Tuesday, January 06, 2015
Church Sound: Upgrading Essential System Components
“How do I get the biggest change in my sound quality for the smallest investment?”
I’m often asked this question from cash-strapped churches that need a sound system upgrade but don’t have the funds to accomplish it all at once.
If it’s the sound operator who approaches me first, his goal is usually to get a new mixing board. If it’s the worship leader, he’s usually focused on microphones and/or monitors.
Then there’s the pastor, who’s most often interested in the solution that will get everybody else to stop bothering him.
So what’s the answer?
I go about solving this dilemma by looking at the number of people who can potentially benefit from each upgrade. With that in mind, what follows is a suggestion of how you can determine your next “best” upgrade.
First, look at what I call the “heart” of the system. That is, loudspeakers, amplifiers and signal processing. Not only are these usually the biggest ticket items (though the mixing console may compete), they’re also the items that in most cases will bring the most significant improvement to a system.
In surveying the heart of your sound system, first check out the loudspeakers to ensure that they’re working properly. Are there blown drivers? Hear any rattles or other strange noises?
Do some research to find out the coverage pattern of the loudspeakers, and map that coverage over your seating area. Is the coverage adequate or are there zones that are being missed? (You can also hear this by slowly walking through the coverage area with the system playing tracks.)
Continue your research and determine the frequency response of the loudspeakers. If they roll off at 180 Hz, it’s not likely that they’re producing the nice “thump” out of the kick drum or any of the deeper lows from the bass guitar.
Next, find out power handling of the loudspeakers and match that up with the power available from your amplifiers. If you don’t have enough “headroom” (available “extra” power), the system will always sound mushy and like it’s being “pushed.” (As in pushed too hard.)
Finally, give the signal processing a good look. If it consists of a number of analog devices (EQ, crossover, delay, etc.), it could be time to upgrade to a quality digital processing unit. Even better, have a new digital processor implemented by a qualified professional who knows how best to use it to maximize the performance of your loudspeakers.
And that leads me to an important point. If you find any of the above aspects lacking during your research, consider bringing in a qualified professional to help make the most of what you have.
While you may be able to address some of these aspects adequately, it’s not a game for amateurs. Quite often, the use of professional assistance, combined with a new component or two, can make all of the difference in the world while still fitting within the confines of a tight budget. And it’s almost always money well spent.
Once the “heart” of the system is taken care of, feel free to move on to mixing boards, monitors, microphones and other accessories.
I look at it this way: the best sounding microphone is only going to sound as good as the loudspeakers reinforcing it.
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.
In The Studio: Checking Drum Tracks For Phase Issues
When working with a multi-miked drumkit recording, it’s important to be aware of phase issues and how to correct them. Recognizing the phase issues will take some practice, but if you go step by step, there shouldn’t be any problems.
Before doing any processing or even setting levels, go through one mic at a time and check the polarity. What you’re listening for is improved punch and low end.
—Starting with the overhead mics check, that those two mics are in phase. It’s rare but not unheard of.
—Next add the snare top mic. Bring up the volume and then try inverting the polarity, listen if the low frequencies change, decide which way it sounds best, with the most low end or punch and continue to the next mic.
—If you have a mic on the bottom of the snare pointing up, it’s very likely it will need the opposite of the snare top mic.
—Add the kick mic, toms, room mics and close-miked cymbals.
—Then move on to panning, balance and processing.
There is no preset for this. Every time you record or mix drums you’ll need to check this and compensate. You’re going to have to use your ears!
If you layer and blend samples, check to make sure they’re all working together in phase as well.
However, changing the timing of individual mics—like you might do with multi-miked guitar amps—is not something that I mess with when it comes to drums. It’s not something that you should need to do.
Jon Tidey is a producer/engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com.
Monday, January 05, 2015
In The Studio: Attaining A Fuller Sound With The Mix (Video)
Sometimes emptiness in a song can be a good thing, but there are times when you may want it to sound more full, bigger and richer. And it can be a struggle to attain that.
In this video, Joe provides some solutions that he notes aren’t new or revolutionary, but they do often go overlooked in the quest for a thicker sound when and where you want it.
Some of it involves using different types of pads, particularly on ballads and songs of that nature, and often, it’s not big or obvious—but it can make a difference in attaining the desired result.
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.
Cast Of Characters: Ever Worked With Any Of These Folks?
Over the years I’ve had the pleasure of working with many great folks in our business… and then there were these people.
Twisty Knob Guy: Wouldn’t stop turning knobs, even when everything sounded great. Either he was just never happy with his own mix, or didn’t want to sit still for a minute and have the promoter think he wasn’t earning his pay.
The Pickup Artist: Spent more time at catering trying to get a date with the servers than he did setting up the show. I kept waiting for him to grab a wireless and do the line from the Mr. Microphone commercial: “Hey good lookin’, I’ll be back to pick you up later!”
The Literary Snob: Why mix or even pay attention to the performers when you can sit at front of house and read a magazine? On the plus side it was an audio magazine, but on the minus side it was devoted to hearing aid technology.
Tweety Bird: Spent more time updating the band’s fans about the gig on Twitter then he did making the band sound good for the fans that actually showed up to hear them. (#BadMix)
Marathon Man: Constantly walked the room, checking the PA. Unfortunately he spent so little time at the console that he missed important things like setting delay tempo or turning off the reverb when the vocalist was just talking to the crowd. It was funny, however, watching him morph into “Sprinting Man” when a guitar lead started.
Texting Dude: A person of the lighting persuasion (not that there’s anything wrong with that) who was more interested in his text screen than following show cues. Wait—the show’s over here, but the lights are over there…what’s going on? Oh, texting again. (Perhaps consulting with an optometrist?)
We Don’t Need No Stinkin’ Vocals Guy (a.k.a., Instrumentals Guy): Buried the vocals so far down in the mix that the singers looked like they were doing an elaborate pantomime of singers. All of that frenetic motion, none of that hot, velvety vocal action. (At the time it was amusing in a “we’ll all laugh about this years from now” way. And we do.)
Baker’s Dozen: Always offered more than a dozen reasons why he didn’t get things done like the rest of the crew. In return, I had just one reason why he never worked with us again. (I prefer a minimalist approach.)
Monitor Monarch: Didn’t like to give performers what they wanted in their own monitors. Never having performed onstage or even played an instrument, nonetheless he still knew best and doled out to the peasantry only what he thought they really needed.
Can You Hear Me Now Nerd: Took a call at front of house during the show, talked for a minute, put his finger in his ear and angrily looked up at the stage, and then wandered out of the venue into the hallway to finish the call. Returned 20 minutes later. (Perhaps it was a conference call with the Monitor Monarch and Baker’s Dozen Guy, strategizing new ways of how not to serve a client.)
BDS (Bruce Dickinson Syndrome) Sufferer: Could attain a pretty good mix with one exception: one instrument not normally at the forefront of the mix was obnoxiously loud. And it was obvious to everyone except him. In the words of The Bruce Dickinson himself, “I got a fever, and the only prescription is more cowbell!” Or in this case, more congas.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Wednesday, December 24, 2014
PSW Top 20: Most-Read Articles Of 2014
As we turn the page on 2014, we’re happy to present the 20 articles that were the most-read over this past year on ProSoundWeb, based upon total page views.
Note that some of the articles that delivered top results over the past 12 months were actually written and posted over a year ago, but they continue to prove of high interest and value to our worldwide readership.
In addition, some very popular articles posted more recently have not had as much time to accumulate traffic as others that have been posted for a longer period of time. We suspect you’ll see some of those fine articles on next year’s list.
Without further adieu, here are the top 20 articles on PSW for 2014.
Most-Read Articles #20 - #16 (Posted Wednesday, December 24)
#20: Ghost In The Machine: Phantom Power
By Bruce Bartlett
#19: loud, Loud, LOUD
By Dave Rat & Keith Clark
#18: Pursuit Of Perfection: Concert Sound For Steely Dan’s Summer Tour
By Mark Frink
#17: All The World’s A Stage… But Does It Need Sound Reinforcement?
By Bob McCarthy
#16: Perception Is Reality: Psychoacoustics From An Audio Engineer’s Perspective
By Todd Hartmann
Most-Read Articles #15 - #11 (Posted Friday, December 26)
#15: Too Loud? Maybe Volume Isn’t The Reason
By Karl Winkler
#14: Do You Speak Geek? The Unique Language Of Audio Analysis
By Pat Brown
#13: Playing With The Click—Advice From A Top Session Drummer
By Bobby Owsinski
#12: Top 10 Tips For Mixing In-Ear Monitoring
By Mark Frink
#11: Special Report: The State Of Production Audio Wireless
By James Stoffo
Most-Read Articles #10 - #6 (Posted Monday, December 29)
#10: Trends & Norms: New Developments In The World Of Digital Consoles
By Craig Leerman
#9: RE/P Files:The Coming Of Age For The Once-Maverick Touring Sound Business
By David Scheirman
#8: Studio Compression: When, Why To Use Slow & Fast Attack Times
By Joe Gilder
#7: Church Sound: What Four Wireless Mistakes Are You Making?
