Just when we all adjusted to 2009’s FCC regulations involving the use of wireless microphones and mobile devices in the TV band, along comes a new wrinkle: the incentive auction.  This is not, repeat not, cause for alarm.  In this post, we’ll break it down into its simplest elements – explain the basic concept, survey the changing landscape, share the expected timeline and diffuse some common misunderstandings about how a spectrum change is likely to affect you and your wireless gear.

The FCC is under intense pressure to reallocate even more spectrum for mobile broadband. The rising popularity of smartphones and tablets has placed an additional strain on the currently available spectrum.  Consider, for example, the fact that smartphones use 35 times more spectrum than traditional cell phones and tablets use 121 times as much.  So the FCC is looking for innovative ways to open up additional spectrum, with the loftier goal of “spurring economic growth and maintaining the country’s mobile leadership.” Legislators are aware of these issues and recently took action to address them.

The bill granting special authority to the FCC was signed into law on February 2012.  It gave them permission to launch a one-time incentive auction that will repurpose some of the broadcast television spectrum. The basic premise is pretty simple: broadcast stations can elect to give up some of their assigned spectrum (either by moving to a different channel, sharing a channel with another station, or going off the air entirely) and offer them, through the FCC, to the highest bidder.  When the incentive auction is complete, the FCC will reallocate the cleared portion of the TV band spectrum to the auction winners.

How it Works. Courtesy: FCC LEARN site

Forward, Reverse and Repacking

Here are some terms you are likely to run across in coverage of the incentive auction.

Reverse Auction

The reverse auction is where broadcasters will establish their price to voluntarily relinquish spectrum rights in exchange for a portion of the proceeds from the forward auction.

Forward Auction

The forward auction is where the potential users of repurposed spectrum bid for new flexible-use licenses. The FCC is familiar with forward auctions in the spectrum context, and has been conducting them for nearly two decades.

Repacking

Repacking involves assigning channels to the broadcast television stations that remain on the air after the incentive auction… This process considers only a reassignment of channels, not geographic moves of stations; however, a station that opts to channel-share may have to move its antenna to a new geographic location – i.e., its sharing partner’s tower.

NRPM

This is the Notice of Proposed Rulemaking that forces government agencies (FAA, FCC, EPA for instance) to listen to comments and concerns of people whom the regulation will likely affect. The Notice detailing the FCC’s options for the auctioning of TV band spectrum “Expanding the Economic and Innovation Opportunities of Spectrum Through Incentive Auctions” runs 200 pages and included milestones shown in the timetable below.

Incentive Auction Timeline. Courtesy: FCC LEARN site

The Major Players

Wireless carriers are the primary proponents of the repack and auction plan, and, as the forward auction bidders, will be the likely beneficiaries.  Not to be overlooked is a profit opportunity for TV stations and the government. Some of the auction proceeds (estimated at approximately $25 billion) will be shared with TV stations participating in the reverse auction, some will be used to build a nationwide public safety communications network in the 700 MHz Band, and costs incurred to broadcasters from repacking will have to be reimbursed. The remainder of the proceeds will be deposited in the U.S. Treasury.

On the other side, the tech industry – Google, Microsoft, the WiFi Alliance, among others – cite the danger of putting power in the hands of a very few, very large companies who can afford to license their slice of the spectrum and use it for only those devices and services they market.  This group believes that free and open access stimulates innovation, investment, and job creation, as companies of all sizes develop new products and services.  They are lobbying for portions of the Band to be set aside for unlicensed use — similar to the allocation of the 2.4 GHz band for Wi-Fi and Bluetooth and the recent White Spaces initiative.

Professional audio interests, from trade organizations such as the National Association of Broadcasters to the country’s major sports leagues to the largest content producers, speak to the impact on wireless microphone users.  These entities remain vocal in their position that they have recently given up over 100 MHz of spectrum during the 700 MHz band reallocation, and that wireless microphone operation in the TV Band must be protected going forward.

http://apps.fcc.gov/ecfs/document/view?id=7022130172

http://apps.fcc.gov/ecfs/document/view?id=7022130256

Shure is at the forefront of efforts to maintain adequate spectrum for professional audio and is actively involved in the Incentive Auctions proceeding.

How It Affects the Wireless Microphone User Today

Since there’s a risk of confusion for wireless users, Shure Notes asked Chris Lyons, who has been leading training sessions at Shure on the general subject of spectrum issues, for his insights.

Which spectrum will be auctioned?

The FCC has suggested a few different sections of spectrum for the auction.  The most likely is a section in the upper part of the existing TV band, beginning at TV channel 51 and extending downward.  How much spectrum is auctioned depends on how many TV stations volunteer to participate, which will likely vary in different cities.

When will the auction take place?

The FCC hopes to conduct the auction in 2014, but has not announced a specific date.  Commissioners have stated clearly that this is the most complex spectrum auction in world history and could be subject to unforeseen challenges.

Assuming that enough TV stations participate and the incentive auction takes place, how long will it take the FCC to repack the TV Band?

Repacking the TV stations into a smaller TV band is the hard part of the process, and could take a few years.  The repacking process is as complex as the DTV transition was, which ended up taking 10 years.

Will any pro audio manufacturers take part in the forward auction?

No, it’s not feasible (financially or technically) for a manufacturer like Shure to own and administer a piece of spectrum solely for its own users.

In 2009, wireless systems that operated in the 700 MHz band were no longer allowed.  Could reallocation of the TV Band make today’s wireless systems obsolete?

Wireless microphones (as well as personal monitors, production intercoms, and similar gear) that are in the spectrum that is auctioned will have to stop operating at whatever date is set by the FCC.  This means that those systems will need to be replaced with units that operate in the spectrum that is still open for wireless mic use.  Failure to comply with FCC rules (just like IRS or EPA rules) is illegal and subject to enforcement action.

Should I be doing anything now to get ready for post-repack landscape?

At this point, the most valuable thing to have is information.  Every facility or venue that uses wireless microphones needs to appoint someone to keep up with this issue as it progresses.  If you own equipment in the upper part of the TV band, it would be wise to begin budgeting for replacement equipment, so that you’ll be ready to act when the available TV channels in your area are finalized.

How can I stay informed of what’s going on?

The FCC’s LEARN Program is quite user-friendly, and spectrum issues are always reported on their blog.  Shure’s website and publications also report significant developments.  And our Product Support department is always a good resource for guidance.

Contributors:

John Chevalier
Bill Gibson
Frank Gilbert
June Millington
Dan Murphy

“John had a semi-acoustic Gibson guitar.  It had a pickup on it so it could be amplified. We were just about to walk away and listen to a take when John leaned his guitar against the amp. He really should have turned the electric off.  It was only on a tiny bit and John just leaned it against the amp when it went ‘Nnnnnwahhhh!’  And we went, ‘What’s that? Voodoo?’. ‘No, it’s feedback.’  ‘Wow, it’s a great sound!’ George Martin was there so we said, ‘Can we have that on the record?’ It was a found object, an accident caused by leaning the guitar against the amp.”   

