Thursday, October 10, 2013
Making It Sing, Part Deux: Further Enhancements To A Vocal Plug-In Chain
Optimizing the key to "the best-sounding band I've ever heard"
Greg Price got the ball rolling last time (here) with the first in a series of articles sharing our knowledge of the vocal chain. It cannot be re-emphasized enough that the vocal is the most important part of your mix, and therefore should be the first focus of any good mix.
When I was a young engineer, I asked my friends who were not musicians or mixers what they liked (or disliked) about my mixes. Without fail, comments were about the vocal. Obviously there are elements of a mix that must be there in order to support the vocal, but ultimately vocals are the key to comments like “that’s the best sounding band I’ve ever heard.”
A very prominent recording engineer/producer once told me that he views a mix as a beautiful plant. The vocals are the flowers that everyone gazes at in admiration, but without a solid root system and stem (drums, bass, guitars), the flowers would not be possible.
Both Greg and I use Waves Audio plug-ins to help achieve excellence in our mixes, but please note that we’re not suggesting using all of these tools all at once. Used sparingly, however, they are akin to bringing a gun to a knife fight.
Previously, Greg concluded at part 4 by discussing the C6 Multi-Band Compression plug-in, so let’s pick it up from there. (See parts 1-4 here.)
4. Multi-Band Comp, Continued
I’ve developed a few presets for the C6 that I believe are a great starting point. Give them a try. Again, they’re intended as starting points, not as “be all, end all.”
The specific vocal preset that Greg mentioned was created out of necessity, for my own needs. I wanted to have a preset where the threshold, attack, release, and ratio parts of the plug-in were optimized for vocal. Basically the goal was to be able to properly gain up any vocal, with the preset as a great starting point.
I experimented for over a month with different bands to come up with it. The floating, additional two outside filters are intended to adjust the frequencies that are problem areas, but the inside four filters should work pretty well as they’re set up.
With the advent of in-ear monitors, I’ve found that singers have become more dynamic over the years. This can be good and bad. Dynamics are a very important part of music, and absolutely are what make a mix stand out. But in the case of vocals, the goal is for every single nuance of the singer to be heard over the top of an often complicated mix.
MaxxVolume (click to enlarge)
One of the tools that I don’t go anywhere without is MaxxVolume. It’s an expander, compressor, limiter, and gate. Prior to using this plug-in, I’ve worked with singers where I had 20 dB of fader swing and chasing going on. If set properly, the MaxxVolume sits my vocal squarely in the mix, and with small fader moves keeping it there, allowing me to focus on other elements of my mix instead of chasing the vocals all night long.
I use the Vocal Absolute Level preset as a starting point with MaxxVolume. The Low Level side of the plug is the “expander” side of the plug, the High Level side of the plug is the “compressor,” the Leveler is the “limiter,” and Gate is the “gate.”
Don’t forget to toggle the loud and soft button above the metering, which adjusts release times of the compressor. Set thresholds for compression and expansion, and set the limiter to engage when the singer is over the top. It will take some experimentation on your part to get the desired result, but you’ll know it when you find the sweet spot.
Finally, there is the gate feature. Wait – a gate on the vocal? Trust me, this is an amazing tool. Have you ever solo’d up a lead vocal and noticed that along with the vocal is a lot of extraneous stage noise? Especially when the singer is running across the stage in front of the amp line, etc.?
This gate works very nicely in eliminating a lot of this noise, and can clean up the mix significantly. Give it a try and listen to your mixes with it engaged versus not, and I think you’ll be pleasantly surprised.
6. Vocal Rider
Most compressors change in timbre the harder they’re hit. More gain on the input side, and most compressors have a “pumping” or a “squished” sound to them. This is timbre change, and for some applications, that’s what we’re shooting for.
But personally, I don’t like my vocal to sound compressed. I want it to sit hot in my mix, but when the singer screams, I don’t want it to sound like every piece of electronics is about to explode.
Vocal Rider does compression and expansion without timbre change, which can be an incredible tool if that’s what you’re seeking. Vocal Rider Live has some extra tools that really help live engineers.
Two knobs are added at the top: Music and Spill. It’s a “look ahead” plug-in, meaning that it actually looks at a signal and can differentiate between a vocal and unwanted ambient stage noise, thereby providing better tracking of the compression and expansion.
Vocal Rider (click to enlarge)
By adjusting the Spill and Vocal Sensitivity knobs, you can get a really nice result of compression and expansion only when the vocal is occurring. The Music Sensitivity knob is an additional tool that allows use of the side chain of the plug-in. If you feed a bus of everything else but the vocal into the side chain, the plug-in actually takes this into account in deciding what to compress and expand.
7. Vocal Group/Bus Compression
Bus compression can be tricky. As the mix fluctuates dynamically, you can be hitting an inserted bus compressor very hard and really destroy the mix. Besides parallel compression (which Greg will cover in the next article), I use bus compression sparingly.
Earlier I noted the timbre change that can occur with some types of compression. Often that’s what I’m looking for with bus compression. Waves has modeled many amazing compressors over the years and has developed some software that really captures the essence of the original piece of gear. Plug-in modeling is an art form. Taking an analog piece of gear and creating a software version that looks, acts, and most importantly sounds, like the original piece of gear is a very hard thing to do.
SSL G-Channel (click to enlarge)
One example of this is the SSL Comp plug-in. In the late 1970’s, Solid State Logic developed large-format mixing consoles that every engineer of the time used (or wanted to use). Almost every favorite record in my collection was mixed on an SSL E Series or G Series SSL console, and the latter included a bus compressor in the master section that became the sound that we all used while mixing records. So any time I need bus compression this is my go-to plug-in.
Touching on presets again – you’ll notice that this plug-in also includes presets from some of the most respected engineers in the business. Who wouldn’t want the settings that Steve Lillywhite uses? I start here and adjust.
