As summer is winding down and fall approaches, you know what that means… Christmas is coming!
Many church music departments have already begun planning for their Christmas music productions; if you haven’t yet, it’s time to get started.
Here is a question to consider: is your mixing console everything you’d like it to be?
If the answer is no, there is good news.
Several manufacturers have recently released new digital mixers that are both cutting-edge and budget-friendly. I know that sounds like an oxymoron, but it’s true!
Here is a brief comparison of three innovative new mixers on the market.
If a digital, “all-in-one” type of mixer is what you’re looking for, I can easily recommend any of the above.
The Behringer X32 can be used for larger venues and permanent installations, while the Line 6 StageScape M20 and Mackie DL1608 are much more portable and useful in smaller venues as well as youth rooms, children areas, and anywhere that doesn’t require a large mixing desk.
That said, any of these consoles would be a wonderful way to mix and capture upcoming events at your church.
Church Audio Video specializes in the design, installation and support of high-quality and affordable custom audio, video, lighting, broadcast and control systems for worship facilities. For more information, visit their website.
GV Audio Chooses Yamaha CL5 For Main Stage At Regina Folk Festival
Console used to mix performances by Emmylou Harris, Arlo Guthrie Tribute to Woody Guthrie, Timber Timbre and many others
Held annually at Victoria Park in Regina, Saskatchewan in August, the Regina Folk Festival is a much-anticipated celebration of folk music.
With more than 5,000 concertgoers in attendance each of the three days, the concert’s main stage this year featured Emmylou Harris, Arlo Guthrie Tribute to Woody Guthrie, Timber Timbre, Cold Specks, Shad, Mavis Staples, Serena Ryder & The Heartbroken, Élage Diouf, Austra, and several others.
Regina Folk Festival used the production services of local sound company GV Audio, which chose a new Yamaha CL5 digital audio console and two RIO-3224D remote I/Os for main stage front of house mixing.
“Most of the visiting engineers were very familiar with Yamaha consoles, but since the CL5 is new, I first showed them the similarities to other Yamaha consoles and then ran them through features in the CL that they were interested in trying,” states Don Hricz, audio engineer for GV Audio.
“They were able to quickly pull a mix together; some had brought their files with them that we easily converted to the CL5,” he continues. “One of the engineers downloaded the CL5 Editor the night before his show so he became quite familiar with the console.”
Hricz said that he and the other nine-plus guest engineers were very impressed with the sound and feel of the console. “The Yamaha CL5 is my new favorite!”
DiGiCo SD8 Mixing Console Joins Inventory Of Columbian Production Company Grupo Hangar
The new SD8 was quickly put to work on Medellín’s famous Feria de las Flores (Festival of the Flowers)
A DiGiCo SD8 is the latest mixing console purchased by Colombian company Grupo Hangar, which provides technical production services to events in and around the city of Medellín.
Grupo Hangar serves live music applications, cultural festivals, sport, exhibitions, and trade and fashion shows, and the company needed a mixing console that could handle all of them.
“We purchased the SD8 for a number of reasons,” says Grupo Hangar producer Santiago Vélez. “First of all it has a wonderful sound - not only in terms of the features to shape the sound, but just by bringing the faders up it sounds amazing.
“It’s also a very versatile desk; each session can be configured to each event’s needs, which allows engineers to work very comfortably. The FPGA technology makes it more stable and faster in its internal processing, which is very important when working with lots of channels and processing. The multiband compressors are also wonderful!”
The SD8 was quickly put to work on Medellín’s famous Feria de las Flores (Festival of the Flowers). The week-long festival is the city’s highest profile event, featuring a pageant, parades and much live music.
“I have found many more things about the SD8 that I like since I began working with it,” says Vélez. “The way that the desk can be adapted to my needs is amazing - the ability to move every fader or group of faders to wherever I need them and the concept of snapshots within the same session are great.
“But most of all, I like the fact that there is no interruption to the audio when changing from one scene, preset, etc to another and the ability to change parameters softly with the programmable time feature.
“Overall, we are really pleased with the SD8. It is wonderful to use, it sounds great and there is nothing we’ve come across that we don’t like. It has been a great success, everyone who works on it falls in love with it.”
The hardware is only part of the story, though. Grupo Hangar is thousands of miles from DiGiCo’s UK headquarters, but Vélez is very pleased with the technical support he has received.
“Because we are the only company in Medellín with the SD8, I was worried that any problem with the desk could be a nightmare,” he says. “But DiGiCo has provided wonderful customer support and their staff have been very patient and helpful with any issues that I’ve had. I am very grateful for that.”
Allen & Heath Releases New Firmware For Three Xone Products
Offers more creative flexibility with an advanced channel FX configuration, and more
Allen & Heath will be releasing new firmware updates for the Xone:DB4 and Xone:DB2 digital FX mixers, and the Xone:K2 MIDI controller.
The new software provides enhanced FX sections and configuration for the DB4 and DB2 mixers, and greater integration between the DB4 and the K2 controller.
Xone:DB4 V2.0 firmware offers more creative flexibility with an advanced channel FX configuration, which allows the user to determine where the FX section, EQ section and FX ON switch are placed in the signal path – including much requested Post Fade FX.
In addition, each FX section can be configured in Classic, Send or Hybrid Mode, to create a subtly different sound for each effect type. The new firmware also introduces a powerful new effect to the DB4’s extensive FX library - Infrabass - a low frequency enhancer that generates infra bass energy from the sub-bass spectrum.
