Church Soundguy

Free Book: A Tale of Two Mixers

2012-12-14T15:58:00.000-08:00

There has been a whole lot of hubbub about several new, smaller digital mixers.

A very short book has been written comparing features on the most popular among them: to sort through the hubbub, and examine them in light of the real world. Part tongue-in-cheek, part real-world review, part spec comparison, it's a useful tool. And hey, it's FREE!

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_____________________________________________________

David McLain, CTS | Technical Sales | CCI SOLUTIONS

Be seen. Be heard.

PO Box 481 / 1247 85th Ave SE

Olympia, WA 98507-0481

Voice: 800/426-8664 x2155 / Fax: 360/754-1566

personal email: dmclain@ccisolutions.com

CCI Solutions online: www.ccisolutions.com

Church Soundguy blog: www.churchsoundguy.com

facebook: www.facebook.com/churchsoundguy

twitter: twitter.com/churchsoundguy

Pro Audio Parade Music

2012-06-28T13:43:00.000-07:00

I’ve been researching: How do you do pro audio for a Fourth of July Parade? In this case, we’re trying to give a bunch of dancers some dancing music and share that music with the audience, but we could be playing music on a float or a trailer. I know lots of churches who are involved in parades nowadays, and their entries always involve music. How do we make parade music so that everybody can hear it, and so it doesn't sound like garbage?

We’ve all seen the little battery powered systems; they’re great for a small group in a quiet environment, but they aren’t enough for sound in a parade: the high school marching band two blocks away will overwhelm it. Let's save these for mission trips or fellowship halls (they're pretty good for that!).

I consulted with Fred Tomke an engineer at QSC Audio. Fred knows his stuff: he’s been using his own K12 speakers on top of a bus in his own local Fourth of July Parade for a few years. OK, Fred, what do I need to power them properly?

It turns out that the only thing you need is a competent inverter for the vehicle. He uses a “basic 800 watt” inverter to power his (2) K12 speakers (1000 watts each), a small mixer, and a CD player. He says he’s never run out of headroom. “The secret is in the power supplies on the speakers: they’ll handle anything from 85v to 240v.”

To connect multiple devices (like the mixer, CD player, and multiple amps), just use a power strip. And we ended up using the smaller, broader-dispersion K8 speakers on this project: The smaller size made it easier to load onto their minivan’s roof rack, and the 105º dispersion pattern means more people alongside the parade route will hear it, even if you lay the speakers on their side (as any sensible minivan driver would do!). It still has the same 1000 watt amp built in, so “loud enough” is not an issue.

The little JBL EON210P system will also work nicely in this environment: a little poweredmixer and two 10” main speakers.

There is one important detail: don’t use an inverter that connects via the vehicle’s cigarette lighter. That lighter is limited, typically, to about 5 amps, and you’ll pretty much need all 6.7 amps that an 800 watt inverter can provide. Instead, use one of the inverters that connects directly to the vehicle’s battery, or extend to the battery with 10- or 12- gauge cables.

Oh, and make sure you vehicle is running. This kind of power consumption will drain your battery pretty quickly.

With this kind of setup, you can get loud enough that the parade officials will come tell you to turn the music down! Or you can use this system for your concert-in-the-park after the parade!

Happy Fourth of July.

Using the Presonus Studio Live Mixer

2012-05-30T08:58:00.000-07:00

The Presonus StudioLive digital mixer may be the most popular small mixing board for churches this year. But as easy as it is to operate, it's not the same as an analog board.
Recently, Presonus's Rick Naqvi did a very detailed webinar on the board. It's an excellent source for learning how to use the new board.

Vocal Microphone Technique

2012-01-16T09:15:00.000-08:00

It's always been amusing to watch the band set up. The guitarist brings his amp, a few pedals, and maybe a couple of guitars. The bass player brings his instrument, and often his own amp. The drummer uses the church's drum kit, but he brings his own sticks and takes the time to tune and position the drums to his liking.

But the vocalist just uses whatever mic is handed to them.