- Paul McCartney
(Source: Many Years From Now, Barry Mile)

It’s pretty much common knowledge among students of pop music that The Beatles’ 1964 recording of “I Feel Fine” was one of the first known examples of feedback as a recording effect, even though The Kinks and The Who reportedly (and intentionally) used it in live performances.   For most musicians and engineers, though, audio feedback is something to avoid.

In this post, we’ll cover some of the fundamentals – what causes feedback and how to avoid it – along with tips from some of our favorite audio pros.

What is acoustic feedback?
Acoustic feedback occurs when the amplified sound from any loudspeaker re-enters the sound system through any open microphone and is amplified again and again and again.  We’ve all heard it – it’s that sustained, ringing tone, varying from a low rumble to a piercing screech.

And what causes it?
The simplest PA system consists of a microphone, an amplifier and one or more speakers. Whenever you have those three components, you have the potential for feedback.  Feedback happens when the sound from the speakers makes it back into the microphone and is re-amplified and sent through the speakers again, like this:

Here’s an example: Let’s say that that you place the microphone in front of the speaker as shown here. If you tap the microphone, the sound of the tap goes through the amplifier, comes out the speaker and re-enters the mic.  This feedback loop happens so quickly that it creates its own frequency, and that produces the howling sound — an oscillation triggered by sound entering the microphone. Placing the microphone too close to the loudspeaker, too far from the sound source, or simply turning the microphone up too high all raise the likelihood of feedback problems.

Pro Tip #1: “The worst is vocalists who cup the mic capsule (e.g. rappers who put their hand around the grill of the mic because they think it looks cool). This invariably makes the mic sound horrible and very susceptible to feedback.  More importantly, it changes the directional nature of the microphone, changing it to essentially an omnidirectional microphone. One trick is to cut everything from 800 Hz to 2 kHz, compress it, and hopefully the horrible howling sound will go away and the vocals will still be intelligible. But don’t forget, the best thing to do to control feedback is turn everything down.”

- Frank Gilbert, FOH Engineer
Park West, The Vic Theater, and The Mayne Stage – all in Chicago

How to avoid it
Here are some suggestions on how to interrupt the feedback loop:
• Move the microphone closer to the desired sound source.
• Use a directional microphone to increase the amount of gain before feedback.
• Reduce the number of open microphones – turn off microphones that are not in use.
• Don’t boost tone controls indiscriminately.
• Try to keep microphones and loudspeakers as far away from each other as possible.
• Lower the speaker output. Move the loudspeaker farther away from the microphone. Each time this distance is doubled, the sound system output can be increased by 6dB.
• Move the loudspeaker closer to the listener. Each time this distance is halved, the sound system output will increase by 6dB.
• Use in-ear monitoring systems in place of floor monitors.
• Acoustically treat the room (if possible) to eliminate hard, reflective surfaces like glass, marble and wood.

Pro Tip #2: “In a well-designed system, the irritating high-pitched brand of feedback isn’t much of a problem unless someone points a mic into a monitor. So long as the performers are careful to always keep their mics pointed away from the monitors, or specifically to point that tail end of the mic at the monitor at all times, that shouldn’t be an issue.

- Bill Gibson, author of over 30 books, producer, performer and Berklee School of Music faculty member

When these solutions have been exhausted, the next step is to look toward equalizers and automatic feedback reducers.

A common technique used by sound engineers is “ringing out” a sound system by using a graphic equalizer to reduce the level of the frequencies that feedback:

1. Slowly bring up the system level until you begin to hear feedback. Now go to the equalizer and pull down the offending frequency roughly 3dB.
2. If the feedback is a “hoot” or “howl”, try cutting in the 250 to 500 Hz range. A “singing” tone may be around 1 kHz. “Whistles” and “screeches” tend to be above 2 kHz. Very rarely does feedback occur below 80 Hz or above 8 kHz. It takes practice to develop an ear for equalizing a sound system, so be patient.
3. After locating the first feedback frequency, begin turning up the system again until the next frequency begins ringing.
4. Repeat the above steps until the desired level is reached, but do not over-equalize. Keep in mind the equalizers can only provide a maximum level increase of 3 to 9 dB.

Pro Tip #3: “The last time I experienced feedback was in a small venue where I was onstage. As a musician and an audio tech, I’m a sound guy’s worst nightmare.  During rehearsal, my headset mic was feeding back and the audio tech kept turning my volume down and telling me that I couldn’t move around. I knew the problem was midrange feedback, so I explained to him that if he just lowered the midrange on the EQ, the problem would go away. He ‘passionately and firmly’ explained to me that the only way to get rid of feedback was for him to lower the volume and for me to stand still.

After enduring the first song, I walked back to the board, reached over his shoulder and dropped the midrange. I sang a couple notes, looked at him, smiled and walked back onstage. (Did I mention I was wireless, too?) The problem was solved and we didn’t talk after the set, but I know he learned something that night.”

- John Chevalier, pro audio/video expert, writer and speaker at InfoComm, NAB and other industry events

Pro Tip #4: “If there’s one thing I’ve learned in all my years of playing, it’s that the sound engineer has to be extraordinarily vigilant   even about protecting the performers’ hearing.

My last bad feedback incident was caused by gain stage being manipulated by the engineer without telling us – after we’d gotten to a good place.   The resulting, shrieking feedback changed everything – there was nothing but pain filling up space between our ears. Many people forget that EQ’ing something can cause a volume change – right in that frequency.

Of course, EQ can remedy volume problems quite easily. Just take a moment to ferret out the offending frequency or cluster of frequencies – band members protecting their ears, of course – and “forensically” attenuate, which will immediately solve the problem.  A hall of mirrors, isn’t it?”

- June Millington
FANNY frontwoman, musician and songwriter, co-founder of IMA

Automatic feedback reducers are very helpful in wireless microphone applications. Remember that microphone placement is crucial to eliminating feedback, and the temptation to wander away from the ideal microphone position when using a wireless is great. If the performer gets too close to a loudspeaker, feedback will result; a good feedback reducer will be able to catch and eliminate the feedback faster than a sound engineer.

And finally …

Pro Tip #5: “The best ‘gear’ a sound person has is his or her ears. Learn to identify the ringing frequency by doing blind ‘what is that frequency?’ tests using a sine wave generator or test tone generator. Have someone dial up a tone and see if you can identify what frequency it is. This is great training to identify the problem frequency during feedback howl and how I learned how to tame feedback.”

Dan Murphy, Sound Tech Director, Lakeside Church

NOTE: Don’t rely on an equalizer/feedback reducer alone to provide sufficient additional output in a sound system where the microphones and loudspeakers are too close together. You probably won’t get the results you need.  For more information about EQ, see this Shure Notes blog post.

KSM9HS (H for hypercardioid and S for subcardioid)

Back in 2005, Shure introduced the KSM9 vocal condenser microphone.  What made this microphone unique was its flip-of-a-switch ability to change from cardioid to supercardioid. Originally, a component in Shure’s premium UR24S/KSM9 wireless system, the mic found an enthusiastic audience that warranted its standalone status.