The DB4 update also includes a new spatial crossover tool to independently adjust stereo width above and below a variable crossover point.
A master output balance control for panning left and right is included, and the microphone/auxiliary input can now be routed to any or all of the four main music channels for multiple FX processing and looping. MIDI channel selection and a new MIDI clock generator enabling simple tempo changes at the twist of a knob adds a new dimension to DAW control.
The Xone:DB2 V2.0 update adds the DB4 features: spatial crossover, Infrabass effect, master MIDI clock generator, and the master output pan control. Additionally, the new firmware offers users the unique ability to cascade the two FX units for even greater creativity.
In cascade mode, the output of the X FX unit is routed through to the Y FX unit and any channel assigned to the X FX has dual processing.
Finally, the Xone:K2 V2.0 firmware and the DB4 V2.0 firmware have introduced a new integration feature. When connected via X-Link, all four of the DB4 loopers can be remotely controlled on the K2, enabling the user to mix between a loop and the un-looped track using the K2 faders, instantly changing the loop length using the momentary buttons, or changing the main loop length.
Available soon, the new software releases ensure that the award-winning DB series remains as the leader in creative DJ mixing tools and enhances the product for those who have already invested in the range.
Diverse System Serving New Reno Worship Center Features PreSonus StudioLive Console
Desk is used to mix the sound in the sanctuary, as well as sending multiple monitor mixes to the musicians, and making live recordings
The Lutheran Church of the Good Shepherd has been a fixture in downtown Reno, Nevada, since the 1950s, and over the years, the church’s congregation has grown steadily. Last year plans were drawn for a new 500-seat worship center to be constructed adjacent to the original sanctuary.
Like many modern churches, Good Shepherd has been expanding outside the traditional Lutheran model, offering a range of services to cater to a broader congregation.
“They offer a traditional Lutheran service early in the morning, followed by a more contemporary service with full band, and then one more traditional service,” says Scott Schmidt of Reno-based JC Productions.
That diversity calls for a flexible audio system that can handle everything from a straightforward organ and choir to a full-on rock band. With that in mind, Schmidt opted to install a PreSonus StudioLive 24.4.2 digital console at the front-of-house mix position.
The desk is used to mix the sound in the sanctuary, as well as sending multiple monitor mixes to the musicians, and making live recordings.
“The StudioLive is a great console for them,” explains Schmidt. “It gives them all the power and features of an expensive digital console, and it fits their budget. We didn’t have to purchase any outboard effects, which saves them money and space; they’re just using the processing that’s built into the console.”
The church’s audio crew took to the console immediately. “They recently hired a part-time technical director, just before the grand opening,” says Schmidt “He’s a young guy, very technically savvy, and of course he took to the StudioLive quickly. But even for an old analog guy like me, the console was easy to figure out.”
The StudioLive’s remote-mixing capability was one of the first priorities, followed closely by live recording. “We ordered them an iPad and a couple of Mac Minis,” says Schmidt.” They’ve set it up to record directly into Capture, and of course, the iPad allows the technical director to walk around the room and make adjustments and to walk up to the stage if need be.”
Schmidt also included some video, with a Christie digital projector feeding a DayLight 16:9 video screen behind the band. For the musicians, the back wall is equipped with four 55-inch NEC flat-panel displays, combining to create a single video wall.
“The system has performed flawlessly for them,” Schmidt concludes. “Everyone’s been very pleased with the results.”
Noise Gates 101: What They Do & How To Use Them To Their Fullest
Deployed correctly and they're great, particularly with drums. Deployed incorrectly, however, and a mess can ensue...
Noise gates, usually called just “gates,” are dynamics processors that you use when you want to automatically turn off a channel if the signal is not present.
They perform this magic by “looking” at the input signal, and if it is below a certain level, the gate is closed.
When we say that the gate is closed, what really happens is that the gain of a device in the audio path called a Voltage-Controlled Amplifier (VCA) has been set to minimum.
Because the various signals you might want to gate are all different, most gates provide a handful of control knobs that let you tailor the gate’s action to your liking.
This article will discuss what those knobs are, and how they affect the gate’s response.
I’ll assume that the gate is connected to a console’s insert loop, and that the channel’s input level has been set properly, and I’ll also use a drum as the example input, since gates are most commonly used on drums.
A typical full-featured gate will have the following controls: threshold, attack, hold, decay and range. Additionally, there may be high- and low-frequency controls.
Less expensive gates may not include one or more of these controls. There are usually LED indicators which tell whether the gate is open or closed. Some gates use an LED bar-graph display to tell you how much attenuation the gate provides.
I find it easiest to start when all of the gate’s controls are set for “no gating.”
Start with the Threshold set to Minimum (largest negative number!), and attack time at minimum, (shortest time), hold time at maximum (longest), decay time at maximum (longest) and the range to maximum (largest negative number). Set the low-frequency control to minimum, and the high-frequency control to maximum.
As the drummer slowly bangs on something, raise the threshold. At some point, the gate will close and no sound will pass through the gate.
The threshold control “looks” at (the envelope of) the input signal, and if the input signal is below the threshold, the gate is closed. The gate opens when the signal is above the threshold.
So, the trick is to find the point where the drum opens the gate, but the ambient noise on the stage, or the next drum, doesn’t. With a little practice, you’ll find it’s easy to narrow that down.