My experience has been that the choice of microphone for the vocalists, especially the lead vocalist, has a substantial effect on her sound in the house, her intelligibility, and even her confidence in front of a crowd. Using "whatever they give me" would be like the the guitarist playing "whatever guitar they hand me," whether it's a Fender Squire or a Paul Reed Smith Custom 24 guitar, or the sound guy saying, "Yeah, whatever. Behringer, Midas, Yamaha, Digico: they're all the same."

The point: if you're a vocalist, find a mic that really lets your voice give its best in your facility. If you're the sound guy, then give real thought to what mics sound best on which vocalist, particular the main vocalists. Try out some new ones if you need to, and teach your team that "This is John's mic!" Or encourage John to buy his own vocal mic.

And of course, audio engineers love working with untrained vocalists, who sing away from the mic, lean into the mic for their loud notes, and cup the grille. The reality is that a good sound system will clearly amplify whatever sound (good or bad) that the vocal mic picks up. It is not to the vocalist's advantage to send a poor signal to the sound system.

Audix created this video, and they make some excellent vocal microphones (and some amazing instrument mics), including some at modest prices. Of course, they use Audix mics in these brief clips. But the techniques are appropriate for any handheld vocal microphone.

Note: this post contains a video clip. If you're having a hard time seeing it, click on the title ("Vocal Microphone Technique") to watch the video on the post's home page. And if you want to share the video with your vocalists, use this link: http://j.mp/VocalTechnique.

Two Days Discovering Nexo

2011-12-14T06:49:00.000-08:00

I’ve been hearing a lot of talk about Nexo speakers over the past few years. I’ve listened to various of their models, and I’ve been impressed enough to recommend them for a couple of rooms, particularly given the outstanding support I’ve been getting from them.

But I haven’t really known the Nexo lineup; until I hadn’t had opportunity to listen critically and extensively to their whole selection of speakers.

I have now.

I spent Monday of this week designing speaker systems for rooms using various software, and I was liking the way the Nexo speakers worked in the planning: the plans looked good, but what did they sound like?

The Cerritos Center in the greater Los Angeles area is an awesome building. We'd be using their main room (what a beautiful room!) the next day to try out the Nexo line arrays.

I had a .dwg model of the Cerritos Center, so I modeled out what various speakers would sound like in there. The weekend is sponsored by Nexo, so I focused on Nexo boxes, in this case, line arrays (Nexo has several). I liked what I saw in the computer.

The NX1 software was easy to work with, so I imported some smaller real-world rooms that I’m working on in the real world. Yep. Looked easy. Looked like the PS series might sound good.

I have to admit, there’s a fair bit of skeptic in me. Any box that promises me a rectangular coverage pattern (in this case, Nexo’s PS series) had better do more than just advertise well! It needs to offer actual rectangular coverage pattern. And more importantly it needs to sound good! But they sure looked good on the computer screen: a couple of boxes and a sub in a room that seats 400.

Nexo's PS series
(Apologies for the cruddy
cell-phone-camera photos.)

Finished the designs, closed the computer for the night, and headed out for dinner with 40 engineers and another dozen or so from Yamaha/Nexo (Yamaha distributes Nexo in the US). Got to know a lot of these guys as human beings. Not a one of them was a jerk (and that’s kind of rare, when dealing with representatives of high-end gear!). For a large number of them, church audio is in their blood, not just professionally, but personally: they appear to live out what they teach during the week. I took it as a good sign.

Next morning, we head over to the Cerritos Center, where I met with head of audio Jack Hayback, and head rigger & carpenter, Rogan Gerard. We talked shop of course, and they took us on a tour of the facility. Then we started listening to speakers. We started with Nexo’s PS series. The speakers hang very nicely, but we listened to them on a pole, just above head height.

I fell in love.

Let me just cut to the chase: I never knew “Sound on a Stick” could sound that good. I measured 112 dB at the back of the listening room, and I have to say, it sure didn’t feel like 112 dB. We were playing vocal tracks (I got so tired of “Bird on a Wire” this weekend! I still don’t know the artist.), and it didn’t sound like a PA playing. It sounded like a woman there singing to me. Singing well.

Randy Weitzel had put up a very nice drum kit behind the speakers, and brought in a fine drummer to show them off. We listened to the same drum kit: voiced the same as the original kit, with zero EQ, zero compression: same drum kit, but more of it. We listened with speakers, without speakers. Even the little 8” 2-way sounded way bigger than it was.