By Chris Huff
#6: Mixing Chicago Live In The 1970s
By Mike Stahl
Most-Read Articles #5 - #1 (Posted Tuesday, December 30)
#5: Church Sound: IEM Mixing For Worship
By Kent Margraves
#4: Church Sound: Eighteen Live Audio Mixing Tips & Tricks
By Chris Huff
#3: 30 For 30: Lessons Learned From Years Of Tuning Sound Systems
By Bob McCarthy
#2: Moving Air: Inside Subwoofer Designs & Configurations
By Craig Leerman
#1: A Useful Tool: Creating & Applying FIR Filters
By Pat Brown
All-Time Top 5 Most-Read Articles (Posted Wednesday, December 31)
#5: Steps You Can Take To Improve Your Mix Right Now
By Karl Winkler
#4: The Mighty Kick Drum Mic: A Love Affair With Channel One
By Mark Frink
#3: In The Studio: Microphone Techniques To Get Great Electric Guitar Sound
By Barry Rudolph
#2: In The Studio: EQ And Compression Techniques For Vocals, Acoustic Guitar
By Cliff Goldmacher
#1: How To Build Your Own Plate Reverb: A Concise Step By Step Process
By Bob Buontempo
Tuesday, December 23, 2014
Modern Advantages: Delving Into The Latest On Compact Loudspeakers
It’s amazing to take a look at the latest generation of portable loudspeaker systems in terms of how far they’ve progressed.
When I was with Electro-Voice as product manager for music systems in the mid-1980s, the concentration was on the design and porting of the cabinets, tweaking the components and passive crossovers for efficiency and uniform frequency response, and protecting high-frequency drivers. A big innovation was using a polyswitch in parallel with a resistor for the latter purpose, nicknamed “the PRO circuit.”
Even mounting sockets and poles in the top panels of subwoofer cabinets were a new innovation. The first power amplifier that EV put into a cabinet was a modestly powered unit in the 12-inch Sb200 sub.
Now a walk through the Winter NAMM show or an online search will yield a variety of portable loudspeakers melded with sophisticated multichannel amplifiers, DSP control, and even display screens – typically priced at less than the equivalent package of loudspeakers, amplifiers, and processors would have cost then, factoring in inflation.
In addition to delivering very good sound and providing integral protection for the drivers, these new systems require fewer cable runs and dramatically speed up the setup process.
Most of the systems in this genre have active subwoofers available to add an extra low-end dynamic while tightly integrating with their full-range companions, usually outfitted with a pole-mount for convenient mounting. Another facet of these designs is that many are well-suited for stage monitoring, with a cabinet angle optimized for this application.
Earning The Name
For years, we’ve been calling them loudspeaker “systems,” and while that’s technically true, many of today’s portable models truly earn that moniker.
QSC K Series
For example, the QSC K Series, available with 12-, 10- and 8-inch woofers, are driven by an onboard 1,000-watt class D power module combined with extensive DSP such as DMT (Directivity Matched Transition) that provides for matched LF and HF across the coverage area to help eliminate “dead” or “hot” frequency zones.
And Intrinsic Correction, borrowed from the company’s line arrays products, maps 65 to 75 spatially-averaged measurements to IIR and FIR filters that actively adjust time, frequency and amplitude response to a maximally flat band pass target.
Also onboard K Series models is a mixer with two combo XLR and 1/4-inch inputs, and stereo RCA inputs, (three audio input sources), two direct channel outputs and a single summed balanced output.
It’s all housed in a rugged ABS enclosure with handles that weighs just 32 pounds for the 10-inch model. The cabinet also has 35 mm pole sockets with Tilt-Direct for aiming/adjustment, and for install applications, there are M10 rigging points as well as eyebolt and yoke mount accessories.
Other examples abound. The Mackie DLM Series, available with 12- or 8-inch woofers, is very compact – particularly for a system with up to 2,000 watts of class D power onboard.
This is primarily due to the proprietary TruSource driver, which combines the HF and LF drivers into a vertically-aligned, common-magnet design, and contributes to a footprint for the 12-inch model of just 15.9 x 15.3 x 14.3 inches (h x w x d) and weight just a touch over 30 pounds in part due to a durable but light PC-ABS cabinet that has a pole mount as well as M10 rigging points and other install options.
DLM Series boxes also incorporate a digital mixer with an OLED screen offering versatile input channels with FX, independent channel level, 3-band EQ and effects control, and 16 channel effects that include reverb, chorus and delay. There’s a multi-band feedback “destroyer,” six modes to tailor voicing, alignment delay up to 300 ms, and three memory locations for instant venue setting recall. It handles mic, line, stereo and instrument signals with XLR/TRS combo and dual RCA connectors.
Also pushing the envelop, PreSonus recently unveiled the SL-Dante-SPK upgrade option to make the StudioLive AI-series the first Dante-enabled active loudspeakers on the market.
The three models in the series are 3-way designs, all with a coaxial 8-inch cone and 1.75-inch compression driver, and 15-inch, 12-inch and dual 8-inch woofer options. These systems are tri-amped (quad-amped in the dual-8 model), with class D amplifiers providing a combined 2,000 watts of power.
StudioLive AI-series loudspeakers include a USB Wi-Fi module to connect to SL Room Control software over a wireless network, and now with the new SL-Dante-SPK card this can be done on a Dante network.
SL Room Control is a system-configuration application for Mac OS X, Windows, and iPad that provides both individual and grouped loudspeaker control, and it includes a 31-band graphic EQ and an 8-band parametric EQ, muting, soloing, and level control, in addition to performance monitoring (over-temperature, click detection, and excursions).
Each full-range system has a combo XLR/TRS line input and an XLR mic input with an XMAX Class A mic preamp and 12-volt phantom power, as well as an XLR audio thruput.
PreSonus StudioLive 328AI
The systems are pole-mountable and have two side handles, interlocking stacking, and M10 flypoints. Enclosures are plywood, and even with the additional components, size and weight are more than manageable; for example, the 12-inch model measures 24.8 x 19.5 x 16.4 inches (h x w x d) and weighs 62 pounds.
A Long Way
Based on my earlier experience at EV, I was especially surprised at this year’s NAMM when I saw the company’s latest ETX Series systems and realized just how far the technology has come. Rear-panel electronics include a 2-channel mic/line mixer and a pass-thru XLR connector, a push-to-enter rotary encoder, and a backlit LCD screen.
The menu accesses a variety of functions, ranging from input and output control, protection and limiting, and presets for music and spoken word applications, to EQ settings for where the system is positioned (on the floor as a monitor, on a stand, closely arrayed with other loudspeakers, or flown) and delay settings when used as a distributed system. The transducers and wave guides have also been redesigned.
At the top of a deep line of portable PA options from JBL is the new PRX700 Series that bridges the gap between smaller gig and multiple-loudspeaker pro applications. Two-way models are available with 10-, 12- and single and dual 15-inch woofers, and there’s also a three-way model and two accompanying sub options (15-inch and 18-inch).
All incorporate proprietary Differential Drive technology developed for the company’s flagship touring systems, along with class D amplification, a DSP input section, crossover, optimization, selectable system EQ, and a dbx Type IV limiter circuit.
The input panel includes two 1/4-inch/XLR-inch combo jacks with a mic/line and ground-lift switch, a pair of RCA connectors, and separate selector switches for both the local amp and pass-thru XLR. Cabinets have a pole-mount cup and integrated M10 rigging points.
Active & Passive
Although active designs are the strong trend, passive designs are still common from most manufacturers, and often, users have a choice of the same loudspeaker in an active or passive design.
For example, Yamaha Pro Audio recently unveiled two new additions to its portable loudspeaker lineup – the active DBR Series and the passive CBR Series, both with 10-, 12- and 15-inch woofer options.
In the DBR Series, all three incorporate proprietary FIR-X tuning that uses linear phase FIR filters for the crossover, while D-CONTOUR dynamic multi-band processing applies optimized settings for either front-of-house or floor monitoring applications. Class D amplifiers deliver up to 1,000 watts of power.
There’s a 2-channel mixer, with a combo jack on channel 1 that accepts both XLR and TRS phone. Channel 2 also has a combo jack as well as RCA pin jacks for input from CD players or stereo line-level sources. The mixer offers the ability to select either channel 1 plus 2 mix, or channel 1 pass-thru.
Enclosures are durable, lightweight plastic, with an optimized 50-degree wedge angle monitoring, while the symmetrical cabinet shape of the DBR12 and DBR15 allows them to be placed in a “mirror-mode” configuration. All models also have pole mounts and rigging points compatible with optional speaker brackets and eyebolts for install applications.
The passive CBR Series are lighter than their active counterparts, equipped with a Speakon jack and a 1/4-inch phone jack. They’re tuned via an in-depth analysis and adjustment of an internal passive network, and a protection circuit limits excessive input to the HF driver to reduce the risk of the damage.