The threshold control is calibrated in decibels. 0 dB refers to the gate’s nominal input level. This means if your gate has +4 dBm I/O, then if you set the threshold to 0 dB, the gate will open when the box input exceeds +4 dBm.
If the threshold is set to -15 dB, then the gate opens when the input is above -11 dBm. Likewise, if your box has -10 dB I/O, then the 0 dB mark means -10 dB.
Next is the attack time. The attack time control sets the time (usually in microseconds, or milliseconds) the gate takes to go from closed (maximum attenuation, as set by the range control) to open (zero attenuation).
The gate doesn’t “wait” the attack time before snapping open; rather, it smoothly ramps the attenuation from max to zero in the attack time. This is analogous to starting with the channel fader on minimum, and fading up to unity, in microseconds.
If you’re playing along at home, you may notice that if the threshold is set so that it’s barely below the drum input level, the gate will “click” as it opens. You can mitigate that click with the attack time control.
Here’s what happens: when the signal goes above threshold, the gate is told to open. If the attack time is too fast, the gate output wants to switch instantly between 0V and some non-zero value—maybe a couple of volts. Now, if you use the attack time to slow down the attack, that voltage doesn’t change “instantly,” but rather smoothly ramps up.
So, the trick is to set the attack time fast enough to capture the drum’s transient, but not so fast that the gate clicks.
The hold time control is obvious—it’s simply how long (in milliseconds) the gate remains open once it’s fully open. Too short a hold time, and you clip the end of the drum’s ring. Too long, and the gate may not close, or you’ll get excessive ring, or what have you.
You can also get weird clacking noises from the gate as it chatters. Too short a hold time and you’ll find the gate might try to open and close and open and close quickly.
The decay time is the same idea as the attack time, except it determines how quickly the attenuation increases once the signal goes back below threshold. Too quick a decay time and you’ll clip off the drum sound tails. Too slow and it may not be closed before the next drum hit.
The range control is what sets how “closed” the gate is. When set to max attenuation (say, -80 dB; negative dB gain is attenuation!), when the gate is closed, there is 80 dB attenuation from the input to the output. That’s closed!
Now, as to why you would want to control the amount of attenuation: imagine putting a gate on, say, background vocals. When the person isn’t singing, the gate is closed and the background noise disappears.
When (s)he sings, the gate opens, and the background noise is present. It’s a bit odd sounding, so by setting the range to, say, -10, the difference between open and closed isn’t so startling.
It’s as if you pulled the channel fader down to -10. If you set the range to 0 dB, there is no gating action at all. Gates usually don’t have a “positive” range, since they do not add gain.
Now, for the frequency controls. These controls let you tune the frequency range the gate responds to. For example, say you’re gating the rack toms, and there’s a loud crash cymbal right above them.
With the frequency controls “wide open” (low at min, high at max), the gate may respond to the cymbals as well as the drum, which is not what you want.
So, you use the frequency controls to set the passband that the gate responds to. Thus for a rack tom, set the LF to something like 50 Hz, and the HF to something like 500 Hz (or whatever).
Since the majority of the cymbal’s energy is not in that passband, the gate won’t respond to it. But the rack tom does have a lot of energy there, and the gate opens when the tom is hit.
It is important to remember that the frequency controls do not affect the tonality of the audio! Also, some gates have slightly-different controls for setting the gate passband.
It’s similar to a parametric EQ – there is a center frequency knob and a Q control. The Q sets the width of the band centered on the center frequency. Some gates may have just the center-frequency control and a fixed Q.
Now, for some technical details: The gate is based on a device called a Voltage-Controlled Amplifier (or VCA). This is a device whose gain (and attenuation) are set by a Control Voltage. The audio signal through the VCA is usually called the audio path.
The Control Voltage is derived by something called the sidechain, which is parallel to the audio path. The gate’s input is split into two outputs; one drives the audio path and the VCA, the other drives the sidechain.
The sidechain uses a circuit called a detector to “look” at the audio and generate a DC voltage that corresponds to the level of the audio.
This voltage is modified by the threshold, attack, hold, decay and range controls to come up with the proper Control Voltage to set the VCA to do the right thing.
You should note that none of the gate’s controls affect the audio directly – they affect the control voltage to the VCA, which affects the audio level. A gate has no affect on the tonality of the audio.
Note that the frequency controls come before the detector. Therefore, they affect what the detector responds to. Also, side-chain inserts come before the detector.
The detector is probably the most important part of the gate circuitry. A poor choice of detector can make a gate built around the world’s best VCA chip a piece of junk!
A noise gate is really a special case of a device called a downward expander. (Aside: an expander affects signals both above and below a threshold; a downward expander only affects signals below a threshold.
That’s an important distinction.) Instead of having a range control, an expander has a ratio control.
Ratios are expressed as in : out, where in is the input level in dB, and out is the output level in dB. Ratios range from 1:1 (meaning no expansion) to 1:10 and higher (lots of expansion).
When the signal is below threshold on an expander, it is attenuated by an amount determined by the ratio control.
So, for example, if you set your downward expander ratio to 1:4, that means that for every 1 dB below threshold, the VCA will attenuate the signal by 4 dB.
A signal at –4 dB, then, would be attenuated by 16 dB. A gate is nothing more than a downward expander with a very high ratio, in much the same way that a limiter is a compressor with a very high ratio.