Finally, on my mix bus, I deploy the L2-Ultramaximizer. It’s the sound of today. Every major recording in the last 10 years has utilized the L2. It has brick-wall limiting in a way that no other limiter can do.
L2-Ultramaximizer (click to enlarge)
My mixes sound bigger and fatter while still able to play within the lines of sound pressure levels that are increasingly becoming legislated.
One other thing to point out that many overlook – every Waves plug-in includes a question mark on the right side. Click it and you go directly to the individual manual. It’s a very handy tool as you start to become acquainted with the plug-ins.
Next time Greg and I will talk about vocal EFX and offer further info as well as a summary of this series.
Ken Van Druten is at “Pooch’s Corner” and Greg Price is at “Greg’s List” at www.waveslive.com.
Fitz And The Tantrums Monitor Engineer Chooses Soundcraft Vi1 Console
Aaron Glas utilizing cue/snapshot feature of the Vi1
Indie pop band Fitz and the Tantrums has recently enjoyed a successful string of live concerts throughout the United States, either headlining its own bills or opening for Bruno Mars for several dates. For every show, monitor engineer Aaron Glas has relied on Soundcraft Vi1 as his monitor console.
Glas is no stranger to the Vi1 and the Soundcraft Vi family. “A few years ago, I was looking for a small-format digital console that could handle 24 outputs and there aren’t many,” he says. “The Vi1 looked perfect. I’ve toured with it for several years now and I’m thrilled to continue mixing on the Vi1 for Fitz and the Tantrums.”
Although Glas has a wealth of experience working with the Vi1 and other Soundcraft Vi Series consoles, his work with Fitz and the Tantrums marks the first instance in which he has used the cue/snapshot feature of the Vi1, something he has found to be an advantage.
“Using the snapshots with the Vi1 has been a great learning experience,” Glas notes. “We’ll always have the full band at sound check and it’s nice that I can recall what we’ve done the previous night and tweak the mixes based on the band’s requests. With the snapshot feature, I’m able to fine-tune the sound more specifically with each successive performance.”
In addition, the snapshot feature enables Glas to quickly and easily adjust to any changes Fitz and the Tantrums make from show to show. “My cues can change with the set list and all the levels can be recalled nightly so it makes for a pretty consistent performance each time,” he said. “The band has a great comfort level knowing they can achieve the same quality audio night in and night out.”
“The level of support from Soundcraft has also been exceptional,” Glas concludes. “Whether it’s Tom Der or Rick Morris, they’re always very helpful and very responsive.”
Wednesday, October 09, 2013
In The Studio: How to Effectively Build A Song’s Groove
Building around the interplay of a number of instruments
The groove is the pulse of the song and every song has one, regards of the genre of music. The stronger the groove, the more people relate to the song, so it’s really important to make sure that the groove is emphasized during a mix. Here’s an excerpt from the 3rd edition of The Mixing Engineer’s Handbook that shares some tips on how to find the groove and make it strong in your mix.
While it’s true that sometimes the groove may be the result of a single instrumental performance, usually it’s built around the interplay of a number of instruments, especially in complex mixes with a lot of tracks.
Normally the groove of the song is provided by the bass and drums, but it’s important to determine if another instrument like a rhythm guitar, keyboard, loop or percussion is an integral part that makes up the pulse of the song.
Usually this can be easily identified as an instrument that’s playing either the same rhythmic figure as the bass and drums, or a multiple of the rhythm, like double time or half time.
After those additional rhythmic elements are discovered, here’s one way to build the groove:
1. Find the instrument that provides the basic pulse of the song (like the drums).
2. Add the lowest frequency instrument that’s playing the same or similar rhythmic figure (usually the bass).
3. Add any additional instruments playing the same or similar rhythmic figure in order of frequency from low to high.
4. Add any instrument playing a similar rhythmic figure, like half or double time.
5. Add any instrument working as the rhythm arrangement element (remember the section about arrangement elements in Chapter 5?) and providing motion to the song (like a shaker or tambourine).
The groove may be attributed to only a single instrument, like in the case of a power trio (guitar, bass and drums) to three or even four instruments on rare occasions. If you’re not sure, the best way to determine what’s playing the groove is to try mixing in different combinations of instruments along with the rhythm section to see if the pulse gets stronger, weaker, or stay’s the same.
Tip: If a new instrument adds to the pulse of the song and the pulse seems lessened if it’s muted, then you have an instrument that’s a big part of the groove.”
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blogs. Get the third edition of The Mixing Engineer’s Handbook here.
Tuesday, October 08, 2013
Antelope Audio Clocking Plays Key Role In Systems For Rihanna “Diamonds World Tour”
Clocking for playback for synchronization, lighting, MIDI, sound effects and vocals
Already well into its U.S. leg, Rihanna’s “Diamonds World Tour” features several performances in North America, Europe, Asia, Africa and Australia, and at the core of the production system is Antelope Audio, whose products facilitate clocking for the entire rig, including playback for synchronization, lighting, MIDI, sound effects and vocals.
“This is the best sounding tour I’ve ever been a part of and we’ve been running really hard,” says Demetrius Henry, playback engineer for the Diamonds World Tour. “We didn’t think we could get our playback rig to sound any better, but the Orion32 interface has taken things to another level —the difference was like night and day.”
Currently, the tour is running two Antelope Audio Orion32s and a 10M atomic clock on playback, in addition to a Trinity | 10M combination at front of house (FOH). The Trinity | 10M combination is a favored selection among top mastering engineers, and serves as the primary clocking duo for the entire production.
In addition to playback and FOH, there are two Antelope Audio Zodiac+ converters being used in conjunction with the keyboard rig on stage, providing artifact free, high-resolution audio for the duration of each performance.
The playback rig for the Diamonds World Tour includes two Antelope Audio Orion32 interfaces and a 10M atomic clock.