The EAW JF Series continues to grow and advance with a steady stream of upgrades, with a range of 2-way models available in both active (NT) and passive versions, with choice of 10-inch, 12-inch and 15-inch woofers. (A few passive versions with smaller woofers are also available).
JFNT (active) models have 1,500 watts of onboard amplification, proprietary EAW Focusing processing, software-accessible front-end DSP, proprietary U-Net audio and communications network, and an adaptable enclosure design.
EAW JF26NT (left) and Grund Audio ACX-2
All models natively offer a combination of portable features and M10 rigging points. Available universal accessories include trim plates that hide handles and provide a connection point for U-brackets and quick release flytrack segments, and adjustable legs for use as a monitor.
Grund Audio offers a broad range of portable PA choices, including the recent expansion of the passive ACX Series with four new active models, two of them stage monitors. designated with an “A” at the end of the model number. The onboard power amp’s multichannel inputs include an XLR balanced microphone input, a 1/4-inch line input, and RCA inputs for use with CD players and similar equipment. Any two sources can be mixed internally.
All ACX models utilize medium-density fiberboard enclosures and provide component concealing grilles, and all have handles. The loudspeakers also have a pole mount and three 2 x 2 flypoints, with optional eyebolts, dual-mount poles, and crank-style and adjustable poles available.
All In One
There are also numerous options available as “all-in-one” systems, incorporating loudspeakers, amplification, all required cabling, and often mixers, microphones and/or wireless systems.
Typically, they’re quite compact and are a good choice for more limited applications such as small stages.
RCF just introduced EVOX, a self-contained, highly portable system with the mid-high loudspeaker cabinet fitting into the back of the subwoofer, which also protects the electronics and system connections. A pole mount is included, with the 3-piece pole allowing the loudspeaker to be positioned at different heights. Cable clips secure connections.
Two EVOX versions are available – the larger EVOX 8 is powered by a 700-watt (RMS) amplifier, with a 12-inch subwoofer and a loudspeaker with eight 2-inch compression drivers rated as capable of achieving 128 dB max SPL. A progressive array shape optimizes directivity while minimizing HF beaming. Total package size is 11.6 x 13.6 x 19.3 inches (h x q x d), and weight is 51 pounds.
Another example is the Yamaha Pro Audio STAGEPAS line, also with two models. Both include two loudspeakers and a detachable powered mixer, along with a pair of loudspeaker cables and power cord.
Yamaha STAGEPAS 600i
The mixer, which fits into one of the loudspeaker enclosures for an even smaller footprint, includes four mic/line inputs and six line inputs (the 400i includes four line inputs). Channels 3 and 4 feature a combo jack that can accommodate XLR and 1/4-inch cables.
In addition, both models offer RCA jacks, a 1/8-inch stereo mini jack, an iPod/iPhone USB input, switchable Hi-Z inputs, and phantom power. The mixer includes different high-resolution SPX reverbs as well as Yamaha’s 1-knob Master EQ offering optimized settings.
An onboard feedback suppressor lets users remove unwanted squeals. A flexible new feature, switchable stereo/mono inputs, can transform each stereo channel into two independent mono channels if an application requires more input capability. The larger STAGEPAS 600i model weighs just 56 pounds.
Another growing segment of the market, column loudspeakers, can also fulfill portable PA duties quite well. With IC Live, Renkus-Heinz provides a very flexible package that can meet smaller applications to concerts for thousands. The basic IC Live module can be used alone or combined with IC215S (dual 15-inch) subwoofers for added LF impact, and the subs can act as tall or wide bases for the mains with interlocking hardware.
Renkus-Heinz IC Live and IC215S subwoofer
IC Live output can be focused more full on the audience, and off of surrounding surfaces, via digital beam steering. Software allows users to define coverage. Multi-channel class D amplifiers with integral DSP control every array element. Proprietary RHAON provides remote control and supervision functions, along with the ability to store up to 10 preset configurations in memory. The thin profile of the modules and weight of just 61 pounds makes them easy to transport.
The Carvin Audio TRx3903 vertical column has nine 3-inch FaitalPRO cone drivers in an enclosure that measures 31.5 x 5.25 x 5.5 inches (h x w x d) and weighs just 21 pounds. It’s driven by outboard power or can be accompanied by the TRx3018A self-powered sub that can also be ordered with an additional amplifier channel with DSP for the TRx3903.
The cabinets have a pole-mount cup located on both top and bottom, and by using the SS7 short pole adapter (included), the units can be safely stacked atop one another as well as on the subwoofer. Eight flypoints (four on top and four on the bottom) are included for 3/8-16 eyebolts, and the optional U-bracket can be utilized for additional mounting options. Connections include two 4-pin Speakon connectors on the rear of the enclosure, one near the top and the other near the bottom. They’re positioned so that a short jumper can link two cabinets together.
HK Audio SOUNDCADDY One
Smaller-scaled systems include the HK Audio SOUNDCADDY, which comes as a single unit (with built-in wheels). The active system’s mid-high section is housed within the slim LF cabinet, and with a press of the hand, it “telescopes” out via an integrated pole that can be aimed as needed.
Three 6-inch woofers cover the LF and six 3.5-inch cones yield mids and highs, driven by a 600-watt amp. There’s also an integral 4-channel mixer.
What’s been presented here is really just the tip of the portable loudspeaker iceberg. Manufacturers are bringing technology developed for pro touring and install applications into the compact (and affordable) range, and systems are simpler than ever to set up and start playing. After building my early-90s vintage system three times in the past couple of weeks for gigs, the thought of a 5-minute load-in and setup is very tempting…
Gary Parks is a pro audio writer who has worked in the industry for more than 25 years, holding marketing and management positions with several leading manufacturers.
Thursday, December 18, 2014
The Differences Between Dynamic Range & Signal-to-Noise Ratio
Adequate signal-to-noise ratio is one of the characteristics of a professionally designed sound reinforcement system.
The terms “dynamic range” and “signal-to-noise ratio” are often used interchangeably, but a closer look reveals that they are not exactly the same thing.
The dynamic range of a sound system is the difference in level between the highest signal peak that can be reproduced by the system (or device in the system) and the amplitude of the highest spectral component of the noise floor.
Every electronic device has a dynamic range that is determined primarily by the limitations of the power supply and the residual noise level of the unit. A strong narrow band component in a device’s noise floor will limit the dynamic range of the system.
Signal-to-noise ratio is the difference in level between the average signal level and the average level of the noise floor. A device that is being driven to some average out put level with common program material should have peaks that exceed this level by 10 to 20 dB.
This is why we operate mixers near their “0” indication on their meter, and the rest of the available voltage swing is reserved for peaks in the program material. The “average” level is of importance, because it is what we listeners (and meters) use to judge the loudness of the program.
If a voltmeter were used to measure the R. M. S. value of the device’s residual noise, the signal-to-noise ratio will be the difference in level between this value (usually expressed in dBV or dBu) and the nominal “zero” output level ex pressed using the same dB reference. This assumes that the device is being driven to its “meter zero” which is where we like to operate most mixers to optimize their gain structure.
The dynamic range of a system (or component) is not dependent on a signal being present. It’s just the difference between the maximum possible undistorted out put level and the highest component in the noise floor (usually “A” weighted). Signal-to-noise ratio requires a signal, so it must be measured under actual use of the system or component.
A system with wide dynamic range may have poor signal-to-noise due to the way that it is operated. Dynamic range can be used to describe the performance that is possible to achieve with a system or device, whereas signal-to-noise ratio might be used to describe what is actually achieved in practice.
A sound level meter could be used to measure the “A” weighted L (sound pressure level) of a live performance at a typical listening position by simply holding the meter in the air, selecting the appropriate scale, and reading the indicator.
The “A” scale is normally used since it is most sensitive to the part of the spectrum where humans are most sensitive (1-4 kHz). Since most sound level meters are of the averaging type, this would yield the average sound level of the performance.
Of course, there will be peaks in the program material that exceed this average value that the meter cannot respond fast enough to read. This “meter lag” is usually on the order of 10 dB but could be higher (or lower) depending on the program material.
Now, if all sound sources on stage were silenced (and mics left open), what would remain would be the noise floor of the system, which could be measured on the same meter in the same fashion. In a properly designed sound system, this noise floor would be established by the ambient pickup of the open microphones (not the residual noise in the electronic components).
In an auditorium with a 40 dBA noise floor, the signal-to-noise ratio at a typical lectern microphone will only be about 37 dB with an average talker (71 dBA @ 2 feet) speaking one foot from the mic. Ten open microphones could increase the noise floor by another 10 dB if their sensitivity and level setting were the same as the lectern mic, since the 10 log (number open mics) = 10 dB.
Unfortunately, the signal-to-noise ratio of the system cannot exceed the worst case condition at the open microphone since the sound system has no choice but to amplify the room noise along with the desired signal. The effects of distant miking and failure to attenuate unneeded mics on the system’s noise floor becomes immediately apparent.