Live sound guys loved it.  Take John Mills, for instance: “This is a solid microphone with sound rivaling some of the best studio mics. If you’re serious about vocals, this is certainly a mic to try out. On second thought, just go get one and save some time and frustration with your vocal sounds.”

And Pro Audio Review said:  “This mic sounds like what you have always wanted a live mic to sound like. That is to say you have the brilliant clarity of a large diaphragm studio microphone in a durable live application. It is crisp without sounding too harsh, but warm and natural for a true reproduction of the human voice.”

The pros were on board and so were Erykah Badu, James Taylor, Buddy Guy and a bunch of other important people.  But back at Shure HQs, the engineers decided to take the KSM9 not only higher and higher, but lower, too. So back there, in Shure’s top secret Technology Annex, the question was, “What if the KSM9 came with switchable hypercardioid and subcardioid patterns?”

“What if …?” is here.  On October 26, 2012 Shure announced the KSM9HS vocal condenser microphone.  To get the Shure perspective (since all the reviews aren’t yet in), we went straight to Shure’s Soren Pedersen. He’s a Product Specialist and in true Shure Associate form, he’s also connected to the music business as a sometimes-recording engineer (rumor has it that one in five Shure Associates are weekend warriors of one stripe or another).  We asked him to tell us what’s so great about this mic.

This is a sibling of the KSM9 vocal condenser microphone.   How is the KSM9HS different?

The two share many design elements like the dual diaphragm cartridge, premium circuitry, and top notch shock mounting. The main difference is their polar pattern selections. Both mics offer two selectable polar patterns; the KSM9 switches cardioid and supercardioid, while the KSM9HS switches to subcardioid (also known as wide cardioid) and hypercardioid patterns.

That seems like a pretty nuanced difference.  What led to its development?

At Shure we’re on a never-ending quest for customer input. From talking to KSM9 users, we found out that there were a couple of features they wanted:

1) a mic with even less proximity effect, and

2) a mic with even more rejection for improved gain before feedback.

Working with our mic lab we found that we could create a version of the KSM9 with two new patterns providing a solution to both customer issues. The subcardioid pattern has greatly reduced proximity effect due to it being very close to an omnidirectional pattern, and the hypercardioid pattern has unbelievable side and off-axis rejection, even more than supercardioid. That helps to isolate the vocal and reduce the risk of feedback.

Has Shure ever produced a hypercardioid or a subcardioid mic?

This is Shure’s first handheld mic that features either of these patterns. We do have one hypercardioid headworn mic in our catalog (WCM16), which is a head worn mic also used for vocals.

Seems like most mics have a niche in either live sound or recording.  Where does this one fit?

Both KSM9 and KSM9HS were designed for stage use and live performances. But they’re also great studio vocal mics for artists who like to hold the mic while recording.

The original KSM9 has switchable polar patterns – from cardioid to supercardioid. But the KSM9HS is on both sides of that – from sub-cardioid to hypercardioid. Can you talk about the specific applications for the HS version?

With the grille removed, a flip of the switch changes the polar pattern.

The applications for the two patterns are quite different from each other, which makes the HS a more versatile vocal mic than most.

The hypercardioid side has amazing isolation and a big warm sound. You’ll really benefit from the off-axis rejection in especially loud environments where you may have stage monitors, guitar amps, or drums all pointing towards the mic which leads to a washy, unfocused vocal sound. On many hypercardioid mics, there’s a concern about too much proximity effect (the buildup of low frequencies as you get closer to the mic).

What’s unique about the KSM9 cartridge design is that it uses a dual diaphragm element. Having two diaphragms helps control proximity effect and results in a very smooth low-end response. Better than many other hypercardioid mics out there.

When switched to subcardioid (sometimes referred to as ‘wide cardioid’), it has more detail and will pick up more ambience, so it’s useful on stages with lower volume levels or if in-ear monitors are being used. Because subcardioids aren’t very vulnerable to the proximity effect, adding in the dual diaphragm to the equation means an exceptionally natural sounding low end. The wider pickup pattern is very “open” sounding and makes a great mic for interviews, Q&A, studio work or capturing multiple singers.

Around the halls of Shure, this is considered a ‘problem-solving’ mic. What are some of the problems it can solve?

The biggest problem it solves is related to feedback and bleed. Condensers are sensitive microphones that provide more detail than dynamic mics, but as the volume on stage goes up, the risk of feedback increases.

The KSM9HS set to hypercardioid has a tremendous amount of rejection so it can really help keep unwanted sounds out of the mic. For bands that use in-ear monitors, feedback is much less of an issue but the KSM9HS can still deliver a benefit.  The isolation it provides will reduce the pickup of unwanted sounds like drums and loud guitar amps and that results in a cleaner and overall better-sounding in-ear mix.

Ambience can be controlled using ambience mics instead of the band’s vocal mics so the engineer has better control on the in-ear mix.

There were undoubtedly some Beta tests out in the field.  Who was using them and what did they think?

That’s Kenny Chesney and the KSM9HS

Our Beta testers were absolutely thrilled with the mic. Some of the artists we tested with included Kenny Chesney, Cage the Elephant, Drive by Truckers, and Mastodon. Most of them having extremely loud stage volumes.  They were all fans of the KSM9, but they wanted more isolation. After evaluating the KSM9HS in hypercardioid, they noticed the improvement right away.

Kenny Chesney used it on his massive summer stadium tour and had some unique challenges that the KSM9 solved for his crew. He sings a song out in the crowd in front of a giant PA during the show and there weren’t any feedback issues at all because the KSM9HS rejects amazingly when set to hypercardioid. Kenny also likes to sing with his hand cupped around the mic, which can typically lead to big challenges in keeping a stable pattern and frequency response, but the KSM9HS is so directional that the response is very consistent regardless of the artist’s mic technique. The sound crew also used a KSM9HS set to subcardioid for guest vocal appearances.

Editor’s Note:  John Mills (yeah, the same John Mills who loved the KSM9 six years ago) was the Audio Systems Crew Chief for the 23-city Kenny Chesney/Tim McGraw “Brothers of the Sun” tour that Soren referred to. Mills said this: “One of the biggest problems with Kenny is finding a microphone that sounds consistent in any situation. You can cup it, then take your hand away, and the sound barely changes. It still sounds like a KSM9. And there’s also way less crowd noise coming in through the PA than there has ever been.”

It’s a condenser mic and it’s also two mics in one. How does it compare to the KSM9 in price?

The KSM9 and the KSM9HS are priced exactly the same ($699) and they are both available as wired mics or replacement wireless heads for use with Shure wireless transmitters.

Who should buy this mic?

Bands, artists, vocalists, and engineers looking to elevate their vocal mic both in sound quality, and pattern flexibility. If you have a loud stage, set it to hypercardioid, and if you like to “work the mic” at a coffee shop, set it to subcardioid.

For more information about the amazing KSM9HS, check out this video.