Kenny Scharetts, keyboard technician for the Diamonds World Tour, appreciates the piece of mind that comes with using the Zodiacs: “We have the Zodiacs racked up directly beneath the keyboards and they are so compact and rock solid. When I turn them on, I know they are going to be there for me — it is a dream to have this kind of stability and reliability.”
Before hitting the road, the production crew put all of the equipment through its paces with several rehearsals, including two full dress rehearsals in Buffalo with a full PA before hitting the road.
At front of house on Rihanna’s Diamonds World Tour, an Antelope Audio Trinity | 10M handles clocking for the overall production.
“Every department was able to get a full testing of their gear to make sure it was show ready,” says Kyle Hamilton, FOH engineer for the Diamonds World Tour. “From the beginning, the Antelope gear has been running smoothly.”
The tour is large-scle in both musical content and technical scope. With a runtime of 97 minutes and a total of 36 songs, including interludes, the production team is running about 115 inputs in total.
With such an extensive rig, the production team decided to take a holistic approach and clock the entire rig using Antelope, in addition to using its premium quality converters across nearly every part of the audio system.
“Sonic changes don’t just occur at the console,” observes Hamilton. “It also has to do with the converters that are used throughout the system, in addition to the clocking. Also, it would not make any sense to have just one aspect of the system clocked — clocking the entire rig with Antelope gives us a consistently high quality result.”
An Antelope Audio Zodiac converter sits beneath the keyboard rig onstage.
On the Diamonds World Tour, the production team is focused on not just quality, but efficiency too. “Rack space is a precious commodity out here on the road — it’s like real estate: you only have so much land.” says Henry. “The fact that we are able to fit 32 channels of I/O on the Orion32 is a dream come true.” At playback, two Antelope Orion32s and a 10M are run through a pair of Apple MacBook Pros, which Henry says are lightning fast.
With a rock solid production team and reliable, great sounding equipment in place, FOH engineer Kyle Hamilton couldn’t be happier with the results they are achieving night after night. “It is immensely satisfying for us to be setting new standards in the quality of our live sound productions,” he says. “Every time we plug in another Antelope device, everything goes up a notch.”
Church Sound: Finally Mixing on a Digital Board? Escape These Three Traps
With great mixing power comes great mixing responsibility
You hear talk of people transitioning into the wonderful world of digital mixing, but you never hear of what happens after they make the transition.
Old habits must be broken, a new way of thinking about workflow has to occur, and digital mixing doesn’t mean you can finally perfect a vocalist’s mix…at least not for two weekends in a row.
The Three Traps Of Transitioning From Analog To Digital Mixing
1. Don’t assume last week’s settings are perfect for this week.
Digital mixers give you a massive amount of EQ control over each input. While I’m grateful for graphical EQs, it’s easy to set them “perfectly” for each musician one weekend and think the next week those settings will still be “perfect.”
Week-to-week, a lot of factors change. Guitarists use different guitars, different pedals, and different effects, all according to the song arrangement. Oh yeah, and then what works for one arrangement doesn’t work for another.
I’m not against using the previous week’s EQ settings as a basis for the mix, but don’t assume it doesn’t have to change.
2. Before setting your gains and faders, check your group levels.
This one still gets me from time-to-time. Coming from an analog work, it’s easy to look at your board and know exactly where your group volumes are set; these could be your groups, DCAs, “SUBs”, whatever your board uses and calls them.
In the digital world, where some mixers work as a “surface” where the faders represent whatever channels you have selected, you might not see your group level fader settings unless you select the mixer’s surface to show them. Yamaha M7CL users know what I mean with the DCA button.
3. What you see is not always what you get.
In the analog work, you can look down at the mixer and see all of your settings (rack unit settings excluded.) In the digital work, what you see in front of you is only a small representation of what’s actually set for a channel. And in some cases, the digital screen before you might not be the same as the channel which you are focused.
For instance, on the M7CL, there is a bank of faders and controls on the mixer which are tied to the bank of channels displayed on the screen. If you change the view to a different bank, say channels 1-8, but think you are on channels 10-16…you are changing the wrong channel. I saw this happen to a tech when he thought he had un-muted a microphone but was working with a different bank of channels.
Additionally, when a problem occurs during a service, such as with a microphone channel, using an analog board, you can scan over the whole board and spot an incorrect knob setting. With digital mixers, you don’t get that ability. Look to the channel you believe to be the problem and make sure you select that channel so your display settings are for that channel.
The Take Away
With great mixing power comes great mixing responsibility. Digital mixers give you a lot of control but to be used effectively, you must know how to use them…and never assume that what worked last week will work this week.
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, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
Friday, October 04, 2013
Allen & Heath iLive Systems Used For Recording Of Top Brazilian Singer
Two Allen & Heath iLive mixing system were utilized to record Brazilian artist Claudia Leitte live for a DVD.
Allen & Heath digital iLive systems were used for the recording of a new DVD by one of Brazil’s most popular singers, Claudia Leitte.
Recorded at a major gig in Pernambuco Arena in Recife, an iLive-112 Control Surface with iDR-64 MixRack was used by monitor engineer, Roque Fausto, and an iLive-144 surface with iDR-64 rack using Dante, was employed by famed engineer, Beto Neves, in the mobile recording truck.
Claudia Leitte is a Brazilian axé singer and former vocalist in the group, Babado Novo.
The singer is currently one of the coaches in Brazil’s version of “The Voice”, and this latest DVD recording entitled Axe Music, is the third DVD of her solo career.
The recording took place during a concert ‘The Greatest Show In the World’, which also featured performances by other well-known Brazilian artists, Saulo Fernandes, Anitta and Wesley Safadão.
For the monitor mix, the iLive system managed 58 inputs, 30 sends and 32 wireless packs (12 mics and 20 in-ear systems). Recording engineer, Beto Neves, was stationed in his state-of-the-art Mix2Go Mobile Recording Unit, which features iLive at its heart.