Now, if a strong vocalist produced 120 dBA into a handheld mic in this same system (not unusual for close-miked vocalists), the signal-to-noise ratio would be a healthier 80 dB, since the 120 dBA -40 dBA = 80 dB. This is why we preach that good mic technique is essential for good system performance, as it ultimately establishes the signal-to-noise ratio of the system.
We use 25 dB as the minimum criteria for signal-to-noise ratio on an auditorium sound system with the highest NOM (number-of-open-mics) required for that system.
On the same system, let’s say that the loudest instantaneous sound that the system could produce in a linear fashion was 110 dBA at the same listener position.
Even if the system were operated at a 90 dBA average level, peaks of this magnitude are certainly possible. The highest program peak would be determined by the loudspeaker being used and the available amplifier power connected to it. We now have one ingredient required to find the dynamic range of the system.
If the loudest component of the noise floor were a narrow band “whine” produced by the air conditioner that measured at 35 dBA on a real time analyzer, then the dynamic range of the system would be nothing to get excited about. (110 dBA- 35 dBA = 75 dB). The dynamic range could be increased by fixing the air conditioner to remove the whine.
As you can see from these examples, it is normally the environment that determines both the dynamic range and signal-to-noise ratio of a sound system, as far as the audience is concerned. Because most electronic components in the system have a dynamic range on the order of 100 dB or more, the sound system itself should never be the weak link when it comes to the end result at a listener. A professional system should have a dynamic range of at least 96 dB in the electronics with all devices operating.
Only in a studio or home theater should the equipment noise floor become a factor in determining either the dynamic range or signal-to-noise ratio at a listener position. One could therefore design a sound system with very wide dynamic range, but the overall signal-to-noise ratio could be quite poor due to the room, etc.
One could also design a system with over 100 dB of dynamic range in each component, only to find that the signal-to-noise ratio is drastically reduced during actual use by poor gain structure in the system calibration. The most common cause of this is amplifiers set “wide open"and mixers operated at -20 dBV to compensate.
In designing systems, we choose individual components that have wide dynamic range, and then calibrate and operate the system for the maximum signal-to-noise ratio that we can achieve.
Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops online and around the world. For more information go to www.prosoundtraining.com.
Wednesday, December 17, 2014
Right Fit: Audio For The Hammerstein Ballroom
In 1906, Oscar Hammerstein I (grandfather of the famed lyricist) opened the Manhattan Opera House, intended as a venue to make opera accessible to a wider variety of patrons than was typical at the time.
In the century-plus years since, it’s also hosted vaudeville and concerts of all descriptions and a wide range of special events, in addition to serving in addition to serving as a scoring stage for films and recording artists.
Now fittingly known as the Hammerstein Ballroom at The Manhattan Center, the elegant three-tiered, 12,000-square-foot facility located on 34th Street in Manhattan has a new sound reinforcement system, part of a two-year, phased renovation and upgrade of The Manhattan Center overall. Yet while the Hammerstein Ballroom was part of the project, the renovations were purely cosmetic and did not include acoustic treatment.
Yet while the Hammerstein Ballroom was part of the project, the renovations were purely cosmetic and did not include acoustic treatment.
Discussions about the new system began in 2012, during which time representatives of the venue were considering various potential designs, comparing manufacturers and looking at the benefits of buying versus renting. The process culminated earlier this year with the installation of the system.
The venue’s L-Acoustics K2 arrays that can be easily moved as needed. SB28 subwoofers are below.
Initially, Peter Auslan, production manager for The Manhattan Center and the Hammerstein Ballroom, reached out to various manufacturers, including L-Acoustics, as part of the system comparison process, and had an ongoing relationship with NYC-based integrator See Factor, from whom they had rented systems previously on a regular basis.
“When I started looking at the feasibility of this project, it was about becoming more competitive,” Auslan explains. “In looking at the pros and cons and crunching numbers, it became apparent there was really only one other venue our size in the city that didn’t have an installed system – our main competitor, Roseland Ballroom. When that closed it also was clear that it was time to up our game considerably.”
In addition to its elegant appearance, the Hammerstein Ballroom is noted for a very good acoustic design. Capacity depends on the configuration of the room. It seats about 2,500 for theatrical productions and musical performances, and several thousand for events held within a central ring.
The main floor is level, and the two main balconies, unusually close to the ground and gently sloped upward, seat approximately 1,200. There are also six shallow balconies on the sides, generally used by VIPs and celebrity guests. The 85-foot ceiling offers multiple rigging points for various audio, lighting and set design applications. It’s a decidedly multi-purpose space, with the addition of the system aimed at deepening the venue’s capability to provide top technology for musical artists as well as upping the level of flexibility for other applications.
Auslan headed up the project, with colleagues Robert Carvell (production department supervisor) and Roy Clark (chief of audio) also integral to the process.
The team began discussions with L-Acoustics, initially focused on implementing K1 arrays before shifting to the more compact K2 that better fit the room and application. A benefit of the L-Acoustics platform is that it’s a staple of major touring artist riders.
“Rider friendliness and support, how easy it is to get replacement parts, how robust it is – we thought about all of those things and L-Acoustics came up with high marks,” Auslan notes. There was a fair amount of interviewing other engineers about the K2, Clark adds.
However, several major brands offerings were evaluated for sound quality, price point and post-purchase support. Another key factor in the team’s decision-making process was the ability to ‘steer’ output, to direct it where it needed to be with a high degree of accuracy. “Once we factored in all of those things, the K2 emerged as the all-out winner,” Auslan says.
The K2 incorporates a combination of proprietary Panflex horizontal steering technology and Wavefront Sculpture Technology (WST) to heighten steerability, particularly important in being able to avoiding too much output reflecting off the ballroom’s large proscenium arch.
Left to right, production team members Michael Luo (Manhattan Center), Alban Sardzinksi (See Factor), Mark Friedman (See Factor), and Robert Carvell, Peter Auslan, Roy Clark and Billy Wong (all of Manhattan Center).
That proscenium definitely presents a challenge, notes Scott Sugden, L-Acoustics U.S. head of application, touring, who contributed design support on the project, utilizing proprietary SOUNDVISION 3D simulation software to help. “We spent a significant amount of time evaluating the design,” he says. “It’s a challenging space because it’s a modified theatre that they’ve essentially made larger by encompassing a certain amount of the stage space as audio space.
“The proscenium, however, is in the same place it was 60 years ago, and the PA is 40 to 50 feet upstage of that, so you’re shooting through a narrow window to hit the rest of the audience,” he continues. “The other difficulty is that the stage isn’t permanent. There are multiple formats for it so the PA had to be modifiable depending on the type of show they’re doing.”
The ability to differentially control the horizontal coverage for different parts of the array with the same box and consequently avoid the proscenium – to narrow the directivity of the boxes in a symmetrical and asymmetrical format – was integral to achieving optimized coverage.
“As the audience gets closer and we gain proximity, we’re able to make the PA wider on the onstage side, but not the offstage side, and can prevent energy from hitting this giant brick wall,” Sugden explains. “Generally speaking in a theatre environment, one of the biggest challenges is vertical coverage. You need to get 50, 60, 70 degrees in vertical coverage and the K2 is able to do up to 10 degrees between enclosures, which is quite a lot for a system of this size.”
A closer look at the L-Acoustics SB28 subs, usually on end, and KARA loudspeakers for fill.
The capacity of the ballroom varies, with two primary configurations available. Concerts with seating and standing room accommodate 3,500, while a diverse assortment of special events can be served with a flexible setup that can host up to 1,000. In addition, the venue’s 24-foot-deep “hard” stage suffices for most events, but it’s usually extended 16 more feet into the house for concerts. This results in two different positions for the main system’s left and right arrays, depending on the depth of the stage.
Moving the arrays, which are flown via CM motors with Motion Labs control, is relatively straightforward. “We change roughly one-third of the angles on the array, keep the rest the same and drop the rig using six motors to change position,” Clark notes. “I should also mention that the K2 does go down to 35 Hz and therefore has the ability to throw deep lows to the back of the room, which is helpful. So when guests walk in they hear amazing low end right away.”
The K2 arrays are made up of 12 boxes per side. “With that configuration, there’s the ability to set the horizontal dispersion differentially,” explains Chris Sullivan, application engineer, East Coast for L-Acoustics, who performed the system calibration in August. “The top three boxes are set to 70 degrees, which is the narrowest dispersion for the top two balconies. The bottom nine we actually go into asymmetrical output and are doing 90 degrees asymmetrical dispersion onstage. So that’s 35 degrees offstage and 55 degrees onstage, which is advantageous in avoiding the proscenium.
“Additionally, since the mains extend down to 35 Hz, we’re getting the majority of the musical content from them and low and high frequencies are similar in output across the venue,” he continues. “It also sounds more musical because the source, say the kick drum’s attack and impact, is coming from one place.”
A dozen L-Acoustics SB28 dual-18-inch subwoofers on the ground provide what Sullivan characterizes as “infrasonic” support: “Moving air, but not necessarily providing a whole lot of musical content, as we’re able to get most of that out of the K2.”