(Note: I’m not big on stage monitors, but the Nexo wedges [45N?] were clear, loud, and were so tight in their pattern that even the drum overheads were in the drum monitor!)

The boxes' horns reportedly put out square pattern: I didn’t measure the exact edges of the pattern, but it sure seemed square to me. The coverage was clearly narrower on one end than the other; I could hear that. The previous days' computer exercises seemed to match real world applications.

Then we listened to Nexo’s Baby Line Array: the Geo-S8. An 8” 2-way box, in a couple of 12-box arrays. Pretty good! Clear, articulate, musical, at 90 feet. Fifteen hundred seats of modest folk music would be a fine fit for the baby line arrays, or a few hundred seats of music with teeth! I had done a project with another small line array recently; I'll bet these could have served that room at least as well.

Then we needed to pull these down so we could put up another array to test. We took down two dozen S8 boxes and hung two dozen S12 boxes in less than an hour. OK. I’m impressed. That was easy. Let's go to lunch.

Line Arrays:
Geo-D, Geo-T and Geo-12
(outside to inside)

After lunch, we came back and listened to the S12 boxes, the very boxes I had been using in the design software to fill the computer model of the Cerritos Center. It looked good in the model, though the room might have been a little toward the big side for these.

They fired them up, and behold: they sounded as the modeling showed: clear, articulate, in the entire room. Well, most of the room; the third balcony was a little weak, but the software predicted that. When we added the subwoofers, it was clearly nothing to complain about. Remarkably smooth. Remarkably even, throughout the room. I like those boxes. And yes, the room might have been a little toward the big side for these speakers. I will have no problem recommending these speakers for a medium size church, and they'll make it sound good!

We listened to the Geo-D boxes (it’s kind of weird, in that it’s pronounced: “G O D”). These are the main boxes for this room, and I can see why: effortless excellence. I walked the entire floor, and three balconies, and maybe a dozen of the loge boxes: the entire room sounded the same! It was a little (!) bit louder in some seats than it was behind them, but the voicing was clear everywhere. I’ve heard it said before, but it was true: there wasn’t a bad seat in the house! These boxes are amazing!

I had taken time to talk with Jack Hayback (away from the Yamaha/Nexo boys) about his experience with the Nexo Geo-D speakers, and how they compared to the two other brands before them. His eyes lit up! He had lots of good things to say, a number of stories, and he compared them to the two (other brands) that he had had before he got the Nexos. This was a sincerely happy audio guy!

More significant, the Lighting Director, John Palmer, told me how clear the audio was when he first heard them. (In my experience, it takes a lot for quality sound to impress the LD!)

Lastly, we listened to the Geo-T: the big dogs. These are the famous boomerang shape that you’ve probably seen on major arena tours around the world, and I can believe it. They shook me to my core at more than 100’, both in clarity and in the solidity of their sound. I can see why the Big Names tour with this gear.

Subs, in Cardioid (!!) configuration:
RS 18 & RS 15

I suppose I should mention that all of the subwoofers for the line arrays – which were shaking my pant legs at 100’ – are cardioid subs. During the “fairly loud” cuts (think Sunday morning volume), we could hold a conversation on the stage behind the subwoofers. During the “concert level” cuts, not so much, but we were not overwhelmed on stage: I could have heard monitors easily. (The subs are even louder, about 3dB, in omni mode, but of course, much louder behind.)

We actually spent the next two hours listening to this track and that (and “Bird on a Wire” on every speaker in the house!), listening loud, listening quiet, to all these speakers, not because we needed to, but because we wanted to. Big line arrays, sounding as good as I’ve ever heard, and more. I have never heard so much detail from James Taylor, Pink Floyd and even Rammstein, and several I don't know. I’m not used to describing Rammstein as “beautiful,” but I did this weekend!

It’s not hard for a big line array to play loud music well. But when they can play the same music softly and gently at 60 dB, with the same clarity as at 120 dB, then I’m impressed. It still had solid, believable 60 Hz bass guitar at 60 dB! I'm not used to that.

Summary: These are very good speakers. The entire lineup is worth paying attention to, and in my opinion, probably worth their not insubstantial cost.