All connections are made via military grade fibre optic cable with redundancy. The audio is recorded via two digital systems based on custom Apple i7s via MADI, which each record 64 channels. The system also has a ProTools interface that converts SSL Delta Link MADI format for PT Digilink.
“Once again, the monitor and recording iLive systems displayed their exceptional quality, powerfully rich sound, amazing dynamic processors, the ease of which they can be easily reconfigured, and their incredible FX rack. The compressors are a dream!” explains monitor engineer, Roque Fausto.
“For this particular show, I started with a scene from one of Claudia’s previous shows and restructured it to accommodate various inputs and outputs required for all of the guest artists and musicians involved. It was easy to do and provided no challenges.”
Allen & Heath
Thursday, October 03, 2013
Avid Expands Capability Of Live Mixing Systems With Dugan Automixing Functionality
Automatically adjusts mic levels faster than what would be possible using manual workflows
Avid has announced a new option card for Avid live systems, the Dugan-VN16, which offers a modular, integrated solution that simplifies mixing for multi-microphone applications.
Developed and manufactured by Dan Dugan Sound Design, the Dugan-VN16 option card is designed for situations where multiple speech microphones are used, including broadcast events, conferences, church services, theater performances, and more.
Specifically, the Dugan-VN16 automatically adjusts mic levels faster than what would be possible using manual workflows. Unlike a noise gate, which can introduce distracting sonic artifacts, Dugan-VN16 utilizes real-time voice activation to automatically lower the volume of unused live speech microphones and raise volume when presenters begin speaking. It can also help in reducing feedback, comb filtering, and background noise without having to manually adjust levels.
“We’re proud to partner with Avid, an industry leader in live sound,” states Dan Dugan, CEO of Dan Dugan Sound Design. “By integrating our patented automixing technology into Avid live systems, we’re allowing owners to deliver higher-quality mixes and expand their businesses to include an even broader range of live events.”
· Integrate automixing technology directly into SC48, FOH Rack-, or Mix Rack-based systems.
· Reliably manages and mixes up to 16 open microphones.
· Provides more system I/O flexibility with 16 channels of ADAT optical I/O
· Offers three operating modes to fit different application needs
· Provides configuration and mixing via the Dugan Control Panel software (included) or the Dugan Control Panel for iPad (sold separately)
“Avid’s industry-leading live sound solutions are built on the most open and tightly integrated platform,” says Chris Gahagan, senior vice president of products and services at Avid. “The Dugan-VN16 option card expands Avid live systems to include patented automixing technology, representing a reliable, easy-to-configure solution that allows live sound professionals to deliver higher-quality mixes, faster than ever.”
The Dugan-VN16 will be available at Avid resellers worldwide in Q4 2013 for $3,750 (US MSRP).
Monday, September 30, 2013
In The Studio: The Power Of Subgroups (Includes Video)
A way of making the mix process more efficient
If you’ve been mixing for any length of time, you probably know how useful (and cool) subgroups can be.
But recently, Joe Gilder has come up with a different take on doing subgroups—a small change, but something that’s definitely had a positive impact in his workflow and has helped speed his mix process.
Previously, he only subgrouped things he felt should go together—guitars, or vocals, or drums—but that approach has changed. In this video, Joe explains his new strategy on subgroups, offers additional specifics, and explains the reasons why it’s making things more efficient.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner. Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.
Blokhed Studios Invests In Soundcraft Si Expression 2 Digital Live Sound Console
A perfect fit for demands of local club gigs
Tom Stiegler, owner of Blokhed Studios (Deer Park, NY) is a drummer, recording, mastering and live sound engineer, and he’s also a self-described “sonic perfectionist” who didn’t want to make the move to live digital mixing until it met his standards. When the Soundcraft Si Expression 2 console hit the market, he was finally convinced to move to digital.
“I hated it,” was his reaction in finding earlier digital models thin-sounding and impractical to use because the faders had to be zeroed out whenever they needed to be reset. He shied away from trying a digital console again until recently, as prices came down and more manufacturers entered the market.
This time, Stiegler was more wary and spent a lot of time “obsessively researching my options” and making sonic comparisons. “Soundcraft was on my list. I always had faith in their mic preamps, and the new Si Expression Series looked like it had the features I needed. The Si Expression 2 was price-competitive with other comparable brands.”
After comparing various digital consoles in his price range, Stiegler chose the Si Expression 2. “I’m a mixing and mastering engineer,” he notes. “I hear everything. What seems like a little bit of a sonic difference to others is a big amount to me.”
Stiegler does a lot of live sound work for local Long Island and New York Metro area bands at outdoor and indoor gigs, where bands have come to rely on him for a good-sounding mix no matter where they’re playing—even in clubs with less-than-stellar acoustics and house sound systems. “I feel that local bands should have a live sound mix that’s as good as any national act,” he states. Easier said than done in some venues, which is why Stiegler often brings in extra PA gear and appreciates the Si Expression 2’s ability to adapt to any live sound situation.
In the hectic environment of club sound, Stiegler will often have to set up the Si Expression 2 in less than an hour—or even have only minutes to build a mix. He finds the console to be adept at getting him where he needs to be fast, thanks to its motorized faders, logical layout and features like Soundcraft’s simple, time-saving cue/snapshot system that lets Stiegler instantly recall any board settings from any club he’s ever worked in – “a beautiful thing.” Also useful is Soundcraft’s TOTEM (The One-Touch Easy Mix) system that reconfigures the console and allows users to press a single key to mix to an AUX, FX or Matrix bus.
Stiegler adds that all the compression, EQ, noise gate and other controls he needs the most are at his fingertips, with the “analog section” of the console right at the top and the touchscreen interface providing ready access to desired functions. Stiegler finds the FaderGlow color-coded illuminated faders a lifesaver in clubs where front of house is thrown into near-darkness when the house lights go down.