The subs are usually on end, six per side, tightly packed, but can be re-arranged depending on the room configuration, with a cardioid mode available to reduce the amount of back-firing energy concerts.
Up to six L-Acoustics KARA line source loudspeakers can be deployed for front fill. “Sometimes four of these function as front fills and two sit on top of the subs, so you can do the DJ version of ‘Texas headphones’ if you want,” Auslan notes. And, the stage is now served by up to 16 L-Acoustics 115XT HiQ active monitors and HiQ subs.
Five LA-RAK modular touring racks, each containing three LA8 amplified controllers, drive all main system loudspeakers and subwoofers. The mobile touring version of the racks, as opposed to the install version, were chosen for portability in meeting the needs of the different configurations.
Completing The Process
Both house and monitor systems are now headed by Avid VENUE Profile digital consoles, selected because they provide plenty of capability in addition to being rider-friendly. System integrator See Factor, involved with the project from the outset, also provided all equipment and implemented a custom fabricated snake and split system.
The new Avid VENUE Profile console at the venue’s front of house position.
“It’s an analog split because it needs to be with these consoles, and there’s a copper fan-out for the stage rack and then coax running out to the house console,” explains Mark Friedman of See Factor. Also on hand now are four Shure UR4D+ dual-channel wireless microphone systems and what Friedman terms an “industry standard” complement of Shure and Sennheiser wired microphones.
The project was completed in September of this year, with the new audio infrastructure making the venue eminently more functional for all manner of applications. “There are two main things to take into consideration in any installation,” Friedman concludes. “One of them is providing the most appropriate equipment to suit the venue and the room. The second – particularly for a venue that has multiple uses and applications – is providing something that’s as close to ‘all things to all people’ as possible.
“Obviously, you need the best fit from a technical standpoint, and then the most universally accepted products out there. In this case the K2 was truly the best fit for the room, regardless of market acceptance, and I dare say there’s probably not going to be one band ever that comes through there that’s not going to accept it.”
Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.
Church Sound: Using High-Pass Filters To Optimize Audio For Video
Let’s admit it, because it happens a lot. Even when you pay careful attention to the audio you record for a video, an use a good mic (you did use a good mic, right?), you can still end up with a bunch of background rumble and noise in your recording.
It happened just the other day to someone at the video production company I work for. They were shooting in a grocery store, capturing some interviews. They used a good shotgun mic, with good directivity to cut down on the ambient noise.
However, there were those dreaded coolers all over the store, and if you listen carefully (mainly because our brain normally tunes them out), you’ll hear compressors running. Back in the studio, it sounds like a truck going by the entire interview.
Because it’s a complex noise source, trying to run a noise reduction program on it probably won’t work well (and even when the noise goes away, it’s often replaced by unwanted digital artifacts of the FFT process used to perform the noise reduction—but that’s another post).
However, we do have one tool in our utility belt that can help (actually two, I’ll get to the second, which should actually be the first, in a second): The high-pass filter.
A high-pass filter (HPF) is just what the name sounds like—it lets high frequencies pass while blocking low frequencies. Basic HPFs are a simple on and off switch with a pivot frequency (the frequency at which it “passes” signal) and slope (how quickly it drops off the signal below the “pass” frequency) set at the factory.
Better HPFs that come with higher level editors like Premier Pro and Final Cut allow you to select the pivot frequency.
Here is an example of an HPF with a pivot frequency of 120 Hz, and a slope of 12 dB per octave (that is, at the frequency 1 octave below the pivot frequency—60 Hz—the level will have been reduced by 12 dB).
You can see how the frequencies above the pivot frequency pass by unaffected, while the ones below get rolled off pretty quickly. This is when it actually gets useful.
For the male voice, the fundamental frequency of the lowest notes one speaks is between 85-155 Hz. For a female, it’s a little higher, perhaps 165-225 Hz. This means that there is no real information that we need below 85 Hz for males and 165 Hz for females.
And in reality, because of the way we hear and the way the voice is produced, there are plenty of harmonic frequencies that our brain will interpret clearly to make up for missing fundamentals.
So let’s say we have a compressor running in the background of a female interview.
We can safely dial up a HPF with a pivot frequency of 165 and not lose any of her voice. We can take it up even higher to eliminate more of the noise, and the clarity will improve markedly.
In fact, the voice will “sound” louder once the low frequency stuff is removed because we can hear it better. So this is exactly what we did for grocery store woman. We dialed up an HPF with a pivot at around 150 Hz, and it totally transformed the audio.
There was still some higher frequency noise, and it was obvious she was standing in the store and not a studio, but the clarity of her voice was improved substantially.
Now, earlier I mentioned we actually have two tools in our tool belt. The other one may be on the mic itself. Many professional shotgun mics (and some interview mics, and the occasional lapel mic) have a HPF built in.
For example, my beloved Audio-Technica 835B has a switchable roll off at 180 Hz at 12 dB per octave.
That means at the lowest fundamental of a male voice the mic will be 12 dB down, which is generally not a big deal unless you’re interviewing James Earl Jones.
Normally, I like to leave this switched on because it eliminates a lot of room rumble, AC noise and other “nasties” right at the source. It’s just a good idea. If you use this when you shoot, you will require less processing in the edit suite.
Of course, you’ll want to listen to it through some good headphones first to make sure you’re happy with the sound.
You do have good headphones, right?
Mike Sessler now works with Visioneering, where he helps churches improve their AVL systems, and encourages and trains the technical artists that run them. He has been involved in live production for over 25 years and is the author of the blog Church Tech Arts.
Tuesday, December 16, 2014
In The Studio: Al Schmitt’s Microphone Approach
After 18 Grammys for Best Engineering (more than any other engineer) and work on over 150 gold and platinum records, Al Schmitt needs no introduction to anyone even remotely familiar with the recording industry.
Indeed, his credit list is way too long to print here (but Henry Mancini, Steely Dan, George Benson, Toto, Natalie Cole, Quincy Jones, and Diana Krall are some of them), but suffice it to say that Al’s name is synonymous with the highest art that recording has to offer.
Here’s an excerpt from my Recording Engineer’s Handbook that covers his usual microphone setup.
Do you use the same setup every time?
I usually start out with the same microphones. For instance, I know that I’m going to immediately start with a tube U 47 about 18 inches from the F-hole on an upright bass. That’s basic for me and I’ve been doing that for years. I might move it up a little so it picks up a little of the finger noise.
Now if I have a problem with a guy’s instrument where it doesn’t respond well to that mic then I’ll change it, but that happens so seldom. Every once in a while I’ll take another microphone and place it up higher on the fingerboard to pick up a little more of the fingering.
The same with the drums. There are times where I might change a snare mic or kick mic, but normally I use a D-112 or a 47 FET on the kick and a 451 or 452 on the snare and they seem to work for me. I’ll use a Shure SM57 on the snare underneath and I’ll put that microphone out of phase. I also mic the toms with 414’s, usually with the pad in, and the hat with a Schoeps or a B&K or even a 451.
What are you using for overhead mics?
I do vary that. It depends on the drummer and the sound of the cymbals, but I’ve been using M 149s, the Royer 121s, or 451s. I put them a little higher than the drummer’s head.
Do you try to capture the whole kit or just the cymbals?
I try to set it up so I’m capturing a lot of the kit, which makes it a little bigger sounding overall because you’re getting some ambience.
What determines your mic selection?
It’s usually the sound of the kit. I’ll start out with the mics that I normally use and just go from there. If it’s a jazz date then I might use the Royers and if it’s more of a rock date then I’ll use something else.
How much experimentation do you do?
Very little now. Usually I have a drum sound in 15 minutes so I don’t have to do a lot. When you’re working with the best guys in the world, their drums are usually tuned exactly the way they want and they sound great, so all you have to do is capture that sound. It’s really pretty easy. And I work at the best studios where they have the best consoles and great microphones, so that helps.
I don’t use any EQ when I record. I use the mics for EQ. I don’t even use any compression. The only time I might use a little bit of compression is maybe on the kick, but for most jazz dates I don’t.
How about mic preamps? Do you know what you’re going to use? Do you experiment at all?
I know pretty much what I’m going to use. I have a rack of Neves that I’ll use on the drums.
How do you handle leakage? Do you worry about it?
No, I don’t. Actually leakage is one of your best friends because that’s what makes things sometimes sound so much bigger. The only time leakage is a problem is if you’re using a lot of crap mics. If you get a lot of leakage into them, it’s going to sound like crap leakage, but if you’re using some really good microphones and you’re get some leakage, it’s usually good because it makes things sound bigger.
I try to set everybody, especially in the rhythm section, as close together as possible. I come from the school when I first started where there were no headphones. Everybody had to hear one another in the room, so I still set up everybody up that way. Even though I’ll isolate the drums, everybody will be so close that they can almost touch one another.
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 go here for more info and to acquire a copy of The Recording Engineer’s Handbook.