About SOREN PEDERSEN:

A member of the Product Management team at Shure, he attended Columbia College in Chicago, studying audio arts and acoustics.  Like many of his co-workers, he is a musician (drums and guitar) and still finds time to record local bands, using “all Shure mics, of course”. 

 

 Todd Ohme, The Detroit Pistons Drumline Director

When most of us think of marching bands, making the drums louder isn’t something that comes to mind right away.  But for Todd Ohme, the Director of the Pistons Drumline, it’s been more like a calling.  Specifically, he wants to rock the house – and that house is The Palace of Auburn Hills, where The Detroit Pistons play to a crowd of over 20,000 fans.

Inside The Palace on Game Day

Here’s how it all started: Todd had been a percussionist throughout his school years and continued performing at Michigan State, where he was a member of their 35-member drumline.  Then he graduated with an engineering degree and reality set in.  Since he wasn’t following a career as a professional musician and drums aren’t traditionally a solo instrument (ask anyone who lives two doors away from a teenager learning to play them), there wasn’t really an avenue for Todd to continue pursuing his passion for percussion.

Then Fate stepped in.  Turns out that just about that time (2005), The Detroit Pistons were looking to add a drumline.  With nothing to lose and everything to gain, Todd assembled a group of MSU drumline members.  The Detroit Pistons drumline, directed by Todd Ohme, debuted in the 2005-2006 basketball season.

A drumline, which may seem obvious to some but not all, is pure marching percussion.  The appeal, according to Todd, is that “drums are universal – there’s no language barrier. It’s entertainment that people of all ages can enjoy so it’s just perfect for an arena like The Palace.” Today’s Pistons drumline, no longer limited to MSU musicians, has 24 percussionists, At each game, they use up to 8 snare drums, 5 marching tenor drums, 3 cymbals, and 4 marching bass drums (which are played horizontally on rolling stands, but more about that in a minute).

Finding the Right Balance

One can only imagine Todd’s enthusiasm in building a drumline from scratch for a major NBA team.  He was on fire. “I really wanted to take it the next level – big energy and big movement.  We want to run out there, amp up the fans in a minute and thirty seconds, and then leave the crowd feeling that intensity.” The Palace has an awesome sound and light rig to support The Pistons’ demand for big, big halftime show productions (and don’t forget, the venue is also a concert arena), so Todd and his drumline were all set to impress. They’ve come a long ways over the past 8 years, Todd says “In the beginning, we may have been a little too intense and scared a few children instead of entertaining them.”

Connecting with fans is the mission. A great opportunity for this is the drumline’s post-game performance, where fans can groove to the beat up close, as they’re leaving The Palace.  Todd says, “We’ll have crowds of over  400 people surround us – and they feel really connected to the experience being that close and seeing how big all of our movements really are.”

Post-Game

 Rockin’ the House

The drumline occasionally uses audio backing tracks to keep their performances more diverse.  Everything from Motley Crue to Rush to Stevie Wonder, but Todd still wanted the drumline to have more of that bone-shaking rock and roll physical effect.

“In thinking about how to connect with the crowd, we looked at miking techniques. We’ve miked the drums right from the start using a few stand mics, mainly to help get the clarity of the sound spread more evenly throughout the arena. But I wanted to bring out the lower frequencies of bass drums and really dial in the sound. When you’re listening to an audio engineer soundcheck a kick drum for instance – the engineer gets the levels right and turns the faders up. When you first hear it through the house system and your pant legs shake with every hit….that’s really impressive.” He knew the drumline could have that same intensity level.  “I wanted the fans to be blown away by the sound.”

The Beta 52A/Kelly Shu™ Mount Solution

Todd set out to find a creative way to wirelessly mic the drumline’s four bass drums.  And did.  “We got the Kelly Shu Bass Drum Microphone Shock Mount for our Beta 52As and it was an absolute no-brainer. The mic is suspended inside each drum so there’s no fear of damaging it or having the mic pick up a hard bounce on the shell.   We’re getting a great sound from inside the drum and it’s given us a lot more gain before feedback, too.  Before this setup, a spectrum analysis of the drum, was showing a roll-off around 70Hz and some resonant peaks around 150 Hz.  Now, without any adjustment we’re getting a much deeper/engulfing sound, and when we bump the EQ in the 40 to 50 Hz range you can hear the roof of the building shake! We have a Beta 52A in each of the four bass drums and we have them all dialed up so the fans can actually feel what they are seeing and hearing. It’s ridiculous.”

We had a lot of trouble in the past, but using a Beta 98 ad/c on the snare head and Beta 52As inside the bass drums has given us an edge.  We also EQ and compress the snare signal to make it fatter and as a whole, sound a lot more like a fat rock drumkit than a drumline.”  Todd’s a happy man.

What’s the Frequency, Todd?

A look inside

The list of wireless equipment at The Palace is pretty lengthy. According to Todd, finding open frequencies was simplified, since the Shure ULXP4 receivers the drumline uses all have a frequency group/channel scan feature.  With the dozens of wireless systems in use on game day, that’s impressive. “We haven’t had a drop-out or problem with frequency interference and the receivers are located up in the control room, about 300′ from the drums.”

One of the few drumline majordomos who continues to perform with the group, Todd Ohme doesn’t really sound ready to lay down his drumsticks or stop thinking of ways to turn the group’s minute-and-a-half sets into a Springsteen moment.  He’s looking to connect with the fans and deliver that palpable excitement.  “If the crowd can see it, hear it and feel it, then you’ve got it all and that’s what it’s all about.”

So you’re not in a drumline?  The mounting system is perfect for gigging musicians, too!

Check out Joe Crabtree’s video review of the Kelly Shu for Rock drummers (and he’s using a Beta 52A) 

In its simplest definition: Equalization involves selectively boosting or cutting bands of frequencies to improve the performance of a sound reinforcement system.

Like many other audio technologies, EQ was born in telecommunications, where filters were used to restore and flatten all frequencies, so that response to all frequencies would be equal – hence the label “equalization”. Hollywood came on board in the early 1930s with the emergency of ‘talkies’, but it wasn’t until the late 1950s and early 1960s that Texas academics in the field of Physics did research that led to precursors of today’s pro audio technology. By the 1980s, graphic equalizers on our home stereo systems were the portals to EQ and many of us had a good time adjusting the sliders.

In the natural world, the sounds we hear are incredibly complex.  As we walk down a city street, we hear the rumble of a subway train, a police siren, the boom-boom-boom of an enhanced bass car stereo passing by or the voice of the person walking next to us.  But in a live sound or recording situation, none of us wants the listener to either cover his ears or struggle to pick out a specific sound source.

What equalization can do when used properly

• Noticeably, but not dramatically, improve the naturalness or intelligibility of a sound reinforcement system by emphasizing the frequency ranges most critical for speech.
• Noticeably, but not dramatically, increase the overall output level of a sound reinforcement system by reducing the system’s output in the frequency bands at which feedback occurs. These frequency bands will differ from system to system based on many variables, including room acoustics, microphone placement/design, loudspeaker location/design, even air temperature.