The Si Expression 2’s iPad compatibility gives Stiegler the ability to make adjustments to the console from anywhere in the room—an extremely useful feature that lets him adjust the mix from anywhere in the venue and set each musician’s monitor mix from the musician’s on-stage position.
In making the move from analog to digital, Stiegler found there was a learning curve, but not too big a one. “The Si Expression 2 is not that hard to learn. The biggest mistake you can make is walking into it thinking it’s going to be too complicated. If you’re thinking about buying a digital console in this price range, the Si Expression 2 is the one to get—it’s not even a question,” he concludes.
Church Sound: What Do You Do When Facing An Acoustical Nightmare?
If you go into it informed, you still stand a fighting chance
On a recent weekend I attended an event that was held in a very challenging space for sound reinforcement, and thought I’d detail it because it’s a good way to discuss some basic audio principles.
This particular event was held in a very, very nice pole barn. Really beautiful. It’s a large steel building, the walls covered by solid wood paneling, the floor a stained and sealed concrete, the ceiling made up of flat steel decking with no acoustical treatment. Dimensions were about 60 feet by 100 feet, forming a rectangle with parallel walls.
We were there for a formal ball, with dancing directed by a “caller” and CD tracks for music, both sources fed into the sound system. When I arrived, the person setting up sound was trying to get an omnidirectional lavalier microphone working. Not a good start. Even before hearing anything in the room, it was obvious there were going to be problems, and using an omni mic—particularly one that would be placed on the chest rather than at the mouth—is not a good way to capture strong, consistent vocal signal.
The system also included a simple 8-channel mixer, CD player, amplifier and a single 15-inch, 2 way loudspeaker (circa 1970s). In a lot of ways things were getting even tougher…
First, as predicted, the omnidirectional lav placed at mid-chest made gain before feedback tough. This became immediately obvious during the brief sound check. Second, the loudspeaker, with its “smile-shaped” horn, had very little effective pattern control.
Third, the mixer had minimal EQ, just some high/mid and low knobs usually associated with very inexpensive units. Fourth, the loudspeaker stood over three feet tall and probably weighed north of 100 pounds, so even if the sound person had a tripod stand, the loudspeaker would be too heavy for it. So instead, it was left sitting on the floor, and with associated equipment sitting on top of it.
Thankfully, he also had a wireless handheld on site, and switched to that instead. Once he got that working, he began testing the CD player. Right away I thought, “You’re playing everything way too loud. In this situation, less is definitely more.”
After some (very loud) feedback, caused when the caller for the dance went to turn off the CD player (one of those pieces of gear on top of the loudspeaker) while holding handheld mic by his side (and pointed right into the high-frequency horn), the event began.
Right away, it was obvious that the space was being excited with way too much energy, and unless you happened to be in the direct coverage pattern of the high-frequency horn, all you could hear was the mash being created by the reflected sound that was bouncing around in the room.
Besides the fact that this venue, while nice, wasn’t conducive to this type of event, the primary problem was direct versus reflected energy. Simply, this means to direct as much energy into the coverage area as feasible while minimizing energy that can reflect off of surfaces.
What would I have done?
1) Outfit the caller with a headset mic. It would sit consistently at the same distance from his mouth, thus providing a much more consistent vocal signal. Further, the caller would have been spared to problem of trying to explain/demonstrate a dance while holding a mic.
2) Add more loudspeakers, and these would be distributed around the edge of the dance floor. More sources of sound, more evenly distributed, helps achieve better direct energy while limiting reflected energy.
3) At a minimum, place the loudspeakers at ear height, or if possible, place them higher and then angle them downward. Again, this focuses the energy on to the coverage space and off the walls.
4) Turn it down! Always remember that in a situation like this, “less is more.”
Thankfully there was just the one mic and the CD player. If there had been a live band—wow it would have been really, really ugly!
As a side note, after the event was over, about a dozen or so attendees got together on one end of the building and sang a song a capella in four-part harmony. It sounded absolutely beautiful. The direct-to-reflect energy was just about right, and the natural reverberation of the room added an incredible effect to the voices of the singers.
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 30 years.
DiGiCo SD10 Console Keys System For Kansas 40-Year Milestone Concert
SD10 chosen for additional production inputs; SD Convert software makes console-to-console transition seamless
Legendary prog-rockers Kansas recently celebrated its 40-year anniversary with a one-off special concert at Benedum Center in Pittsburgh.
Accompanied by a 41-piece symphony orchestra in the first act, the next offered a traditional, rockin’ set of their classics. The event brought together original members for the first time in 30 years: the five touring members (original members Phil Ehart, Steve Walsh, Richard Williams, and long-time members Billy Greer and David Ragsdale), as well as guest spots by original members Kerry Livgren and Dave Hope.
One day prior, the production crew—including PM/FOH engineer Chad Singer, monitor engineer Derek Papp, audio tech Darian Cornish, Benedum A1 Chris Evans, guitar/violin/bass tech Jeremy Vig, drum tech Eric Holmquist, equipment relocation specialist Rusty Banks, and LD David Manion—embarked on a large-scale lighting and video load-in followed by backline and audio for the band in time for an afternoon symphony rehearsal.
The Benedum house PA was comprised of Meyer M’elody arrays and two 700HP subwoofers per side, supplemented by front fills, out fills, a center cluster, and balcony fills. Singer opted to go with a DiGiCo SD10 at FOH (running on an Optocore fiber snake) with a DiGiRack (on a MADI snake).
“We’ve been using an SD8 for FOH since 2012,” says Singer. “We typically do 60-70 shows per year so the console was out with me throughout the year that time. I rented the console from Rock N’ Road Audio in Atlanta, who have a wide selection of DiGiCo consoles available and have been big DiGiCo fans for a long time now.
“For this show, I needed more inputs than our usual band only inputs, so the natural choice was the SD10—especially since the new converter software was able to take my existing show and load it into the SD10. Immediately, I was able start to building the extra inputs. The transition was seamless. In general I have enjoyed using DiGiCo products because they have great preamps/ converters with plenty of headroom. The routing and layout of the consoles hardware and software has the most analog feel of any digital console out there.