Finally, regarding gate usage: Gates are a double-edged sword, and they can be immensely frustrating devices if the drummer is either very uneven with his playing, or if he’s got an excellent sense of dynamics.
Too much gating and the good drummer’s quieter parts get clipped off (unless you’re riding them, which you have to do). Not enough gating, and the lousy tone of the bad drummer rings forever.
A typical console may have dozens, even hundreds of knobs and buttons and faders.
Each one has a specific function, but one is more important than all the rest. It’s typically at the top of the channel strips and it’s called “gain” (or sometimes “trim”).
It is perhaps the most misused and misunderstood control on the whole board. Get it set wrong and no amount of fading, EQ or outboard processing will fix it. Get it right and the rest of your mix will come together much, much easier.
So, how do you get it right? Well, it depends. (Great - thanks Mike!) Seriously, it depends—on your board. I wish I could tell you to turn the gain up until you get to 0, then you’re done. That may work, but it may not be optimal. You have to do some experimenting and listening.
Follow along and I’ll walk you through the process, then we’ll look at some specific things to listen for.
For starters, adjusting gain needs to be done in a methodical manner. If you’ve ever been to a concert early and watched a sound check, you’ve seen how it’s done. Turn all the faders down on the board, and start with the gains all the way counter-clockwise.
Start with one instrument, say the kick drum. Ask the drummer to kick, kick, kick, kick. He keeps going until you tell him to move on. If your board has a PFL or Solo button on the channel, push it for the kick drum (or whatever you’re starting with).
The PFL should route that channel to your main meters (check your manual), so keep your eye on it and gradually bring the gain up. When it starts to peak around 0, you are close. Now you can start brining up the monitors, and then house fader.
Repeat this process with all the other instruments on stage. Then move on to vocals. At the end of this exercise, you should have all the instruments and vocals hitting 0 or a little more. At this point, you may be done. Or not.
It all depends on your board. Some boards have a lot more headroom than others, and if you cap the levels at 0, you are not fully utilizing all the gain they have available, and are not maximizing your signal to noise ratio (the difference in signal level between the noise floor and the signal or music).
Other boards are pretty much spent at 0, and if you send 10-16 channels all at 0 to the main buss, it will overload and you will distortion. Or you may just be on the verge of clipping all the time.
This is where you need to listen and pay attention to your board. The console we have at our church will take +8 inputs all day long, mixed into groups and to the mains with no hints of saturation or distortion. So we can run stuff hot, and maximize our signal to noise ratio. On the other hand, I once used another board at another church that would be completely out of headroom if you ran all the inputs at 0.
Play with your board, try different levels and once you settle on a level that works, stick to it. Make it systematic so everyone uses the same gain structure. Once it’s repeatable, you’ll have better, more consistent sound every week.
Another Part Of The Picture
This weekend I was reminded of another gain setting that is just about equally important (perhaps even more so), and that would the gain on the wireless mic the pastor is using. Here’s what happened.
I have been in the process of revamping our entire wireless microphone family over last few months. The new wireless systems have been really great. I’ve also been making the switch to new mics for said wireless systems.
The challenge is that we have some speakers (“talkers”) who like the new mics, and one (turns out he’s the new Sr. Pastor) who doesn’t so much. Since he’s new, I’m cutting him some slack and letting him use a lav (for now…).
And that’s the rub. We have one body pack that is “assigned” to the speaker for the weekend. Sometimes we’ll plug in a lav, other times an 892, and each mic has a different sensitivity rating; some speakers are loud, others are quiet. If you read the first part of this article closely, you know where this is going.
Just like the input gain on your console, the bodypack also has an input gain setting (at least it should – if it doesn’t go order a new one that does). Sometimes it’s a rather coarse “0”, “-10” switch; other times it’s a little control in the battery compartment that needs a tweaker; sometimes, it’s a handy thumbwheel on the side of the transmitter.
The problem is, too often we sound engineers get so busy, we jack in a mic, drop in a battery and hand it to the speaker who is already running to the stage for a sound check. We crank up the gain on the board as he says, “Check one, two…are we done?” and hope for the best.
It’s not until he’s up on stage at the beginning of the message that you hear the familiar crackle of some sound gremlins having a bad day. You check your console gain, everything is fine; you may even check the compressor, the EQ and everything else.
Check the wireless receiver. If it’s a good one, it will have an audio level meter. A less good one will have a clip light. If you see clipping, or the meter is maxed out, you’re in a world of hurt. You’ve gone and done it – you’ve used up all the headroom in that little bodypack.
And it’s not like you can run up on stage during the message, reach into the pastor’s back pocked, grab the mic and tweak the little dial down a bit. Oh no, you’re hosed.
Something I’m trying to get my engineers to be more cognizant of is the wireless mic gain. We used to put the mics in a tray and put them in the green room for the “on stage” folks to just pick up. Now, we’re keeping them at the house console.
That way, we can help them get the mic fitted properly, show them how to use it if it’s new to them and most importantly, adjust the gain on the pack before they’re 100 feet away on the stage (and while we can lay eyes on the receiver so we know what we’re doing!).
So here’s my procedure (which will soon become the law of the land at the church). Speakers and actors must pick up the mic at the sound board. Before they will strap on said mic while standing there and give us a realistic level while we adjust the gain on the pack.