 Expert Tip #1

“When running live sound, I occasionally run into a feedback situation. When this happens, I quickly determine the frequency that’s feeding back, and by that I mean, ‘Is it high-end feedback? Mid-range feedback? Or low-end feedback?’ Once I’ve identified that, I turn down the appropriate knob on the EQ. It’s fast. It’s easy. And it works every time.”

- John Chevalier

What equalization can’t do

• Make a poorly designed sound reinforcement system work satisfactorily. Every sound reinforcement system is subject to the laws of physics.
• Improve intelligibility problems caused by reverberation, reflections, mechanical vibration, high background noise levels, or other problems caused by the location or physical design of the room. These problems are acoustical in nature and can’t be solved electronically. They must be resolved with acoustical solutions, such as sound absorbent panels and heavy drapes.
• Improve intelligibility problems caused by the talker being too far from the microphone.
• Improve the performance of sub-standard audio components in the sound reinforcement system.
• Eliminate distortion or noise problems caused by mismatched audio levels between system components.
• Improve echo return problems in teleconferencing systems.

How to approach equalization

FOH engineer and touring pro John Mills likes to think of each song as a line, with each instrument making up part of it.  If there are too many instruments or frequencies taking up the same space, the line gets bumpy and the mix gets muddy.

Expert Tip #2

Here are some general guidelines to consider when you’re trying to find your space.

20Hz to 80Hz: This is your sense of power in an instrument or mix.  It’s the stuff you feel more then hear.  The kick drum and bass guitar are down here in this range.

80Hz to 250Hz: The area where everything comes together.  This is where a lot of things can go wrong and too much in here will make a mix sound sloppy.

250Hz to 2kHz:  Most of your fundamental harmonics are in this range.  These are some of the most critical frequencies to building a solid mix.  Learn what instruments are most dominant in these frequencies and clean up around them.

2kHz to 5kHz:  Here you will find the clarity to almost everything.  But be careful, too much of a good thing can start to sound harsh.  This is an area where subtly is the key.

5kHz to 8kHz: Mostly sibilance and “s” sounds.  Much of the vocal consonants are defined in this range.

8kHz to 20kHz: Brilliance is the word here, the top end of cymbals.

“Becoming a master of EQ is like becoming a master painter.  Sometimes you just have to throw some paint on a canvas and see how it works.”

- John Mills

Approach equalization gently and slowly! After every adjustment, listen carefully to the resulting sound. The goal is to improve sound quality as well as increase the gain before feedback. When the system is loud enough and/or clear enough, stop equalizing! Also, stop equalizing and examine the complete sound reinforcement system in detail whenever the equalization causes degradation in the sound quality.

JOHN CHEVALIER is a pro audio/video expert, writer and speaker at InfoComm, NAB and other industry events.

JOHN MILLS is a touring FOH engineer and the expert behind the TechTraining101 site offering sound advice for engineers at all experience levels.

Contributors: Frank Gilbert, FOH Engineer at Chicago’s Mayne Stage, The Vic and Park West; Kent Morris, Peavey Electronics, Cornerstone Media; Stuart Rosenberg, Partner – SPACE and League of Creative Musicians

Live performance venues deal with it all the time.   In this post, we’ll break down the subject of acoustics for you into its most basic components, talk with a trio of our experts about how they handle venue challenges and offer some helpful hints that won’t require you to hire an acoustician.

A little history

The study of acoustics goes all the way back to the 6^th century, but the study of modern architectural acoustics is as recent as the 1900 when Wallace Clement Sabine designed Boston’s Symphony Hall according to scientific acoustic principles (based largely on reverberation time). Before then, the creation of great sounding concert halls was more a matter of luck than good design. Now concert hall acoustics is a recognized area of acoustics that straddles architecture, engineering and physics.  And Sabine must have done a pretty good job, since Symphony Hall is cited as the #2 concert hall in the world for acoustical quality. (Vienna’s Grosser Musikverinsaal is #1 and Carnegie Hall is #8.  London’s Albert Hall, which most of us know from “A Day in the Life” is ranked a distant #58.)

Acoustics By Definition

Acoustics is total effect of sound, especially as produced in an enclosed space. It’s the scientific study of the generation, transmission and reception of sound.  But here’s a simpler definition: It’s the total effect of sound in a room.

Audio expert Pat Brown explains how it relates to live sound reinforcement: “The sound heard in an auditorium by a listener is a complex combination of the sound produced by the gear and the way that it interacts with the room. It’s a fact that most of the sound heard by any listener gets there only after many, many interactions with the room’s surfaces. Each reflection modifies the sound a bit, and after several interactions, it looks nothing like what left the loudspeaker in the first place. The room places its own signature on all sounds radiated into it, which can either enhance or corrupt the sound. Good gear doesn’t sound good when used in a bad room”.

The fact is, there are no “good” or “bad” acoustics, only “appropriate” or “inappropriate” acoustics for the intended application.  Appropriate acoustics are those transmission and reception qualities in a room that enhance the musical performance and/or improve the ability of the listener to understand speech.  Music benefits from longer reverb time and constructive delay, so a room with those characteristics is considered “wet”.   On the other hand, speech reinforcement benefits from a “dry” room without the reflections that cause reverb.

So many advances have been made in the area of sound, that the bar for acoustical standards has been raised, along with the expectations of the audience.  No matter how great the performance is, it’s going to be a bad experience for the audience that can’t understand it.” 

Structural Concepts and Solutions: Isolation, Absorption and Diffusion

Isolation prevents sound from escaping its intended destination. By using dense door materials, tight seals, and double pane windows, transmission can be kept in check. Additionally, minimizing wall and ceiling penetrations will improve isolation.

Absorption uses a combination of dead air and mass to prevent the reflection of acoustic energy back into the occupied space. From mineral fiber to acoustic foam, different types of absorbers stop detrimental reflections that mask intelligibility.

Diffusion creates multiple small reflections from one or two large ones. The result is more pleasing to the ear, with an increased sense of “space” and “depth” as compared to a harsh two-dimensional slapback reflection. In a large space – where the slap and flutter from large parallel surfaces creates unpleasant sound –  diffusion, in the form of acoustical clouds and canopies, can rearrange the reflected energy into a usable soundscape.  

Example of acoustical clouds – Armstrong SOUNDSCAPES

With a careful study of the desired room signature, changes can be made to improve the overall sound in any room, whether the need is for isolation, absorption or diffusion.

We asked Kent Morris, President at Cornerstone Media and Worship Market Manager at Peavey Electronics, for some tips and tricks that you can use to improve room acoustics:

1. Divide and conquer. There is no rule that says the drums have to be placed at center stage. Move the drums to one side and place the singers and musicians on the other side. The physical separation will improve the clarity of each section and might increase gain before feedback in the vocal mics.

2. Baffle the sound. Build inexpensive sound baffles out of 2×4 wood frames and fill them with mineral fiber. Cover the sides with acoustic cloth and place the baffles (also called gobos) between the instrument amps and vocal mics to prevent sonic bleedthrough.