“I don’t find myself ever having to dig though menus or pages especially during show time when all I want to think about is being creative. The consoles just make sense. I think my favorite part of the DiGiCo line is having 12 faders per bank. I had to do a fly-date show with a different console a few weeks ago and had only eight faders per bank.”
For the occasion, Singer managed a total of 92 FOH inputs and 10 FOH outputs (Left, Right, center fills, lip fills, subs to the Meyer Galileo Processer, TB, stereo symphony mix, and stereo band mix to monitor desk), with the SD Rack handling the 44 band inputs (14 drum channels, two bass, six guitars, two acoustics, six keyboard, two violin, four vocals, intro music, guest inputs for guitar, bass, keyboards, and vocals, plus four audio from video channels, and four emcee mic inputs.
The orchestra consisted of 41 members and the DiGiRack handled all 46 symphony input channels (20 DPA 4099 string mics, six woodwind channels, 10 brass channels, four tympani mics, three percussion mics, harp, and a stereo keyboard.
“Even with a pretty controlled stage volume,” notes Singer, “the only way to have a fighting chance to clearly hear the orchestra without a large phase spear from the drums is to avoid sectional area mics and close mic everything.”
The monitor desk only received the split inputs from the band inputs so Singer sent him a stereo string mix, a stereo band only mix (Derek combines it with his ambient mics for the band’s ear mixes), and a Talkback channel. “The only tricky part of the mix was creating a band-only mix and a symphony-only mix for returns to monitor world,” he says, “the desk was pretty quick making two more stereo auxes and dialing up post-fader sends of the appropriate channels.”
Singer says he found a lot to like about the SD10. “The desk can do so much. Even with all those inputs I did not need any extra gear to make audio life any easier. In addition to the 12 faders per bank, I’m a fan of the 12 CG’s. You get 24 on the SD10, so grouping can be real specific. Also, the dynamic section is very versatile and with some channels such as the tom-tom channels, I’ll use a normal comp followed by a gate.
“I’ll then send the 6 toms to a group where I can have a multi-band to catch low-mid frequencies and follow that with another comp set more like a limiter at the end of the chain and then blend that group in with the original tom channels that are already assigned to the master bus. All controlled with one CG. Then other channels like the symphony violins I use the multi band to catch upper mid frequencies and or the de-esser to knock down some of the shrill factor. I like using those to bring down cymbal and snare bleed without beating up an EQ and therefore altering the sound of the violins or cellos as well.
“The dynamic EQ is cool too. I use it on bassist Billy Greer’s vocals. He mostly sings in upper registers, so I find myself padding the upper mid-range and hi- frequencies while he sings with EQ and compression. But when he talks to the audience in between songs he speaks softer with a low voice. So instead of always cutting low-end and adding high-end to the EQ manually, I just set the dynamic EQ to do it for me.
Singer makes use of spill sets to handle all of his outs, pink noise and iPod music all in the same fader bank. “I have a ‘Dust in the Wind’ spill set for when they switch from loud guitars and drums to soft acoustics with only a few channels. It just makes organization of channels easy. Also, the DiGiTuBes on acoustic guitars and violin are pretty cool. I don’t know if it’s the right thing to do, but I slam the drive on the acoustics until the noise floor gets to be just too much. Then back it off a bit. It seems to make finger picking have some life and make it sound like it is just on the edge of taking off.”
He’s making use of all the onboard effects for reverbs, delays, and chorus effects, using mostly plate reverbs for drums, acoustic guitars, violin, vocals (and single tap delay and double effect for vocals), as well as a chorus on the acoustic guitars.
As for outboard gear, Singer uses the Waves SoundGrid bundle for instrument specifics. “I’m using Renaissance comp/exp on kick drums, Renaissance AXX comp on guitars, C4’s on snare, vocals, and violin, and an H-Comp on toms and keyboard groups where I can really clamp down on the compressor and then find the right blend between the compressed and original sound with the mix knob.
“On my master bus, I have a chain that’s the H-comp, C4, L1 limiter, then an analyzer. It’s nice to have two insert points on each channel so you can really put the out board processing where it needs to be placed exactly in the signal flow either pre or post EQ/dynamics. It’s also great to have parameter control of those through the SD 10. Since they all change from day to day depending on the venue, new strings, new drum heads, or even if the performer is playing stronger or softer than usual.”
Friday, September 27, 2013
Church Sound: Mixing Like A Pro, Part 2—Channel Layout
Grouping channels in a way to find them quickly and easily
Editor’s Note: Go here to read Part 1 of this series.
Experienced audio people have the ability to mix while keeping their eyes on the stage and the crowd. They know that they need to continually watch their musicians for visual cues in order to catch any trouble early and to verify that the people on stage are able to connect with each other.
Just as important, you should be regularly watching the audience to make sure they are engaging with the music and message. The surest way to verify that the mix is full and engaging is to watch your audience and see if they are into it.
I regularly teach new audio people that if they don’t see at least a few heads bobbing or toes tapping in the audience, they need to listen critically to their mix for changes that need to be made.
Know Right Where To Find It!
So if you’re going to mix like a pro and keep your eyes scanning the stage and the audience, having your eyes locked on your mixer is not an option. You need to be comfortable enough on your mixer that you know where everything is so you can move quickly to any given input without searching for it.
I can’t tell you how often I see churches make the mistake of having a poor input channel layout, which frankly contributes to struggling with the mix. Inputs should be arranged in a way that makes sense to anyone who walks up to the console, and should be clearly labeled so there’s no guessing or hunting for inputs. It seems like such a simple concept, yet so many churches don’t practice it.