They will then proceed to the stage at the appointed time for sound check and we’ll do the gain trimming and level adjusting for the house (and monitors if necessary).
At the end of the service, the mics will be delivered back to the sound board so that batteries can be recharged and so we don’t have to chase people all over the church looking for them.
Yep, that input gain control is the most important setting, whether it’s on the bodypack or the console. Getting this right just makes your day go so much easier. Get it wrong and you’ll hear, “Why was Jack all crackly and distorted for the whole message – it was really distracting!”
And that, my friends, is not good sound (apologies to Alton Brown).
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Mackie Updates Master Fader App (v1.2) For DL1608 Digital Live Mixer
Free and can be downloaded and installed directly to an iPad via the App Store
Mackie has just launched an update to the Master Fader control app for the DL1608 16-channel digital live sound mixer with iPad control.
Master Fader v1.2 is free and can be downloaded and installed directly to an iPad via the App Store (here). It can also be download via Mac or PC using iTunes.
The DL1608 offers 16 Onyx mic preamps and the performance of 24-bit Cirrus Logic AD/DA converters. Seamless wired to wireless iPad control allows a user to mix from anywhere in the venue, providing the mobile freedom to control not only the mix, but also plug-ins like EQ, dynamics, effects and more.
The DL1608 also supports up to 10 iPad devices so musicians can control their own monitor mix on stage.
Does your mix suffer from these four common mix maladies? If it does, that’s OK because there is a remedy for each one. The next worship service could be your best sounding mix.
There are different mixes a sound tech must create depending on the congregational preferences and pastoral requirements.
Using an example with a very wide mix difference, a mix for a “hip” youth group will be much different than a mix for a church with an older congregation and traditional music. This is simply part of the job. It doesn’t mean one mix is better or worse as they each meet the existing needs.
Regardless of the mix requirements, there are four signs your mix is suffering. The biggest problem in recognizing these signs is getting over your own bias for your mix. It’s your mix, so it’s perfect, right? I thought I used to get a great snare drum sound until another sound tech showed me what I was missing.
Before you ignore these signs, try the fixes during your next sound check and listen to the difference.
The Four Signs Of A Mediocre Audio Mix
1. Mix lacks low end emphasis and energy
The bass, the lower-end drum toms and the kick drum play a huge role in filling in the low-end sound while also giving the music the right amount of energy. A mix that doesn’t have these properly pulled in will get you an overall sound that lacks energy and feeling. Or, as I like to call it, vibe.
Start by adding “too much” of the kick drum into the mix. Once you find it overpowering the overall mix, start cutting back the volume. If you have an electric drum kit all on one channel, use the low-end EQ to control the kick’s presence in the mix. Listen for a spot where the kick drum gets you the right vibe. I was at a church where I could set the kick volume by my ability to feel it in the floor of the sound booth.
Use the above for all the low-end instruments. Traditionally, you bring up the volume until it’s where you want. However, you’ll find you can often find a better volume spot by pushing the channel much hotter and then pulling back.
2. All channels have the same volume level in the mix
A mediocre mix is easy to spot when all the instruments and the vocals sound like they are at the same volume level. The mix lacks depth. There is no subtly to any part of the mix. There is no leading instrument or vocal.
Honestly, I’m not sure why this problem is so prevalent but I have an idea. During the process of setting the gain structure, most of us get the volumes in the same range on the mixer meter. At this point, you should then set all the volumes in the right relationships to each other. It seems, the problem occurs when that last part isn’t done.
Start your next sound check by bringing in the drums to the level that fits the room, then bring in each instrument that’s higher in frequency. Then bring in the vocals until you end with the lead vocalist which should be on top of the mix. You can check out these articles for more information on volume balancing:
3. Instruments and vocals lack clarity and distinction
Each instrument and each vocal needs to fit in the mix so the best qualities of each are present in the mix.
It’s like my grandmother’s cooking. She had all the right ingredients but she couldn’t season a dish to save her life.
The EQ process is the seasoning. I’ll be honest, I used to sneak in a bottle of hot sauce on the days that she made her chili.
Let’s say you have two singers, an acoustic guitar, an electric guitar, a bass, and a drum kit. The default “noon” position on the channel EQs isn’t going to help your mix. Decide what you want your mix to sound like and then start making EQ changes.
For instance, if you want the acoustic guitar to have a bright feel, then cut a bit of the low’s, boost a bit of the highs, and then work out the mid-range frequencies to give you the right amount of presence and body from the guitar.
Creating clarity in any one instrument is more than this article could cover so consider these articles;
I hear this when a mix is lacking in the higher and lower frequencies. The first sign mentioned was surrounding the low frequencies. Let’s look at how a mix might not have enough high frequencies.
High frequencies can come through high vocals, bright sounds like high strings on a guitar, and cymbals, just to name a few. This isn’t to say you have to push high frequencies just for the sake of filling in frequencies in the high end.
Consider it like this; mute the cymbals in your mix. How does that sound? Turn them up to the right volume. How does that sound? If you have a lively, energetic song, you’ll likely want a nice bright sounding mix. Push up the highs a little on the cymbals. Now, how does the whole mix sound?
Listen to your whole mix and listen for frequency holes where you can fill in the frequency It’s like an example I’ve used before—you’re painting a picture with music. You need to fill the whole canvas with color. You can use pastels, earth-tones, whatever color selections you like.