3. Aim high. Point the instrument amps at the people using them to decrease stage volume and prevent high levels from reaching the front rows.

4. Isolate the drummer. Use a panel behind the drummer to reduce rear reflections while improving the direct energy reaching the drum mics

5. Ditch the dish. In fan-shaped rooms, avoid aiming speakers directly at the rear wall. With a curved back wall, energy is reflected onto the stage, causing detrimental delays.

It can also happen that the solution is surprisingly simple and maybe even accidental.   A favorite venue of this Shure Notes editor is Space – a completely rehabbed and intimate music venue that has boasted the likes of Leon Russell, Taj Mahal, Dr. John and many others.  When we asked Stuart Rosenberg, the impresario behind the club, for an example of an acoustical problem and solution, here’s what he told us:

“When we built out our control room, we discovered that our design yielded significant resonance in the area of 30-100 Hz – a classic problem encountered in rectangular spaces, and potentially a vexing one that can prevent accurate low-end monitoring.  Much to our amazement and delight, we discovered that whenever we opened the door to the studio storage closet, the bass loading immediately disappeared – the volume of that space acted as an extremely efficient bass trap – and the frequency response of the room became linear. Problem solved!”

What a Sound Engineer Can Do

Let’s assume there’s nothing more you can do with the physical space.  Now what?  For another perspective, we turned to Frank Gilbert who is FOH and Monitor engineer at three popular music venues (once movie theaters and vaudeville houses) Park West, Mayne Stage and The Vic. We asked him to choose one and he talked about some of the challenges at The Mayne Stage.

Mayne Stage, formerly The Morse Theater circa 1912

“It’s a very deep room and I’ve noticed that unlike other places, when the room fills up with people, it doesn’t deaden up some of the bouncy low-mids.  We do drape the stage on all three sides with heavy-duty curtains and that helps a bit. At Mayne Stage, the FOH PA is a mono cluster with a pair of flown subs.  We do a lot of jazz trios., and with the upright bass, the flown subs start to rattle because they’re not coupling with any other speakers.

It’s a great sounding room with a great sounding system but with some of that deep stuff below 80 Hz, it shakes the HVAC in the room, so I just end up putting a high-pass filter on the upright bass and sometimes I even mute the subs and let the amplifier in the backline fill the room.   It’s such a live room that sometimes that’s enough, especially with those quieter shows.

With a rock show, you’ve already got a couple of guitar amps and a drum kit banging away, then your acoustic focus becomes getting the vocals to be heard. My favorite move in one of those situations is to use a Beta 58 instead of an SM58 – that always helps.   Sometimes in that room, a Beta is the right vocal mic.

We haven’t done anything in terms of acoustical treatments.  It would be really nice if we put curtains on the walls behind all the audience areas to deaden it up a little, but it’s actually a great-sounding room.  It’s well suited to piano jazz and cabaret but lately we’re doing more and more rock shows.   That room is about controlling bass and trying to control low-mids – bass management is really the name of the game there and when we have a full-on rock band, getting the vocals to sit well above the band is al little bit of a trick because there are no front fills at the lip of the stage – just the big cluster of speakers flown at the top of the room.  If it’s going to be a really loud band, sometimes I’ll take a couple of little speakers and put them at the lip of the stage, close to the audience.”

Frank Gilbert played in a band at age 15 and almost 25 years later, handles FOH sound at Chicago’s The Vic, Park West and The Mayne Stage.  He has a home studio and also does location recording for Metro Mobile Recording.

Kent Morris is the President of Cornerstone Media, worship market manager for Peavey Electronics, a speaker and a writer on pro audio topics.  He is a frequent contributor to Shure Notes.

Stuart Rosenberg is a musician, composer and producer. He is also the force behind SPACE, the Society for the Preservation of Art & Culture that brings live music and legendary musicians almost every night of the week to an intimate performance venue in Shure’s former hometown of Evanston, Illinois.

Readers of Shure Notes have enthusiastically embraced our annual review of audio myths, which Tim Vear shatters (or not). Here he delivers the answers to some of the most recent questions that originate from Shure customers who call Tim and his fellow Apps Experts for help.

Myth: Loud sounds can blow out a microphone

False.

Generally it is not at all likely that a loud sound will damage a microphone.  A dynamic microphone can handle levels above 150dB SPL. That’s not a volume that can be achieved short of shoving a microphone into the exhaust of a jet engine.  It is almost impossible to physically damage a dynamic microphone at any achievable sound level.

Classical ribbon microphones that use an aluminum ribbon can be damaged not by sound pressure but by a puff of air.  A blast of air could physically push the ribbon out of its magnetic gap far enough to stretch it. (It’s worth noting that Shure’s ribbon mics use a material – Roswellite^® - that can withstand air pressure making them about as durable as any other Shure microphone.)

Condenser microphones can experience distortion if the sound pressure level exceeds the rated capability of the mic.  But the distortion is coming from the electronics of the microphone and not the capsule itself. The electronics are essentially a fixed gain amplifier. Any amplifier has a maximum output level that it can deliver before it distorts.  So if you increase the input level to the amplifier beyond what the max should be, it will drive the amplifier to distortion.  It’s the pre-amp in the condenser microphone that’s distorting.

Many condenser mics feature a switchable attenuator that allows the mic to handle higher sound levels.

Myth: A wireless microphone can’t cause feedback.

False.

We get this question every once in a while from someone who may be using a wireless mic either for the first time or in a new way.  They’re surprised to experience feedback when using a wireless mic in front of loudspeakers.

They believe that if they eliminate the wire, they’re eliminating the feedback loop.  The fact is, the cable is just being replaced by a radio wave, so acoustically; the system behaves exactly like a wired system.

It may be triggered by someone who is accustomed to using a wired mic that’s tethered to a mic stand or pulpit.   A wireless mic gives the user the ability to walk around in the audience or among the congregants and potentially in front of the house speakers. In that case, the system may very well cause feedback.

If the user is too close to the speakers or the gain is turned up too high, feedback will occur in both wired and wireless systems.

Myth: Digital wireless microphone systems are interference free.

False.

Some people believe that the error correction capabilities of digital systems can compensate for extreme radio interference.  But the fact is that any radio microphone or wireless device can experience interference because there is no frequency band that is reserved specifically for an individual’s wireless microphone.   At minimum, there are other wireless microphones operating in the same frequency range and if they happen to collide, interference will occur.

Each frequency band has its own unique interference sources – it could be television stations in the UHF band or baby monitors in the 900 MHz band, WiFi hotspots in the 2.4 GHz band – but all frequency ranges are shared by different users that can interfere with one another.

Whether the transmission is digital, analog or some unknown modulation that hasn’t been discovered yet, if it depends on a radio signal, it is subject to interference.

Myth: It’s OK to use just one earphone for in-ear monitoring.

False.

There’s a functional reason and there’s a health reason.