Sunday, December 14, 2014
Back & Better Than Ever
Fall Out Boy is back, and as this year’s recently concluded months-long concert tour demonstrated, the American rock/punk quartet is more popular than ever. The tour comes following the band reuniting after going on a hiatus, in advance of the release of its sixth studio album, American Beauty/American Psycho.
Originating in Chicago’s punk scene, Fall Out Boy formed in 2001 and is still comprised of original members Patrick Stump (guitar), Pete Wentz (bass), Joe Trohman (guitar), and Andy Hurley (drums). All players contribute vocals, with Stump taking the lead role in that regard.
In the process of reuniting, the band also sought to reinvent its sound, focusing a bit more on a pop style. Two veteran touring engineers were on hand to support that effort: Chad Olech (Alice in Chains, Deftones, P.O.D., Thin Lizzy) at front of house and Pasi Hara (System of A Down, Slash, HIM, Serj Tankian) at monitors. And as with all previous Fall Out Boy tours, Clair provided the systems and tech support.
Olech’s been working with the band for more than a year, and he called in Finland-based Hara to handle monitors while the tour was in Europe. Both agree that it’s a great group to work with, and that the mantra, shared by the artists, is to keep things straightforward.
Engineers Pasi Hara (left) and Chad Olech at front of house with an Avid VENUE Profile console. (Credit: Pasi Hara)
Methods Of Capture
A long-time staple of Fall Out Boy’s gear approach is Shure microphones and wireless systems, and that continues. Stump utilized a KSM9 capsule in tandem with a UHF-R wireless system, which he’d switched to prior to Olech and Hara coming onboard. Despite that, the two engineers worked with him in evaluating several other options, but they all kept coming back to the KSM9.
“This mic works well with Patrick’s tonality,” Olech explains. “He has a pretty strong upper-midrange voice that the KSM9 fully captures,” with Hara adding, “In my experience there aren’t many people that this mic does not work well with. The KSM9 is a really, really good microphone.”
Bassist Pete Wentz using his tried and true SM58. (Credit: Steve Jennings)
As they’ve done for years, Wentz and Trohman sang into hard-wired SM58s, while drummer Hurley was supplied with a BETA 56A supercardioid dynamic that works with his vocal signature while being compact and unobtrusive, mounted on a boom with a gooseneck for extra control of positioning.
A relatively new development was that the guitar amps were miked – previously they had been taken direct. Initially this was handled with KSM32 side-address cardioid condensers before a transition to KSM313 bi-directional ribbons that everyone was quite pleased with.
Hara used a single mic on each amp, positioned at the center. Bass was taken direct via a Tech 21 SansAmp, with no amp cabinet on stage.
The drum kit was captured with a largely rock n’ roll approach. A BETA 91A was inside the kick and a BETA 52A was in the hole, with Olech applying a bit of Waves InPhase plug-in at front of house to put the two in sync.
A dual mic method was also taken on snare, with a BETA 56A on top and a KSM32 on bottom, and they were also phase aligned with an assist from the plug-in. “The Waves tool makes a world of difference,” Olech notes.
BETA 98A miniature cardioid condensers were applied to the toms, another mic drawing high praise from Olech. Meanwhile, overheads were technically underheads, with KSM137 end-address condensers positioned closely under both of Hurley’s crash cymbals. The same went for ride and hi-hat.
Keeping It Simple
Both engineers did their mixing on Avid VENUE Profile consoles that have been another tour staple. Neither goes over-the-top in terms of processing, with Olech noting that he utilized Phoenix CraneSong as well as Waves C6 and SSL ChannelStrip plug-ins to tailor Stump’s vocal mix, completing it with a bit of reverb and delay.
Fall Out Boy in concert on the recent tour. (Credit: Steve Jennings)
Hara inherited the previous show file, and stripped back the processing to an extent. The goal was keeping each monitor mix tight and with minimum latency. “These guys want to keep it simple, have their own instrument and vocal on top of their monitor mix, and everything else significantly below that,” he explains. “Other than that, there’s some click and stereo backing track blended in, and that’s it.”
The Fall Out Boy stage was devoid of loudspeakers – not a wedge, side fill or even drum “thumper” to be found. It helped keep down overall noise and wash, contributing to a very clean, distinct sonic signature in the house.
All of the musicians wore JH Audio JH16 Pro in-ear monitors fed by Shure PSM 1000 wireless personal monitoring systems. Hara is forthcoming in his praise of the PSM 1000s. “They’re the most solid systems in terms of RF that I’ve ever worked with, and provide excellent sound that can be punchy when you want it,” he says. “At this time they’re really hard to beat.”
Hara was responsible for all RF coordination on the tour, with more than 30 active channels (plus spares) to get in line at each stop. This included the Shure guitar wireless systems favored by Wentz (UR4D), Stump (also UR4D) and Trohman (ULXD4Q). He performed frequency scans prior to each show, utilizing the Shure Wireless Workbench platform and Axient 600 spectrum manager to help get it all coordinated.
The tour’s Clair i5 arrays deployed at a stop on the tour. (Credit: Pasi Hara)
Hara and Olech both enjoy long-standing working relationships with Clair, further aligning their goals and those of the band.
Clair provided i5 full-range line array modules joined by i5b low-frequency array modules, typically deployed 12 per side to cover the amphitheatres and arenas visited by the tour. Lower octaves were reinforced by 24 BT-218 dual-18-inch subwoofers on the deck.
Olech notes that the draw at each venue was exceptionally strong, with mostly sold-out shows. “It’s kind of interesting,” he concludes. “The band went away for a few years and actually managed to grow their fan base when they returned. They’re also a real pleasure to work with, and our vendor support was amazing. That kind of enthusiasm makes for a great tour.”
Friday, December 12, 2014
Church Sound: The Perception Of Loudness Versus The Reality
It seems like every week I end up in a conversation about volume levels and that “so and so” thinks it’s too loud. Today, let’s focus on our perception of loudness.
In the space where I mix or act as producer almost every Sunday, we hold two very distinct services. The early service is more “traditional” (whatever that means) with brass, strings, organ, choir, and a worship team of vocalists.
The second service is all-out contemporary praise and worship with a little edge to it. The typical set includes drums, bass, acoustic, lead guitar, keyboard and one or two vocalists, and the musicians have mostly grown up in the 1980s and 90s, so big guitar (also used to be big hair) tend to rule the day.
We try to make sure sound pressure levels for this service don’t get out of hand, and to help get a better grip, I’ve started setting up a dB meter for every service to keep an eye on the average levels. My friend Trevor, who mixes these contemporary services, refers to it as the speedometer (and I’m not sure if he means that in a positive way).
Recently I decided to also use the meter during the traditional services so we could have some comparison. I’d expected the contemporary service to be 15 dB hotter than the traditional service, so was surprised that on certain Sundays (when the brass played), the traditional service actually hit peaks slightly hotter than the contemporary service!
For consistency, I monitored both services using a dB A-weighting. The music in the traditional service averaged around 82 dBA, with peaks running around 88 dBA. The contemporary service averaged around 85 dBA with peaks in the low 90s.
With just 3 dB separating the level of the two services, why was nobody complaining about the levels in the traditional service?
One obvious reason is that A-weighting was developed for lower volumes, and the frequency response is weighted to the human ear. This means that the low-frequency energy generated by the bass and the kick drum in the contemporary service does not register as much on the A weighted scale.
Why A-weighted measurements? OSHA loudness exposure ratings are based in A-weighting. Really, the scale of weighting, A or C, is not all that important as long as the same one is used every time—AND—we understand what the weighting represents.
But I believe the main reason we don’t get complaints is due to musical style. In the traditional service, the organist playing an offertory can easily hit 90 dBA, and people will say it is “powerful.”
In addition, the peaks in our traditional services are generally very short lived—for example, the trumpets will hit an accent note that jumps out but it’s gone in an instant.
So I maintain that it really comes down to taste and the composition of the music. I usually enjoy a 90 to 95 dBA contemporary Christian concert, but wouldn’t stay in the room for a heavy metal band playing at that same level.
My parting suggestion would be to document the levels of your services so that you have a consistent reference point to refer to. This comes in really handy when the inevitable and constant subject of volume levels rears its head again… and again… and again…
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.
In The Studio: Eight Key Mixing Mistakes—And How To Avoid Them
Most recording musicians, engineers and producers are well aware what a difference mastering can make to our mixes. And as we’ve discussed in previous columns (such as Audio Mastering Basics: Taking Your Music That Extra Step), mastering is an art form in itself, and is best placed in the hands of a specialist.
But even expert mastering engineers can only accomplish so much, and it’s largely dependent on the raw materials they’re given to work with. With that in mind, here’s a look at some of the top mistakes people make in preparing their mix for mastering, with the help of veteran mastering engineer of Universal Mastering Studios West, Pete Doell.
1. Too Much Bottom
Excessive low-end is probably one of the most common problems in mixes coming from project studios. Usually this is directly related to the mixing environment. The average home studio or project room is lacking in real acoustical treatment is and rife with reflective surfaces and bass traps.