But you can’t paint a picture with all red tones or all blue tones. A picture painted with many similar colors gets you a picture that’s hard to interpret. It’s a cow, no, it’s a house, no, it’s a VW microbus.
By using a wide range of colors (frequencies), you can create a variety of beautiful musical paintings with depth and feel and emotion.
The Take Away
The best way for quickly improving your mix is listening to it objectively and comparing it to the above four signs. Mixing is a wonderful creative process, but it’s also a process that takes time and skill and patience and evaluation.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
Historic Princess Theater Goes Multipurpose With Allen & Heath GLD Mixing System
Provides mixing capabilities and more for a wide range of applications at revitalized theater
Born as a 1920s motion picture theater, the historic Princess Theater in Harriman, TN reopened in March, 2012 as a multi-purpose performing arts and education center.
Now restored to its art deco grandeur, the Princess hosts concerts, movies, dance performances, community and educational events and local church services.
The Princess benefits from a close association with Roane State Community College which operates the theater and Channel 15 Television which has its headquarters next door. As part of the renovation, Roane State equipped the Princess with a state-of-the-art sound system including an Allen & Heath GLD digital mixing system.
“Roane State decided to do it right from the start,” states Megan Anderson, who manages the theater for Roane State. “We wanted a high-performance sound system that was versatile enough to serve the needs of all of the groups performing at the Princess yet it needed to be easy to learn and operate for our educational programs. The GLD is great for all of this.”
Jon Chemay, Roane State theater technician, operates the GLD at most events. He appreciates the GLD’s versatile EQ and the color-coded custom channel labeling.
“Most events do a sound check,” says Chemay, “but the GLD can be set up quickly for those groups that come in at the last minute. I’ve experienced digital mixers that seemed like they had hundreds of confusing buttons and knobs. The GLD feels like analog to me. Easy to learn and use yet very versatile.”
A local church holds services in the Princess each Sunday. Chemay has set up a “scene” on the GLD which recalls all of the settings and labels for the church including its praise band, which allows the church sound tech to operate the system without affecting the settings for other events.
The sound system was installed and commissioned by Allied Sound of Nashville, with Steve Stanford of Allied noting that the GLD’s main input/output rack, the AR2412, is located near the stage along with a pair of AR84 expansion racks. This allows microphone snake boxes to be positioned where needed on the stage for different events.
Stanford also points out the CAT5e connection from the AR2412 and AR84s to the GLD-80 console. “In an old facility like this,” Stanford says, “it’s generally very difficult to route a large, multichannel analog snake from the stage to the mix position. The GLD’s built-in digital snake made this easy.”
Stanford adds that the Princess has another AR84 I/O expansion rack at the mix position for wireless mic receivers and other sources brought in for a specific event.
You may nod your head in agreement as you read it, but when it comes to working in the studio, it may be obvious that you’re not completely on board.
(And I’m no exception. I have to constantly remind myself as well.)
Ready? Here it is:
The key to a full mix lies in the RECORDING phase, not in the MIXING phase.
Let me flesh that out a bit.
Probably 70 percent of the questions I get from readers have to do with mixing. That’s understandable. Mixing is difficult.
BUT…we tend to make mixing MORE difficult by how we record our tracks. We race through the recording phase, and then wonder why our mixes don’t sound “full.”
What I’ve found is this — if my mix doesn’t sound full, it’s because of one of two things:
1) I didn’t record enough tracks.
2) I didn’t record the RIGHT parts.
You’ll notice that neither of those have anything to do with my mixing skills.
Take-away point: the better your recordings sound, the better your mixes sound. Boom. Period.
I know, I know. It’s easier said than done.
But I’ve got a “secret weapon” that I’ll use if a particular song isn’t sounding full enough.
It’s not a huge wall of guitars. It’s not a bunch of synth pads. It’s not even a great B3 part (although I LOVE to use B3.)
It’s something that works on nearly any type of song. Big huge rock tune? Yep. Intimate ballad? Yep.
What is it?
(Or BGV’s as we like to call ‘em.)
There’s nothing quite like a big, huge, lush BGV part to fill out a chorus, or draw the listeners’ attention during the bridge, or make the turn between the chorus and the verse interesting.
Over the years, I’ve recorded HUNDREDS of BGV tracks. And I’ve developed a pretty cool system for getting great-sounding BGV tracks.
There’s a method to my madness.
Next week, I’ll be doing a live training for VIP members. I’ll cover how I go about writing, recording, and mixing BGV’s.
To become a member (and get access to this video and over a dozen other in-depth training videos), go here.
P.S. You don’t have to be a singer to use this tactic. In fact, if you can learn the ropes of getting a great BGV part, your vocalist clients will LOVE you. (A lot of them want to record big BGV parts, but they just don’t know how.) You could be their hero.
Joe Gilder is a Nashville based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.
Roland Systems Group Announces Interactive V-Mixing Training Coming To Atlanta
Hands-on and interactive approach is designed for mix engineers, church techs, consultants
Roland Systems Group has announced an Interactive V-Mixing Training course that is coming to Atlanta on October 23, 2012 from 1 pm to 9 pm.
This new hands-on and interactive approach designed by the team at Roland Systems Group is aimed to equip engineers, church techs, consultants and re-sellers with complete information on the V-Mixing System solutions along with the latest information on updates, iPad integration and new workflow ideas for improving rehearsals, sound quality, performances and overall production.