Create a Standard Channel Layout
There are two common ways to layout the inputs of the console. The first one is what is very typical on concert tour riders and is the old school way of laying out your console. It’s the method that I tend to prefer, and it’s the layout I’m very comfortable mixing with—meaning that regardless of what console I’m on, I can find channels pretty fast without having to search for them.
Starting with channel 1 and working across, this method looks like:
Rhythm (electric and acoustic guitars, pianos and keyboards)
Playbacks (like CD, DVD, computer, etc)
Drum Mic Channel Layout
If you’re miking the drums, chances are that you’re taking up several channels on your mixer to control the sound of the kit. Having an order for the drum mic channels is also necessary for you to make quick, accurate adjustments without having to think too much about which fader controls which mic.
Here is a common order for average drum set ups based roughly on mic priority:
Kick | Snare | Tom1 | Tom2 | Tom3 | Overhead L | Overhead R
Alternate Channel Layout
Another accepted method for stages that stay set up the same week after week is to lay the console out in the order that you see things on stage. The idea behind this layout is that the inputs are left to right as your stage appears from left to right, so as the physical matches the visual.
This method can make a lot of sense for churches that leave their setup the same all of the time, though for most of us that just isn’t reality. As an example of this setup, if the stage from left to right included a singing bass guitarist, lead singer/acoustic, drums, singer and an electric, your inputs would look like:
Everything In It’s Place!
Regardless of what method you choose, the key is to choose some kind of organization that will group your channels in a way that you can find them quickly and easily, without searching. A console that has mic assignments spread out all over the place is tough to operate without focusing on the console.
With a consistent and logical channel layout, you’ll be able to mix quickly while keeping your ears and eyes up and active.
Duke DeJong has more than 12 years of experience as a technical artist, trainer and collaborator for ministries. CCI Solutions is a leading source for AV and lighting equipment, also providing system design and contracting as well as acoustic consulting. Find out more here.
Thursday, September 26, 2013
Church Sound: A Primer On Phantom Power For Microphones
We use it all of the time, but what's it really doing?
Condenser microphones need phantom power to operate their internal circuitry. The power is supplied to the mic through its 2-conductor shielded cable, and can be provided either from a stand-alone device or from a mixing console (at each mic connector).
The microphone receives power from, and sends audio to, the mixer along the same cable conductors. It’s called “phantom” because the power does not need a separate cable; it’s “hiding” in the signal conductors.
According to DIN standard 45596, phantom powering is a positive voltage (12-48V DC) on XLR pins 2 and 3 with respect to pin 1. The cable shield is the supply return. There is no voltage between pins 2 and 3. Pin 1 is ground; pin 2 is audio in-polarity, and pin 3 is audio opposite-polarity. Also, pin 1 has 0 volts; pin 2 has a positive voltage, and pin 3 has the same positive voltage as pin 2.
The phantom on/off switch in many consoles is labeled “P48” to signify “phantom power 48 volts”.
Figure 1, top, shows a microphone plugged into a stand-alone phantom power supply. Inside the phantom supply (Figure 1, middle) are two equal resistors R.
Figure 1. Phantom power equivalent circuits.
They supply equal voltage to pins 2 and 3 with respect to the pin 1 ground. Inside the mic, phantom power is tapped off two equal resistors (or a center-tapped transformer).
Figure 1, bottom, shows how the phantom current travels through the mic cable from right to left:
1. The current leaves the DC power-supply positive terminal and goes through two equal resistors.
2. The current travels along the mic cable to the mic.
3. The current is recovered inside the mic and goes through the mic circuitry.
4. The current returns to the DC power-supply negative terminal via the cable shield.
Some microphones or mic capsules work on DC bias rather than phantom power. A separate wire supplies B+ to the mic capsule. You’ll see this arrangement in lavalier mics or choir mics between the mic capsule and its XLR connector.
The mic capsule itself runs off DC bias, while the XLR connector houses a circuit that runs off phantom power. That circuit converts phantom power to DC bias for the mic capsule, and balances the signal.
Why Condenser Mics Need Powering
Let’s explain why condenser mics need power in order to operate.
In a condenser microphone transducer (Figure 2), a conductive diaphragm and an adjacent metallic disk (backplate) are charged with static electricity to form two plates of a capacitor.
When sound waves strike the diaphragm, they vary the spacing between the plates. This varies the capacitance and generates an electrical signal similar to the incoming sound wave.
The diaphragm and backplate can be charged in two ways:
1. By an externally applied voltage (from phantom power). This arrangement is called “external bias” or “true condenser”.
2. By a permanently charged electret material in the diaphragm or on the backplate. This is called “internal bias” or “electret condenser”.
The output of the condenser mic capsule is extremely high impedance so it is very hum-sensitive.
To bring that impedance down to a usable value, an impedance-converter circuit is connected to the capsule output.
Figure 2. A condenser transducer (mic capsule).
This circuit is necessary whether the capsule is electret or non-electret. The converter needs a DC voltage to power it, and this voltage is supplied by the phantom power supply. Sometimes other transistors are added to give the mic a balanced output, and these components work off phantom power too.
In contrast, a dynamic microphone needs no power because it has no active electronics. It generates its own electricity like a loudspeaker in reverse. In a moving-coil dynamic microphone (Figure 3), a coil of wire is attached to a diaphragm.
Figure 3. A dynamic (moving coil) transducer or mic capsule.
This voice coil is suspended in a magnetic field. When sound waves vibrate the diaphragm and its attached coil, the coil vibrates in the magnetic field and generates an electrical signal similar to the incoming sound wave. The voice-coil leads are soldered to XLR pins 2 and 3 inside the microphone.
In a ribbon microphone (Figure 4), the diaphragm is a thin metal foil or ribbon. Sound waves vibrate the ribbon in a magnetic field and generate corresponding electrical signals. The ribbon leads are soldered to XLR pins 2 and 3 inside the microphone.
Figure 4. A ribbon transducer (mic capsule).
