel you want minus'd from that mix. (god I'm tired of mangling this phrase by now)

The stereo mix then feeds to a few matrices, notably 'Broadcast L/R' and 'MonoPGM'. The mono matrix sends a fold down wherever I need it to go: comms, listening stations, recording devices, etc. Both program outputs get the Portico Master Buss Processor because it has a few little features that I like to have: blendable comp circuits, a hard limiter, additional stereo sweetening with the width control, and a side chain for my low end to pass uncompressed.

This feeds an output to a T.C Electronics Clarity meter so I can monitor my broadcast levels and feeds all recording, streaming, calls, and miscellaneous destinations. Outside of the console it carries over a Dante network and patches mix minus busses into comms systems to set up IFBs for remote talent communication, sub-mixes feed to monitors console so they can mix for talent, iso records and direct outs patch to a Tascam Dante recorder, and backup recordings to a USB drive on the console itself.

Anyway, that's how you spend some time building a stressful, highly important and incredibly boring conference call to be nicer than a Zoom call. This seems like a lot, but honestly most of this is built into my default file at this point because it works so well for mixing broadcast.

OH! One more tip/trick. In Yamaha consoles (and probably others but I'm a one-trick pony) you can link channel parameters together so that adjustments on one will follow on the pair. So if I make a cut for -3dB at 600hz on my primary call channel the linked back up channel will receive the same adjustment.