The result is an uneven response across the bass spectrum, with some notes being overemphasized and others being practically inaudible. This translates to a poorly balanced low end in your mix.
Doell offers an important pointer: “The most egregious mistake is that people’s monitors aren’t placed properly,” he says. “Speakers need to be as far apart from each other as you are from them. So if your mix position is, say, three feet from either speaker, the speakers should be exactly three feet apart. Moreover, if the speakers are too close or too far from a wall, the apparent bass response will be off.”
2. Terrible Treble
On the other end of the spectrum, high-end can also cause its own issues. While not as hard to hear in the project studio environment, those high frequencies can show up differently during the mastering phase.
A de-esser, like the Precision De-Esser Plug-In, is a good way to nip sibiliance in the bud before mixing.
“Most mixes will want a bit of ‘polish’ or ‘shine’ in mastering,” says Doell. “When this good stuff is applied, sibilance can really creep up. Do yourself a big favor and de-ess your vocals, maybe even your hi-hat just a bit, even if you don’t hear too much of an issue. Your mastering engineer will thank you.”
The bottom line, as Doell points out, is to use EQ wisely and sparingly.
3. No Dynamic Range
This is probably one of the most discussed topics in modern music mixing circles. Over the past decade or so, the quest for radio airplay has created a battle for attention that has manifested itself in loudness – the perception being that louder the track, the more it will grab the listener.
It’s a mentality that started with TV and radio advertisers (notice how a loud commercial gets your attention) and is a direct result of today’s vastly improved compressor technology, which has enabled us to create “radio mixes” where everything is loud, punchy and in your face.
The problem with pumping up the apparent volume on your mix this way is that it works by compressing the dynamic range of your tracks. Dynamic range is defined as the difference between the loudest and softest sounds in your track.
Ideally, the tracks you deliver to the mastering house should have peaks of around –3 dB for the loudest material (for example, a snare hit), while the rest of the track should average in the –6 dB to –8 dB area. That would give your peaks somewhere around 3 dB to 5 dB of dynamic range.
The problem with compressing dynamic range (or, equally hazardous, normalizing a track’s relative volume), is that you effectively rob your mastering engineer of the resources to do their job.
A good mastering engineer applies meticulous use of multiband compression – bringing up the punch and presence of the bass, adding clarity and sparkle to the high end – all by using different compression algorithms for different spectral bands.
Many inexperienced mixers will apply a “mastering compressor” plug-in, using a preset that creates a loud but muddy low-end, a bright and aggressive high-end, and little room for the mastering engineer to add — or de-emphasize — anything.
“Sometimes clients desire a ‘loud’ mix, but they have done little or nothing to control the dynamics of their mixes,” says Doell. “I like the analogy of getting a super sexy paint job for your car — asking the mastering engineer to do the entire job with one ‘coat of paint’ is not the smartest move. Layering the limiting (by compressing the vocal, bass, snare, for example) will allow a MUCH more gorgeous detailed, deep shine on the final product!”
On a related note, try to avoid over-compressing individual tracks for the same reason. Often a mastering engineer will get a track that’s well within dynamic range, but with a vocal track that’s been normalized to the verge of distortion. Again, it leaves little room for mastering to bring out any subtlety or nuance in that vocal.
4. Lack of Panning
It’s important to give your mix some dimensionality by balancing different elements within a nice, wide, stereo field. All too often, people tend to pan everything at or near the center, creating a cluttered-sounding mix that lacks definition. While certain elements should typically be centered (kick, snare, vocal and bass come to mind), panning is a great way to achieve separation between guitar parts, background vocals and other parts of the mix.
“It’s always good to pan some elements of the mix just a bit off to one side,” says Doell. “If you have a blend of guitars, horns, backing vocals, etc., keeping the middle less cluttered allows your ear to hear more distinctly all of that cool production you’ve worked on. You’ll also need less EQ and effects to pick these things out in the mix.”
5. Phase Problems
With most DAWs offering unlimited tracks, the temptation to record everything in stereo is strong, and elements like a nicely-recorded stereo acoustic guitar can add depth and character to a track.
But be careful to check your mixes in mono to avoid phase cancellation from poorly-placed mics. Only by soloing the stereo tracks will you be able to hear whether certain frequencies “disappear” when the two channels are summed to mono.
Taking a moment to check and correct phase issues as you go will head off lots of problems down the road.
It’s not just stereo-miked instruments that can fall victim to phase cancellation. According to Doell, “Often I’ll get a track with ‘hyper-wide’ elements in the mix that achieve that ‘outside the speakers’ effect by making one side out of phase. Just try hitting the mono button and watch that cool keyboard, string pad, background vocal stack, whatever, totally disappear. Even if you never anticipate having any need for mono (AM radio anyone?), when you do this, your balances aren’t what you think!”
This same principle also applies to reverbs. It’s all too common to have that lush hall you placed on the vocal just vanish in mono.
6. Poor Vocal Placement
It’s hard to be objective on placing vocals in a mix, particularly if it’s your song. After all, you know the lyrics, so it’s easy to forget that other people don’t.
And in most cases, a track can sound equally “right” whether the vocal is sitting a bit in front or a bit behind the track. Many pros will do two or three alternate mixes of a track, one with the lead vocal a bit up, one with it a bit down, and one in the middle. It’s a luxury of choice that most mastering engineers are happy to have.
7. Misaligned Tracks
This one is a no-brainer. When you send stems (separated groups of tracks, like drums and bass, guitars, backing vocals) to mastering, make sure they all start at the same place. “This is another pet peeve of mine,” says Doell. “If the lead vocal doesn’t come in until 0:30, that stem should have 30 seconds of silence at the top.”
8. Not Knowing Your Room
“I always like to start my mixing day by listening to some records I know and love — ideally in the musical style I will be working in — in the seat I will be sitting in to mix, and over the same D/A converter,” says Doell. “Then I will be much more readily comparing apples with apples. I am blessed to work in a ZR Acoustics (Zero Reflection Acoustics by Delta H Design, Inc.) room at Universal Mastering. But if I am working elsewhere, it’s important to know how the room I am working in is participating in what I am hearing, before I start making any decisions.”
As you might imagine, there are countless other stumbling blocks that can trip up your mix and make life challenging for your mastering engineer – certainly far more than we can list in this column. As always, the bottom line is to use your ears, listen carefully, and learn the rules before you break them. If all else fails, keep the potential mistakes above in mind, and you’ll be on your way to better results.
For more mastering tips and tricks, check out Pete Doell’s Mastering Tips with Precision Mastering Plug-Ins.
Daniel Keller is a musician, engineer and producer. Since 2002 he has been president and CEO of Get It In Writing, a public relations and marketing firm focused on audio and multimedia professionals and their toys. Despite being immersed in professional audio his entire adult life, he still refuses to grow up. This article is courtesy of Universal Audio.
Thursday, December 11, 2014
Are Tech-Trained Millennials The Right Hire For AV Integrators?
Been to an InfoComm show lately? If you have, something you may notice is that the average age at the show is, well, let’s be PC and say not young.
For many that have been around the business for some time, they may not have noticed the age creep taking place, but for many who come year after year and see the same faces you may have seen a wrinkle or two that wasn’t there before.
Nevertheless, the industry is aging a bit and now is the time for owners and leaders to start thinking: how can we bring some youth back? How can we inject a little technology enthusiasm and get those darn kids to take their knowledge of computers, tablets and social media and turn it into something good for our business?
Begging the question, is it time to start looking for tech savvy millennials to come into our businesses and help us evolve?
Sure, the reputation of the younger generation isn’t great. I think most would agree that with each new generation that comes there is one that came before that is touting why the next generation will fail. Perhaps it is generation-centrism or maybe just a lack of understanding of what is next, but it is true. Right boomers? You guys had little confidence in Gen X.
So let’s get past the stereotypes that have millennials being basement living, entitlement crazy, grammar lacking hethens and let’s look at why millennials may solve more problems for most integrators than they create.
It Starts With The Tech Revolution
By in large the biggest problem that the integration industry has right now is that it is suffering with its identity. Are we A/V or are we IT? Does A/V as we know it really exist and if so how much longer will it be around?
We all know that mobile, social and cloud are going to be the determinants of what is next for tech, so how many more electronic whiteboards and projectors can we really sell? And while claiming the death of certain technology is certainly premature, the fact is anyone who can spell their name right (on a mobile device) knows that the way tech is being used is changing.
However, for many integrators, those making the technology decisions are less than comfortable with the changes. Few, if any, regularly do video on their mobile device and most haven’t seen just how good, free and very inexpensive collaboration services work.
But guess who does use these things every day? Guess who knows the upside and the downside of these applications? And furthermore who has a keen awareness of the privacy and security issues that most of these emerging free(ish) services offer?
That is right, millennials!
By instilling a little bit of youth into your organization, you are opening the door to adding team members who may not remember the old days when projectors drove 40 points of margin, but they will know all about emerging tech and why people want it. Frankly speaking, that is what most integrators need.