You can plug a dynamic or ribbon microphone into a phantom supply without damaging the mic. That’s because the voice-coil or ribbon leads are not connected to pin 1, so no current from the phantom supply can flow through them.
However, if there’s any imbalance in the phantom voltage applied to pins 2 and 3, a current will flow through the microphone voice coil or ribbon (which is connected to pins 2 and 3).
Or if one terminal of the coil or ribbon is accidentally shorted to the grounded mic housing, a current will flow through the coil or ribbon. For this reason, it’s best to switch off phantom power for dynamic and ribbon mics.
Using a Stand-Alone Supply
You can buy a phantom power supply from your microphone dealer or online. Some supplies are AC powered; some are battery powered; some are both. Some can power a single microphone; others can power several at once.
In any case, you plug the supply in series with the mic cable. The supply has XLR-type input and output connectors, one pair per channel.
Connect a mic cable between your microphone and the supply’s input connector. Plug another mic cable between the supply’s output connector and your mixer mic input.
If you need to convert a low-Z condenser mic to high-Z unbalanced using a transformer/adapter, run the mic through a stand-alone phantom supply before converting it to high-Z unbalanced.
Cautions for Use
Don’t plug a mic into an input with phantom already switched on, or you’ll hear a loud pop. If you have no choice (as during a live concert), mute the mic channel when you plug the mic in.
Make sure your phantom voltage is adequate for your microphones. Some mics start to distort or lose level if the phantom voltage drops significantly below 48 volts.
To measure the phantom voltage at your mixer’s mic input, get a DC voltmeter and measure between XLR pins 1 and 2. Do the same between pins 1 and 3.
Be aware of phantom-voltage sag. Microphones draw current through the phantom-supply’s resistors, and that current causes a voltage drop E = IR across each resistor. The higher the current drain of a microphone, the more it drops the phantom voltage at the mic connector. Current drain is usually specified in the mic’s data sheet.
Power supplies are rated in the total number of milliamps they can supply. Make sure that the total current drain of all the mics plugged into the supply doesn’t exceed the supply’s current rating.
Avoid having phantom in a patch bay because someone is likely to patch in and cause a pop. If you must patch into a jack with phantom on it, mute the input module that the mic is connected to, or turn down its fader. Mic-level patches should be avoided anyway.
Some phantom supplies cause a hum when you plug in a connector that ties the shell to ground. Float the shell. This also helps to prevent ground loops.
Since the cable shield carries the DC return, be sure the shield and its solder connections are secure. Otherwise you can expect crackling noises—especially when the cable is moved.
Some microphones work on either internal batteries or external phantom power. In most designs, connecting the mic to phantom automatically removes the battery from the circuit. Otherwise, the battery would severely load down the phantom supply. It this appears to be happening, remove the battery.
If a condenser microphone doesn’t work due to low phantom-supply voltage after the mic is plugged in, try these suggestions:
1. Supply phantom from a better-regulated console.
2. Use a mic with less current drain or with lower phantom-voltage requirements.
3. Add a voltage regulator to the supply voltage.
AES and SynAudCon member Bruce Bartlett is a recording engineer, audio journalist, and microphone engineer. His latest books are “Practical Recording Techniques, 6th Edition” and “Recording Music On Location.”
Mackie Adds Apple Lightning Connector To DL1608 & DL806 Digital Mixers
Enables direct docking of the iPad (4th generation) and iPad mini (using the new iPad mini tray kit)
Mackie has introduced two new versions of the DL1608 and DL806 digital mixers that include the Apple Lightning connector, enabling direct docking of the iPad (4th generation) and iPad mini (using the new iPad mini tray kit).
While both of these iPad devices currently work with DL mixers wirelessly via Mackie’s Master Fader app, physical docking offers advantages. In addition to device charging, users will have access to 2-channel recording straight to the iPad and the ability to playback audio from nearly any iPad app straight to the DL mixer.
The new DL1608 and DL806 featuring the Apple Lightning connector are currently in production and are shipping to dealers and distributors worldwide.
The Master Fader Control App is available for free from the App Store on iPad or at www.AppStore.com/MackieMasterFaderControl, while the My Fader Control App for iPhone and iPod touch is available for free download from the App Store at www.AppStore.com/MackieMyFaderControl.
Wednesday, September 25, 2013
New SSL Live Console Now Shipping
Three to Britannia Row and two more to SGroup in France
The new Solid State Logic (SSL) Live console, which debuted at Prolight + Sound in April, 2013, is now shipping.
The first three consoles all shipped to U.K.-based global tour production company Britannia Row for use on Peter Gabriel’s forthcoming European ‘Back to Front’ tour, and another two shipped to SGroup in France for the imminent Amel Bent tour.
Console manufacturing production for 2013 has been sold out since July, and details of the new commercial partner network for SSL Live are available in the “Where to Buy” section of the SSL website (here).
SSL CEO Antony David states: “The on-schedule completion of the new Live console is an important milestone for SSL. This has been one of the biggest developments we have undertaken for some time and marks the first application of our new Tempest digital platform. We have been very encouraged by the response from mix engineers, rental companies and our channel partners since we presented the console in April this year. Demand has substantially outstripped our initial production plans, but we will return to reasonable lead times by early 2014.”
Since April, SSL has expanded its dedicated Live product team with key hires including Jason Kelly as Live Consoles product manager based in the U.K. office, and Jay Easley as vice president – Live Consoles to lead SSL’s live sector sales operation in North America.
Certified training courses have also commenced, with focus on commercial partners and initial purchasers. A training program for the wider operator community is scheduled to commence from January, 2014.
The SSL Live will be exhibited at next month’s 135th International AES Convention in New York and at ISE in Amsterdam in February 2014. With its latest offerings, SSL will relocate to Hall 8 at Prolight + Sound in Frankfurt in March 2014 and the company will exhibit at InfoComm for the first time, in Las Vegas in June, 2014.
Solid State Logic