Attendees will experience interactive sessions that include a hands-on component plus mixing audio and interacting with a live band.
John Broadhead, VP of Technology and Communication for Roland Systems Group who helped designed the session, explains, “Although classroom style training is important, this particular training enables the participants to get their hands-on the product in the context of a real venue alongside a live band – real-world, very practical and effective.”
The training will take place at The Sanctuary Church located in the Atlanta area, in Kennesaw, GA. This unique space is designed with the stage located in the center of the room and side fill screens for IMAG and presentation slides.
Hands on sessions in the afternoon will include live tracks so attendees can walk-through the process of setting up channels, personal monitors and tuning a mix. The evening session adds a live band in both a rehearsal and performance context.
In addition to practical advice on system design and configuration, attendees will experience first hand how to make rehearsals and training more effective with integrated instant playback, virtual rehearsal and recording.
Attendees will also get the latest in personal mixing techniques using the Roland M-48 solution to help tighten band performances and increase musician satisfaction. Also included is a video portion showing how to switch and webcast an event in HD or SD.
Registration is free but space is limited. Go here for more information and to register.
Mixing vocals to pre-mixed instrumental is an issue that just about every engineer will encounter.
The idea of recording vocals to a mixed down instrumental has been around for decades. On rare occasions that vocal and instrumental would be released as the actual song.
In Hip Hop, this is fairly common, and comes with the intention of the combined vocal and instrumental being the final release. Because Hip Hop producers mix their tracks with this intention, there’s a unique set of skills that comes into play for throwing vocals into the mix.
A pre-mixed instrumental is probably somewhat compressed (or, sometimes highly compressed). It’s been “maximized” to play loudly.
The challenge here is that loudness isn’t as simple as turning the level up. After a certain point, loudness can only be achieved by increasing the density of the record, rather than the volume. The issue is that if the track is exceedingly dense, there is no room for the vocal.
In fact, the first thing one has to do is turn the instrumental down in volume significantly in order to make anything work. By this token – the louder the instrumental – the quieter the end mix. Ironic, no?
We need to approach this mix with the forethought of compensating for whatever quality the instrumental is coming in as – highly compressed, mp3, possibly in mono, the list goes on.
But first thing’s first. Get the vocals sounding the best you can on their own. There are some great tutorials for this here on this site. I’d start with Mixing Rap Vocals Part 1. I also recommend checking out Ken Lewis’s video.
Since the information on vocal treatment is covered elsewhere, I’m going to focus on working the vocals with the instrumental, and treating the instrumental.
The goal is to get the vocals to sound like they belong with the instrumental as much as possible. So we need a two pronged approach:
—Open up the instrumental to allow for the vocals to fit in, and;
—Match the vocal to the instrumental.
And we need to do this as non-destructively as possible.
Opening Up The Instrumental
To open up the instrumental, try to do as little EQ as possible. EQ is the method most people seem to recommend, but EQ is used to correct tonality issues.
If the instrumental is too dense, that’s really more of a dynamics issue. Turn the instrumental down, and use an expander or transient designer to rebuild the peaks of the drums. It can take a while to get used to using an expander—3 to 6 dB of dynamic range adjustment is usually a good start.
If the record is clipped, you may need to use something like iZotope RX to get your peaks back.
Sometimes an instrumental is so compressed, there’s no transient at all for an expander/transient designer to grab. In this case, you may need to get inventive. Often times, I’ll find drums that match or compliment the drums in the record. I’ll lay my own drums into the track. If you can’t make it, fake it.
Another place you can get a little dynamic is in the “side” information. M-S processing can allow you to separate the sound that lives in the center from the sound that lives on the sides. Pulling the center down but keeping the sides up a bit can help in creating a sense of dynamics, and a “pocket” for the vocal to live in.
If there’s anything that really steps on the vocals, notch it out a hair. Subtlety is the goal here. It’s impossible not to make compromises when you are EQ’ing the entire instrumental.
Once the vocal is in a good place on it’s own, there comes the game of “matching” it to the instrumental. The idea is to make the vocal sound like it belongs with the instrumental.
For example, the instrumental may be highly compressed, and your vocal may be very dynamic. The result here is that the vocal will always either sound like it is buried by the instrumental, or so loud that it feels like it’s hanging over the mix.
Neither of these are great, but if I were to err, I’d lean toward the side of the vocal being over the mix. You can minimize this effect by compressing the vocal until it starts to compete with the density of the instrumental.
This takes a lot of tact, and often requires a number of compressors working in tandem in order to reduce the negative artifacts of over compressing a voice. If you are careful in your approach you can get the vocals pretty darn compressed without them sounding “bad.” But it’s a negotiation.
The instrumental may be very bright or very dark. You may want to lean your vocal EQ to compliment this. A bright vocal on a dark track will sound a little unglued.
This goes for the rest of the frequency spectrum, and overall tonal qualities like distortion. If the track has a distorted or saturated quality to it, you may want to purposefully distort the vocals. Obviously don’t do it if it’s too detrimental, but it’s something to consider.
Lastly, match the space. If you have a mid-side processor, solo the side signal, or turn the center signal way down. This will expose the quality of the reverbs used in the instrumental.
Finding a similar reverb/delay, and EQ’ing the reverb to match the tonality of ambience in the instrumental will help bond the vocal and instrumental in the most transparent way. This takes a while to get a feel for, and is somewhat subject, but it’s highly effective.