The story about why we as AV integrators are needed is changing somewhat. The ability to present, communicate and collaborate hasn’t changed but the ways and means are changing at breakneck speed that the industry is struggling to keep up with.
Mobilizing youth and infusing it into your organization isn’t only important, but may be the key difference between the companies that make the needed leap to what’s next and those that get left in the dust.
So where are your millennials? Is it time to start thinking young?
Daniel L. Newman currently serves as CEO of EOS, a new company focused on offering cloud-based management solutions for IT and A/V integrators. He has spent his entire career in various integration industry roles. Most recently, Newman was CEO of United Visual where he led all day to day operations for the 60-plus-year-old integrator.
Go to Commercial Integrator for more content on A/V, installed and commercial systems.
Alternatives To The Ol’ Ball & Stick
When I began working in pro audio, I pretty much copied what everyone else was doing when it came to microphone selection and placement – “ball” mics for vocals and “stick” mics for instruments and amps, with hardly any “studio” mics on stage except when live recording was being done.
Then came a show with an older soundguy who proceeded to mike the stage in a very strange way, or at least it was strange to me. He put a ball mic on snare and toms, and then grabbed a 1940s-era RCA Varacoustic, set it to a figure 8 pattern, and positioned it a few feet away from the congas and bongos, pointing the rear of the mic off stage.
I asked him about both approaches. He replied that the ball mics could take a beating from errant drumstick hits better and that the ball could be easily replaced, unlike the head of the most popular stick mic.
And he showed me how the figure 8 setting picked up in a very narrow pattern front and rear while providing very good rejection at the sides, with the rear pickup pattern pointed where it was quiet, so now he would hear only the percussion in that one mic.
It made sense to me, and sure enough at sound check, the drums sounded as good as I’d ever heard and the percussion mic worked like a charm.
The experience encouraged me to think outside the box (or more accurately, outside the ball and stick) and really focus on the end result rather than defaulting to the most common way of doing things. Over the years I’ve discovered (and “borrowed” from others) some approaches that are particularly useful in challenging situations.
Sometimes it’s all about placement, but there are also several unique mics that can help get the job done. Let’s take a look.
Heil Sound PR 31BW
Larger & Smaller
Want a large diaphragm dynamic but the size of a standard unit is too big? Take a look at the Heil Sound PR 31BW, a shortened version of the PR 30 developed in collaboration with Bob Workman, the front of house engineer for the Charlie Daniels Band (hence the “BW” designation).
The two models both have a large element, humbucking coil, and share the same specs, but the PR 31BW is only 4 inches in length, perfect for mounting to drums, placing underneath cymbals, or even inside a Leslie speaker cabinet. Heil also makes the Handi Pro Plus that’s again just 4 inches long, so it can come in handy (see what I did there?) in many tight-space applications.
The Electro-Voice N/D468 dynamic supercardioid instrument mic has a swivel head that provides a wide range of placement options, perfect for drums, horns and guitar amps. The large (2.05-inch) diaphragm delivers across a very wide frequency response of 20 Hz–22 kHz and can handle high SPL.
Telefunken has a shorter version of the M80 called the M80-SH. Both have the same low-mass capsule and super-thin capsule diaphragm that offers a wide frequency response of 30 Hz–18 kHz. The smaller SH version also ships with an XLR cable with right angle plug to further reduce length, so it’s a choice for drums, percussion and vocals, and specifically, could be applied for singing drummers who don’t want a lot of bulk behind a vocal mic that can get in the way of their playing.
DPA d:vote 4099
DPA Microphones offers a range of very good solutions for tough situations. The d:vote 4099 condenser is a very small supercardioid that fosters quality capture of drums and percussion instruments, capable of handling high SPL. It’s so low profile as to be almost invisible, with a flexible clip-on system providing fast, stable and repeatable attachment and a flexible gooseneck that can be positioned at different angles to fit different drums and allow a variety of sound nuances.
And by detaching the clip from the gooseneck and re-mounting it turned 90 degrees, the number of mic positions is doubled. Note that this mic is also a solution with a variety of other instruments, such as string and wind instruments, and piano, with a variety of tailored mounting options offered.
And if you need a very small mic for drums and don’t have phantom power available, or you just prefer the sound of a dynamic element, the Sennheiser e 608 provides an answer.
The supercardioid element is attached to a clamp and gooseneck mounting for positioning. Extensive damping and shock mounting isolates the capsule and signal from extraneous vibration, noise and impact, and a humbucking coil protects against induced electrical interference.
Another option along these lines is the beyerdynamic TG D52d. While small in size, this little dynamic can handle very high pressure levels and is not very noticeable around the kit.
Hanging & Standing
Small microphones such as the Countryman ISOMAX 2-H provide a workable and often very good solution for choirs, large vocal groups and orchestras. They can be hung above the performance area while remaining pretty much invisible.
The ISOMAX 2-H is available with a choice of omnidirectional, cardioid and supercardioid patterns (and a choice of black or white color), offering a wide frequency response and the ability to handle high SPL levels when necessary.
Sennheiser e 608
Along these lines, Audix Micro Series mics – along with their MicroBooms – come in handy for stages and other situations where it’s not possible to hang mics. These carbon fiber booms come in 24-, 50- and 80-inch lengths, attach to any mic stand, and have a built-in gooseneck.
They can be used with M1250, M1255 and M1280 mics, with elements available in omni, cardioid, hypercardioid and supercardioid (shotgun) patterns. The result is a clean, elegant look that puts the mics into suitable positions.
A frequent problem, particularly with school and church choirs, occurs when soloists come forward to sing and don’t stand directly in front of the solo mic, or they grab it off the stand and don’t point it directly at their mouth. A cardioid pattern mic used to be my choice for solos, offering a fairly wide pickup pattern, but a better choice might be the Shure KSM9HS handheld vocal mic that offers both a hypercardioid and a subcardioid pattern.
Basically, a subcardioid pattern is a wider cardioid pattern that still offers sufficient rejection on loud stages. So particularly with the subcardioid setting activated, the KSM9HS works well for miking a group or for solo vocalists who go off-axis. It’s available in charcoal and champagne colors, and is also offered in a wireless version.
Speaking of wireless, my company handles a lot of corporate speeches, and a big challenge is getting good gain before feedback when using wireless headworn mics in a room where the PA consists of ceiling loudspeakers, with some that can’t be turned off located directly over the stage.
The subcardioid pattern of the Shure KSM9HS
The Audio-Technica MicroSet BP894 directional microphone can help, with a rotating cardioid capsule that can be aimed directly at the mouth of the wearer for higher gain before feedback and very good rejection of noise.
The lightweight unit hooks securely behind either ear, or for maximum stability, there’s also a dual-ear mount. It’s available in black and beige colors, and in both wired and wireless versions.
New & Old
Fans of Dante networking should be pleased with another Audio-Technica development, the ATND971 wired mic that transmits audio and control data together over Dante network protocol.
This boundary design is well suited for boardrooms, meeting spaces, podiums and any place else needing a low-profile approach.
The mic also includes an integrated programmable switch that can control anything from triggering a video camera’s pan/tilt to controlling a room lighting preset.
Now, A-T has followed this up with the just-introduced ATND8677, a mic desk stand that fosters Dante of any phantom-powered condenser gooseneck mic with a 3-pin XLRM-type output. Further, it could actually be used with any mic by simply plugging the mic’s cable into the desk stand.
As you may have gathered from the articles I’ve written about my mic collection in previous issues, I love vintage models. Unfortunately most older mics aren’t usable because of deterioration or just the fact that their performance is nowhere near quality we demand today.
Luckily, a few manufacturers make modern mics with a vintage vibe that’s currently popular with a lot of performers.
Shure never stopped production of the iconic model 55 and today the body shape is available in two models, the 55SH cardioid and Super55 supercardioid. MXL debuted the CR77 dynamic stage vocal mic last year. with a “pill” shape evoking the styling of an old RCA type 77, with a distinctive black chrome, perforated metal grill and matte black body.
Ear Trumpet Labs Chantelle
Ear Trumpet Labs is a newer manufacturer that makes mics with a great retro look. You might have spotted Faith Hill singing into an ETL Chantelle during a duet with Tim McGraw, who was using an Edwina model, at the recent 2014 ACM awards. Not only are the mics cool, but so are their names.
Finally, the Granelli G5790 is one of those problem solvers that prompts the question, “Why didn’t I think of that?” Well it’s because two recording engineers beat you (and me, and all of us) to it.
Seeking easier positioning of a Shure SM57 around a crowded drum kit, they modified a standard 57 by giving it a bent shape, and after a few prototypes, they were able to attain the same sonic signature as the original.
A lot of audio folks liked the idea, so they began offering them for sale. You can purchase a modified SM57 (and SM58) directly from them, or you can purchase a DIY conversion kit and modify your own 57 or 58. Either way, the result is a mic that can tuck into tighter spaces with the same sound as the original.
Now why didn’t I think of that?
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.