Not much to report these days. We’ve been keeping busy adding new covers, polishing up live versions of the original music from the album we released last summer and tweaking sounds for the new live show.
As the eternal optimist (though I’ll admit, my shine is waning) I’ve been able to take this time off and re-invent the details of the “live fits experience”. Music, and the performance of it, is never static, always dynamic, try as I might to get “one live sound setting to rule them all”. What follows may be “music geek” stuff, but I’ve got some time and I like to share. I’m still learning and am aware I don’t know everything. This is meant to share what I’ve learned. Take it or leave it.
I noticed quite some time ago I mix very differently in the studio for a record than I do for a live show. It used to be we lacked the outboard gear to recreate the studio in a live environment, so that was a limitation. Also, so much compression on vocals (and other stuff) live is problematic due to compressed mics hearing themselves and casting awful feedback. Lastly, parallel compression, delays and reverbs in the studio can be very controlled, live we don’t have a traveling engineer with us due to budget constraints, so how to come up with a “one size fits all” solution? Nobody wants to hear delays when you try to speak between songs.
First off, I learned a long time ago form my buddy Neil that setting a unified gain level is huge. I have used -10db ever since. That means everything coming into the board peaks at -10db. Nice and cool, no clipping. Smooth. This will help keep your levels from clipping when you start adding everything together in a buss.
Let’s start with the drums and parallel compression. A trick I am quite fond of in the studio. Just so happens our x32 digital sound board has a compression mix parameter. I have made quite a bit of use of it on the drums. I even have a small amount across the instrument buss (like 20%) to glue everything together. A tiny bit goes a long way! Since our drums are all digital (except for the hi-hat, thats real) I need to be sure the “room” coming from the TD-50 feels real and is loud enough. So many time when I hear triggered drums mixed live the sound… flat. They sound like samples. This is what gives these kits such a bad rap and why drummers profess to hate them. Understood! I wouldn’t want to sound like crap either! By making sure the room around those samples is brought up in the mix, the drums become so lifelike, people not looking would never know you are using a digital kit. I use the 1176 compressor in the x32 across the cymbals and room sounds. This makes a huge difference. The plus sides of a digital kit are big. You can control the volume of everything, in any room, for any show. How many times have you had a show ruined by a snare drum or cymbal that was just way too loud and reverbiant for the room? The other great option is multiple kits! Why use one drum kit the whole night? Swap ‘em out (level matched of course) and keep the overall interest of the audience high. I’ve also made a bunch of my own drum samples for use live. I could write a whole book on this, but for now all you need to know is you can sample your favorite snare, kick tom, whatever in the studio and play it back on the TD-50. Blend it in with other sounds for realistic playback. When the Conniption Fits roll into a show, you’re hearing at least 10 completely different drum kits cycled through the night.
Onto the bass. Jamie and I came up with 4 solid amp options with varying stomp boxes he uses throughout a show. The Helix makes all of this a breeze. His main goto amp is the Ampeg model. I have some compression after the amp and effects, an LA-2A built into the presets before it hits the soundboard. In the x32 I call up the LA-2A once again and dial in just a touch more of light compression on the bass channel. Jamie is a very dynamic player and we need him always present and punchy, but those slaps and pops would take your head off if not for the compression!
On my guitar, most of my Helix presets are made with dual amps panned hard left and right. Because we are a trio, I’m trying to make everything huge. If we had another guitar player I wouldn’t do this, instead both guitar players would use single amps panned somewhere less aggressive. As it is, however, my guitar is wide so it can be loud and present without covering up the bass and drums that are down the middle. On the x32 I add just a touch of compression at a mild 3:1 ratio to mainly keep the preset changes at a unified volume. I have also spend the time with a meter going back and forth from preset to preset making sure they all peak at the same level. This is a MUST for any guitar player. How frustrating for any sound engineer when everything is dialed in and suddenly, bam, out of nowhere a screaming guitar because they changed presets!