Functional: With only one earphone in place, you have only the possibility of monophonic sound.  Whatever the mix is that you’re listening to; you’ll only hear it in mono, which removes all of the spatial cues that you get from a stereo mix.  A stereo mix makes it not only more pleasant to listen, but also enables you to pick out the parts that you need more easily by virtue of their pan location, not just their level.

Health:  If both ears are exposed to an open sound field, the perceived level of sound at each ear will be equal.  However, if one ear is blocked from the outside sound but presented with that same sound via an earphone while the other ear is still exposed to the open sound, the relative level perception changes.  It is found that in order for the sound levels to appear equal, the level of the earphone sound must be about 6dB louder than the level of sound at the open ear.   This is called the “occluded ear effect.”

Here’s an example: Let’s say the left ear is being exposed directly to ambient sound and the listener wants to match the sound level that’s coming through the earphone in the right ear (the occluded ear) so that the volume appears roughly equal.  By measuring sound levels at the eardrum, studies have shown that the listener will increase the earphone level at least 6dB above the open ear level to perceive an equal level. So in a loud environment where the open ear may be subject to 120dB SPL, the earphone may be cranked up to 126dB SPL just to compete. Over time, this increases the risk of noise-induced hearing loss.

It’s not a huge problem in a quiet environment – a jazz band or maybe a theatrical application – but for a heavy metal, rock and roll or even a loud country band – it’s a serious consideration.

Myth: Phantom voltage is the same thing as bias voltage.

False.

Phantom power is a dc voltage (11 – 48 volts) that powers the active circuitry of a condenser microphone. Phantom power is normally supplied by the microphone mixer, but may also be supplied by a separate phantom power supply. Phantom requires a balanced circuit in which XLR pins 2 and 3 carry the same dc voltage relative to pin 1. So if a mixer supplies 48 volts of phantom, XLR pins 2 and 3 of the microphone cable each carry 48 volts dc relative to pin 1. Of course, the mic cable carries the audio signal as well as the phantom voltage.

Bias is a dc voltage (1.5 – 9 volts typically) that is provided on a single conductor.  Unlike phantom power, bias does not require a balanced circuit. Bias supplies power to a single Junction Field Effect Transistor (JFET) connected to the output of an electret condenser mic element. The JFET acts as an impedance converter, which is a necessity in any microphone design that uses a condenser element. A condenser element has a high output impedance (>1,000,000 ohms).  The JFET lowers the output impedance of the condenser element to about 1000 ohms to feed the signal to the rest of the microphone active circuitry.  Typically, the JFET impedance converter is placed very close to the condenser element to prevent the loss of high frequencies that would occur from even a short length of high impedance circuitry.

How are phantom power and bias voltage related?  In a full-size condenser microphone, phantom power operates the active electronics in the body of the microphone – and those electronics have the function of converting the unbalanced signal from the condenser element to a balanced output signal. The electronics may also provide some equalization and possibly some amplification.  Finally, those same electronics provide the bias voltage that powers the JFET impedance converter connected to the actual condenser element.

Since the JFET impedance converter allows some length of connecting cable without high frequency loss, it is possible to separate the main condenser electronics from the actual microphone element.  This is the form used for condenser microphone designs in which the microphone element is connected to the main electronics assembly by a miniature cable up to perhaps 30 feet in length.  This cable conducts bias voltage from the main electronics to the JFET attached to the condenser mic element.

In some condenser microphones, the bias voltage must be supplied on the same conductor as the audio. Condenser elements with a built in JFET use this configuration and employ a single conductor, shielded cable. Other designs utilize separate conductors for bias and for audio. Consult the manufacturer’s data sheet to find out the exact wiring configuration for a specific microphone.

It’s important to understand the phantom vs. bias distinction when using wired microphones with a wireless transmitter.  A typical wireless bodypack transmitter provides bias voltage on its input connector. Any condenser microphone element with an integrated JFET can be plugged into such a transmitter.  This includes most condenser lapel mics, headset mics, as well as boundary mics and miniature gooseneck mics that have removable electronics assemblies:  these microphones only require bias voltage provided by the wireless transmitter.

But you cannot plug in a Beta 87 or a KSM32 or some other full size microphone that needs phantom power into a device that only provides bias voltage.  Those microphones can only be used with mixers or other devices that provide phantom power.

Here’s what you need to remember: A condenser microphone that requires phantom power will not work with an input that only supplies bias voltage.  Similarly, a condenser microphone that requires bias voltage will not work with a standard phantom power mixer input.

Know of any other audio myths that you want busted?  Leave a comment/suggestion and we may include it in a future blog post!

In our final Part 4 video, we discuss the myth that the SM58 is not supposed to have ‘proximity effect.’

Get up close and personal with an SM58, and the sound gets all warm and deep.  It’s called ‘proximity effect’.  In this video, Shure’s Chad Wiggins explains that this is perfectly normal – and how a savvy singer can use it creatively.

SM58 Facts vs Fiction Overview

• Part 1 – Myth: The SM58 is “old technology”
• Part 2 – Myth: The SM58 grille should never dent
• Part 3 – Myth: The SM58 causes feedback
• Part 4 – Myth: The SM58 is not supposed to have ‘proximity effect’

In part 3, we discuss the myth that the SM58 causes feedback.

When feedback strikes, the microphone usually takes the rap.  But feedback isn’t all the mic’s fault.  In this video, Chad Wiggins comes clean, and reveals that feedback is an acoustic conspiracy.

SM58 Facts vs Fiction Overview

• Part 1 – Myth: The SM58 is “old technology”
• Part 2 – Myth: The SM58 grille should never dent
• Part 3 – Myth: The SM58 causes feedback
• Part 4 – Myth: The SM58 is not supposed to have ‘proximity effect’

In Part 2, we discuss the myth that the SM58’s grille should never dent.

Like scars on a heavyweight prize fighter, a bashed and bruised grille reveals the long, hard road that an SM58 microphone has traveled.  Like a bodyguard, the grille’s job is to take a beating to protect the “VIP” – the more-delicate microphone transducer inside.  In this video, Chad Wiggins explains why a 58 keeps working long after other microphones have gone down for the count.

SM58 Facts vs Fiction Overview

• Part 1 – Myth: The SM58 is “old technology”
• Part 2 – Myth: The SM58 grille should never dent
• Part 3 – Myth: The SM58 causes feedback
• Part 4 – Myth: The SM58 is not supposed to have ‘proximity effect’

In Part I, we discuss the myth that the SM58 is “old technology”.

It’s true that the SM58 has been around longer than many bands.  But believe it or not, the technology that makes it work is still cutting-edge today!  In this video, Shure’s Chad Wiggins explains why other dynamic mic manufacturers work so hard at imitating the fabled 58 – and the one thing they all get wrong.

