Video

Not necessarily. On this particular console, we've purchased third-party plugin packs. The console has many onboard resources, as well, but one of the most powerful reasons to go with some of these modern digital consoles is being able to load in third-party, studio-quality plugins for the live environment. An example...Waves Audio has great plug-in packs for purchase.

Hey Ken: We recorded the tracks live in the room during a Sunday band rehearsal. We used Protools to record, which is integrated into our Avid D-Show console.

We rigged it up for this video. We used a Protools interface with eight outputs, and ran the individual tracks into the inputs of the console, simulating a live band. For example, for drums, we ran the recorded kick channel output into the first channel of the console...and so on for each instrument.

As always, it depends. One of the downsides of compressing anything is that the noisefloor (background noise outside of the closely-mic'd source) volume is increased while the closely-mic'd source's volume is decreased (at least the peak volume moments). In most stage layouts, the background vocalists appear near the drummer, whose loud drums and cymbals tend to overtake the microphones of the background vocalists. Therefore, when you compress the vocalists, the drum volume (and usually the cymbal volume most) increases. So...you can compress them, but generally, it causes more problems than it's worth.

Virtual Soundcheck is Avid's name for the ability to multitrack each instrument in ProTools while the band is rehearsing...and then be able to listen back to those recorded tracks later (after the band leaves) as if the band was still playing. It's like getting a limitless soundcheck. It's one of the most game-changing solutions offered to a modern FOH engineer.

That's all personal preference. Some people love the ring, other people hate the ring. It all depends on how annoying it is...at least from my view. You can usually find it between 100Hz-300Hz...and can cut with a narrow band to limit it. When you get your EQ set to that 100-300 range, you can boost the EQ temporarily to find the exact tone of the ring. Once you find it, cut it to whatever degree you want.

When we recorded this video, we were using live multi-tracks that we captured on a Sunday morning. We patched each channel into the back on the M7CL inputs as if each channel were coming in live. (Those were not studio tracks, but actual live drum tracks.) The only difference between that staged environment and the live environment (for practical purposes) is that in most churches, the acoustic sounds of the drums will be leaking into the room along with the mic'd sounds of the drums. So...the principles we use apply, but in a live room with live drums, more volume and low-mid energy will be coming off the stage. In addition, chances are that the snare and cymbals would be bleeding into the vocal mics. So...there are certainly some differences, but the principles of what to listen for remain the same.

Ken: this is a common problem. I'd recommend having her use a handheld microphone in these moments. (We've done that before in our room.) It may seem funny, but that's the only practical way to get her volume over a band at reasonably high volumes. To keep it simple, handheld microphones, and their typical focused patterns, have much more headroom (gain before feedback). So, you can push the volume much louder before hearing the feedback (ring). Try that! She can transition to her countryman once the band is done. (And, I'll admit, the way her voice will sound will be different between the two mics.)

Fade time is important for automating SMOOTH transitions. I can think of almost nothing worse than sudden changes in volume between elements. That slow fade will cause the changes between band and video, speaking and music, or any other drastic change in volume almost subliminally smooth. On digital consoles, I like using a 4-5 second crossfade for most things. Consider a moment at the end of a worship song, when the pastor may come out and speak...but you only want keys playing. You have a challenge because the guitars, drums, bass, BGVs and whatever else are still up in your mix. You may opt to create a scene where everything is removed from the PA except keys and your mainstage communicator. But, what if the band is still noodling softly, or what if they played a big note and are holding it? A sudden change of volume will be jolting to the audience, so a longer crossfade will create a gradual decrescendo of the instruments you don't need in the next scene. Much better than quickly hitting the mute buttons on everything, or having to quickly, but smoothly, pull down the faders on individual instruments, VCA groups, or whatever else. I over-explained that, but wanted to be clear.

With digital consoles, saving settings has never been easier. So, there is absolutely nothing wrong with saving your EQ settings for singers who are there every week...AS LONG...as you don't tune out and stop listening with critical ears. Changing microphones, a singer having a cold, or a myriad of other reasons can change the tone of her voice. So, keep listening in spite of saved settings. There is a lot of speculation between the differences of Type 1 and Type 2. I believe the console defaults to Type 1. This may sound snobby, but I always preferred the subtle softness of the Type 2 EQ. If my memory serves me correctly, the waveforms were smoother (rounder) on Type 2, which might explain why Type felt more musical to me. Here's a better explanation from a forum: "Yamaha digital consoles (except PM1D) have a choice of EQ Types. They are just different ways of doing the maths that results in EQ. We recommend Type II as being more "musical" or "analogish" and in almost every case it is agreed it sounds better. So why have type I? Well Type II does exhibit some unexpected EQ responses when used for several very strong negative EQs, the type of multiple notch filtering used by monitor engineers or for bad problem solving. Type I is always predictable, but practicably not as nice!! So the choice is left to the user." -Tree Tordoff www.blue-room.org.uk/lofiversion/index.php/t10831.htm l I've been using Type II exclusively since an initial 2 minute headphone test years ago, but after reading that Type I is more predictable, especially in regards to cuts, I'm wondering if maybe I should start using that for speaker EQ. Looking at a transfer function, do you think you would get a smoother or more predictable response by using Type I since you had to worry less about close band interaction? I'm thinking Type I for the science/calibration side of the equation and Type II for the art/voodoo side. Does that make sense?"

Thanks, Ken. Keep searching for content. It's one of the most common consoles on the planet.

If you're lucky enough to have analysis software in the room, one of the coolest tricks is to bypass the EQ on the tom channel (so the console is making no adjustments to the frequency range). Now, boost the volume so the analysis reference mic gets a good listen to what the mic is amplifying in the room. Check the frequencies in the software...and find the center of the waveform of the loudest frequency (depending on the tom, it'll be between 130 HZ down to 70-80 HZ). Now, reapply the EQ and boost slightly at the frequency that you wrote down for being the main frequency of the tom. This changes are soon as retuning occurs, humidity increases/decreases, and the head stretches. So, this is something that needs to be done relatively often for best results.