Vocals. Maybe the most important element and the most “mailed in” by many bands I’ve seen. All my favorite records have highly processed vocal tracks. That doesn’t mean tons of delay and reverb. It means they are EQ’ed and compressed to be precisely what complements the singers voice. Like I said earlier, I could never compress a vocal live like I do on a record. The feedback would be a nightmare. But I think I’ve come up with a close compromise. I have set up three stages of light compression. It’s on my vocal pedal (TC-Helicon Voicelive), a touch on the channel, 4:1, 48ms attack, 16ms release, and again on the vocal buss 4:1 50ms attack, 63ms release. I set the threshold on all so it just digs in a little bit on a normally sung phrase. It’s almost the same with the eq. There is some brightening going on at every stage to achieve that clear, upfront vocal without the ‘woolliness’ you would hear on a record. Feedback is always my enemy here, and always where I have to compromise on the brightness and then, compression. As for effects, on my vocal they all come from the pedal. I need to be able to turn them on and off to suit the song, and the show. Again, nobody wants to hear delays when talking in-between songs! I’ve come up with a couple of reverb delay combos that work very well. Reverb is a plate with a maxed pre-delay. I think it is 100ms, (Tc-Helicon limitation) but I’d go to 150ms if I could. This gives the dry vocal presence before the reverb kicks in. Keep the decay and level of the verb moderate so it’s natural. As for the delay, I love slightly wide lightly chorusing delays that are then fed into the reverb a small amount. This keeps the delay from sticking out like a sore thumb, and still draws attention to the singer and the space around them. I’ve set up three different presets. A quarter note delay, a quarter & eighth stereo delay, and a classic tape slap delay. All the delay times are offset just a hair on the left and right feeds for width. Every song works with one of those delay set ups. Ideally, I like the dry vocal right down the middle with the delay/verb wide around it. As for doubling and harmonies, I’ll use a tight double, and a single third above harmony. Both of these I can kick on and off as needed. That’s it. On Jeff and Jamie’s backup vocals, I have them fed to a plate and a wide stereo slap in the x32. It seems to work for the parts they need to do. I don’t want them washed out, but I don’t want them in your face either.
As for the main mix, I love the Fairchild mastering compressor model in the x32. I use it in the Mid / Side mode to bring the stereo information forward just a bit. You can easily overdo this! But again, as a trio, I am trying to get us sounding as huge as possible. I’ve set it so the gain reduction only moves of very hard hits or sections, but is not really digging in at all. The sides dig in a touch more and I bring the gain up just a bit to widen the stereo image. This really works wonders for getting the drums and guitars to surround the listener. again, with everything coming from direct sources, it allows us to have 100% control over the stereo spectrum in the room. The Fairchild is an odd beast with some unique parameters. Here’s what I’ve learned.
Bias gives us a control over the ratio of the compression as well as the knee width, adjustment from 0%-100% in 5% increments
From the UAD Manual: Bias controls the ratio of compression as well as the knee width. As the knob is turned clockwise, the ratio gets lower and the knee gets broader. The threshold also gets lower as the knob is turned clockwise. It’s more technically accurate to say that this control simply changes the knee width, since no matter where it’s set, the ratio always approaches true limiting eventually. However, the knee becomes so broad that it becomes more practical to speak of the ratio changing, because for reasonable amounts of compression (less than 25 dB), this is the case.
Balance adjustment from 0%-100% in 5% increments.From the UAD Manual: Balance controls the bias current balance. It always controls one channel of the compressor, regardless of what the nearby Meter Switch is set to. The point of perfectly calibrated bias currents is achieved when the screw slot is at the 12 o’clock position (the default value). At this setting, the amount of additive signal deflection (“thud”) which happens due to an attack is minimized. Setting this control counter-clockwise from this position results in a thud of one polarity on transients, and going clockwise produces a thud of opposite polarity.
If you’ve read this far, you are probably a musician or sound engineer. I’d welcome your feedback, comments or questions. Lets all come out of this pandemic a little better than we went into it.
Cheers,
Stevens (email me)