SM58 Facts vs Fiction Overview

• Part 1 – Myth: The SM58 is “old technology”
• Part 2 – Myth: The SM58 grille should never dent
• Part 3 – Myth: The SM58 causes feedback
• Part 4 – Myth: The SM58 is not supposed to have ‘proximity effect’

But this summer, he delivered the goods to over 1 million fans during the 23-city Kenny Chesney/Tim McGraw “Brothers of the Sun” tour where he was Audio Systems Crew Chief.  The tour ended with two sold-out shows at Massachusetts’s Gillette Stadium where over 110,000 fans broke all records for the most concert tickets sold in New England history.

We’d heard that there were over 69 tractor-trailers needed to move the extravaganza from city to city, so we asked John about that.  The truth was even more jaw-dropping: 22 tour busses, 39 trucks for staging, 40 gear trucks and 10 trucks of terraplast flooring to cover the grass in sports arenas.  That’s a total of 111 vehicles.

A sound crew of 11, including John’s crew of seven and Chesney’s team of FOH and monitor engineers, kept the audio worthy of the thousands of dollars some ticket scalpers were getting for front row seats.

John Mills

A lot of trucks, a lot of dates, a lot of crewmembers and a lot of sound checks. Since most of our readers aren’t traveling with that kind of entourage, we asked John to share his thinking on sound checks for non-globetrotting musicians and their sound people.  Sure, the scale is a little different, but the steps are the same whether the gig is at a local club or Mile High Stadium.

Now back in his Nashville office as Vice President of Morris Light and Sound, he broke it down for us into perfect thirds:

Line Check is when the engineer and a friend verify that everything is plugged in correctly and each wireless mic, personal monitor, monitor wedge, main speaker, subwoofer, etc. is working.  Engineers, we should do this before every performance, and it should be done BEFORE the band arrives.

(Editors’ Note: You can find a preflight checklist in an article titled “Check/Re-Check” on www.TechTraining101.com)

Sound Check is all about the engineer.  Band, we need you to play a small sample of what you are playing or singing that day… no 50% volume here or timid “check 1, 2, check 1, 2”. We really need a decent version of what you’ll be doing.

This is where we set things like gain and EQ, the two most important considerations to a musician.  If you check too quietly, we’re likely to turn you up, but when you give it your all in the show, we will have to turn you down. Therein lies the problem with most monitor mixes.  If you give me quality input level checks at this time, it’s likely that your mix will not change in the live performance.

There are other pitfalls here, too.  For instance, a lot of times I’ll ask the drummer to just play all his drums for a minute.  Most drummers will unconsciously play softer if you just ask for a kick, then a snare, etc.  When it comes to vocalists, I will already have checked the acoustic or keyboard and will have them start the chorus of an easy song so they don’t feel so silly singing all alone.

Once we’ve made it through most of the inputs, I’ll have the band play a short chorus of a song where everyone is playing and singing.  This is often the exact same song every day. When you find one that has everyone playing and harmonizing, it’s a good idea to use it all the time because everyone gets used to it. Keep in mind we have not really adjusted monitors yet. At this stage, I get a chance to see all inputs and make any quick final adjustments to gain levels.

Then I ask who needs what in their monitors and I make those adjustments. If your band is on personal mixers, now is the time to make any adjustments.  Musicians, if you make adjustments to your mix before I’ve had a chance to settle in on gain structure, your mix will change.

The FOH engineer needs to be clear about being finished with this step, letting them know that you’re ready for them to work on their mixes.

Also, when you’re setting gain and rough EQ during this sound check, take no more than 10-15 seconds per input.  Have this discussion, and expect that they’ll give you a reasonable level, then set the gain quickly, grab a quick listen to the EQ and the MOVE ON.

The band loves it when we move fast.  You can always come back during the next section and make more tweaks.  If you spend 10 minutes getting the “perfect” kick drum sound, no one will be interested in helping with the rest of this process. 

Rehearsal. Tell the band “I’m finished adjusting the major components, so does anyone need any adjustments to their mix?”  If the answer is yes, make the adjustments and have the band do another short chorus, ask again, and if everyone’s satisfied, say “Thanks for taking the time, I’m now in tune and can confidently respond to the house mix and any needs you may have.”

What we just did by taking a few minutes (FOH folks, I mean a few minutes — less than 10) is that we have tuned our instrument and can confidently hit the first chord. There may still be a bunch of little adjustments to the mix, but for the most part, you should be in the ballpark.

Performers: during rehearsal, the sound engineer reserves the right to really mess with the house sound.  Sound people, please take the time to mute the mains and see how loud the stage sound is, turn up the harmonies a little too loud so you can get the right blend and then set them back down in the mix.  Turn the drums on and off, turn the subwoofers on and off.  Take some time to see how your instrument is responding.  But do this as quickly as possible, since we also we want them to be able to rehearse.  Finish rehearsals with the house sound on.  Many of their monitor mix adjustments will be very different if musicians don’t hear the house sound blending with their wedge or ear monitor sound.

Once you get gain structure set, don’t change it, especially after sound check and especially, especially (did you hear that?) not after rehearsal.  Unless something is about to explode, do not mix with the gain knobs; doing so will adjust everything downstream, including the bands monitors.  Let me say it again: Unless a signal is about to pop the top off the little red clip LED, leave it alone.  This is why it’s so important that we got the gain right during sound check, and that the band gave us real world rehearsal signals.  If you constantly adjust the gain during the show you’re changing performers’ mixes, especially if they’re on “ears”.  There are these little slider things closer to your hands that we should be using now to balance the mix. That may sound a little sarcastic, but some of us need to be reminded.

What else can change a musician’s mix?  The room and the audience.  Sound check is always different than the show.  I recommend running sound check a little louder than you will run the show. It will honestly seem just a little louder without bodies in the room even with no changes to any faders or gain knobs.  This is because the human body is made up mostly of water (and one of the best sound absorbing materials is big bags of water). It’s normal for sound check/rehearsal to sound louder than the live performance. Making it a little bit louder will also help you establish any feedback issues that may flare up in the show.  When the audience comes in, they absorb some of the sound, so if you have audience mics that the ear monitor folks are relying on or if your band is mostly on wedges, the change in sound when an audience fills the room will change the performers perspective on their mix.

Now that you have run sound check/rehearsal a bit loud, back that main fader back where it belongs.  You have established a max volume as well as knowing you are stable feedback-wise.  Do not “gain up” any inputs, or push the main fader above this point.  You run the risk of not only feedback, but also splash back on the stage that will overtake the bands monitors.  You know you hit this mark if the band is on wedges and they all start asking for monitor changes.

And last but not least: If you’ve had (and especially if you have not had) a good rehearsal, never make any changes based on what you remember you wanted to change.  Meaning —  if the band has left the stage, don’t think “Oh, I needed some gain on the acoustic guitar and I didn’t want to mess him up in rehearsal…. I’ll add it now.”  Stop right there… go back to Line Check in this post and read it again.

About John Mills:  John is a 20-year veteran of live sound. In addition to his work with touring acts, John is Vice President at Nashville’s Morris Light & Sound www.MorrisLightAndSound.com, writes regularly for Worship Musician and is a great resource for church tech teams with helpful advice on his TechTraining101.com website.   He’s also a featured speaker at many pro audio seminars across the country.