Matthew Weiss is the head engineer for Studio E, located in Philadelphia. Recent credits include Ronnie Spector, Uri Caine, Royce Da 5’9” and Philadelphia Slick.
Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.
Dogwood Audio Supplies Soundcraft Vi1 Console For U.S. Tour By Celtic Rock Band IONA
Dogwood also supplied the system for the band's previous U.S. tour in 2010
U.K.-based Celtic rock band IONA recently finished a successful tour throughout the United States while deploying a Soundcraft Vi1 as its front-of-house console, supplied by sound provider Dogwood Audio of Chicopee, MA.
Dogwood also supplied the system for the band’s previous U.S. tour in 2010, and based on that successful effort, the company was asked to do it again for this tour. In addition, Dogwood owner Mark Jerrell was also brought on board as the audio engineer
After a conversation with Dave Bainbridge, IONA’s lead guitarist, the two decided the Soundcraft Vi1 console would provide them with the most flexibility.
“There are many challenges of traveling during a 2-week tour,” says Jarrell. “The stretches between gigs are not long and there isn’t much time to set up at the various venues. I have to be able to have everything up and running in no time and with the ability to recall the presets with the Vi console, I never have a problem.”
Another challenge for Jarrell is the varying size, acoustics and equipment of each venue. “We see everything from a large theater to a small club,” he notes. “With the Vi1, all I have to do is adjust the EQ in the different venues—it’s straightforward. What I love is that once I adjust the settings for each venue, I can then save it as a file for when we come back.”
The custom channel layers also help Jarrell to have a smooth show, as he is able to arrange them to suit his needs. “The desk is a lot of fun to mix on,” Jarrell states. “It’s such a warm sounding console! I use the Lexicon effects pretty heavily, the tap delay, reverb and more. It has the feeling of a nice analog desk; making an adjustment is always easy to do.”
With such a dynamic group, Jarrell appreciates the flexibility the Vi1 offers. “IONA likes to change up the set list sometimes. The road manager is also a percussionist so every once in a while they call him up to play along. I am able to easily control each microphone individually and it is really easy to adjust gain and the high pass filter if needed.”
“Overall, it has been a pleasure to tour with IONA and a lot of fun using the Soundcraft console,” Jarrell concludes. “I can’t wait until the next U.S. tour comes around. I already have all my presets and files saved so I’ll be ready.”
There I was, on Christmas Eve Eve, listening to the rehearsal tracks for our Christmas Eve service, tweaking the mix and thinking, “Hmmm. Something is missing.”
In fact, it wasn’t that something was missing; it was a big band, with 11 vocals. If anything was missing, it was space. With all those instruments and vocals packed in there, the mix was sounding a bit dense.
Whenever I get that many vocals on stage, I’ll start panning them left and right. I typically break up each group—tenors, altos and sopranos—to left, just off center and right.
That is, I’ll have one tenor, one alto and one soprano panned hard left, another from each group hard right and a third from each group just off center. I’ll split the off-center ones on either side of the lead vocal. That alone does wonders for spreading out a dense vocal mix.
And that was a good start, but I wanted a bit more. I had a lot of mid-frequency build up in the center with two electric guitars, sax, percussion, drums and B3/keys. The piano and keys were already stereo, so I panned sax, percussion and electric around a bit to open up the mix. I can’t go too far with any of those since we don’t have a really good stereo PA, but it does help open up some space.
That left the B3. I’m a big B3 lover, and I like it to be very present in the mix. But it was competing with the vocals. We mic our B3 top and bottom with AKG 414s. It’s not my ideal set up, but it’s what we have right now. I normally pan the top mic to 30-40 percent left and the bottom to 30-40 percent right. That helps, but it wasn’t really giving me what I was after.
Then I thought about some recording studio tricks of double tracking an instrument and offsetting one track by just a skosh. On the SD8 (and almost every other digital board), that’s pretty easy to do, so I gave it a try.
I double patched my B3 High mic into two mono channels, one panned hard left, one panned hard right. I turned on channel delay on one of those channels and spun the knob. Wow! It didn’t take long for the sound to almost explode out of the loudspeakers. It went from this very tight focused sound to a huge sound field in no time.
I played around with different delays and settled on about 5 msec for one channel. In our PA, adding more didn’t really help a lot, and less wasn’t wide enough for me. Your mileage may vary.
Once the whole band got going, the B3 sound was amazing. Instead of being in the middle, fighting for spectral room, it seemed to be coming from the sides of the room, filling in all around the rest of the mix.
I was able to keep the level up higher than usual without any competition for the vocals. Kevin Sanchez, one of my FOH engineers stopped by for one service, and without knowing that I had done anything said, “What did you do to the B3? It sounds amazing!”
To give you an idea what I’m talking about, I’ve prepared three sample tracks. The first is just the standard, Mono B3 high mic from this weekend. The second is a recreation of what I did on the SD8; the same track, double patched, panned hard left and right with the right side delayed by about 5 msec. The third track is an intercut of the two.
I recommend you listen with headphones or loudspeakers set wide enough apart to actually impart some sense of stereo sound.
This technique could also be useful on guitars, or mono keyboard sends. It would also be really interesting to try this on a double mic’d piano (low and high), processing both the low and high separately.
Now that we’re running ivory as our standard piano, I can’t try that, but next time we bust out the mics, I’ll give it a shot.
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
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