They are not the right speakers for everybody; they’re not inexpensive, for one thing.

Anything beyond the PS series probably need professional installation, and certainly need professional design. And while I haven’t seen it, I imagine there may be a room that they just don’t fit well in. The PS series can handle aggressive worship music comfortably in a mid-size church, as can the Geo-8. The Geo 12 will do wonders in a large church. And if your room is big enough, the Geo-D will make it sing! The Geo-T is not, in my opinion, needed for anything much smaller than a stadium concert.

But if you have some room in your budget for quality speakers, do NOT overlook Nexo.

(Obligatory plug: if you would like a speaker system designed for your room – and you have a budget for it, drop me a line, and we can talk about Nexo, or a dozen other lines. But we will talk about Nexo. At least for a little bit.)

Getting Started with Stage Design!

2011-09-17T09:33:00.000-07:00

Veteran Technical Director and CCI Solutions Church Relations Director, Duke DeJong helps you get started creating modern worship stage designs.

Churches across America both big and small are looking for ways to enhance their worship spaces with color and texture. I love this trend because it brings more visual artistry to the church and it helps our relevance to the younger generations who continue to become more and more visual. For churches looking to enter into the world of stage design, here are a few thoughts to get you started.

Finding Resources!

The key is to find out what your resources are. I'm a huge fan of LED and intelligent lights but most churches aren't able to start with those. Many churches have a few spare dimmers and par cans lying around and that can be a great way to get you started. Pick some colors you like, put those gels in your lights and aim them at something reflective. You've now added color to your stage.

Let there Be Light!

Next, find out what type of materials you have to light. I love a stage with darker colored walls so I can put something that lights well in front of it, helping any over flow light to disappear. Some of my favorite things to light include various fabrics like Poly Muslin, Poly Sheen and Spandex (must be fire retardant), Coroplast and even Bubblewrap.

Reflective Properties!

Really anything that reflects light has potential as a design element. You can make great structures out of metal, wood or even PVC and then cover them with something light friendly. The opportunities are endless and to get started you should see what materials people in your church have access to or the ability to work with. Whatever you have access to should dictate what materials you start with.

Wrapping it Up!

If you can't come up with an original idea, find some designs from other people that look like something you can do a variation of and simply try it out. If it works first try, awesome! For most of the designs I've done I will see elements I like somewhere and then I do a little tweaking and adjusting to make it something that works right for my space. If something doesn't work just right one week, try tweaking it for next week. The key is to use whatever resources you have access to, find a concept you like and try it out. If you ever want to bounce ideas off of someone or need some ideas to get you started, let me know.

Duke DeJong
Church Relations Director
CCI Solutions

Duke has more than 12 years of experience as a technical artist, trainer and collaborator for ministries. Duke travels around the country for CCI Solutions and is available to help your ministry. Join Duke on Facebook at www.facebook.com/ccisolutions.

The Art (and Necessity) of Compression

2011-07-15T20:54:00.000-07:00

By Duke DeJong
Church Relations Director, CCI Solutions

A while back, I got the rare opportunity to work with the youth band at our church. These guys have an incredible heart and passion to worship and have loads of raw talent which translates into a powerful time of worship. When they lead, as a worshipper I feel free and emboldened to praise God the way He created me too. When they lead, as a sound man I have to work as hard and quick as ever to create a decent mix to help facilitate that worship.

More Compressors Please!

As is the case with most youth bands and even many churches, they are not using state of the art or high dollar gear for their services. Now don't get me wrong, they are not operating with the bare minimum. The system includes an Allen and Heath console, JBL speakers and subs, and solid system and signal processing. What I longed for that night was individual compressors.

Keeping the Vocals on Top!

Maybe this never happens to you, but in a mix including 3 vocals, an un-caged drum set, two electrics, acoustic, bass, and keyboard, I had a hard time keeping the vocals out on top to lead the group in worship while keeping the music strong. The vocalists on the team are gifted in leading worship, but for a variety of reasons (key of the song, dynamic range, mic etiquette, etc) their volumes were all over the place that night and the second I took my finger off their faders I would either lose them or have way too much of them. With 7 stage monitors, acoustic drums, 3 amps and a very small stage, I was dreaming for a few compressors to help me layer the mix the way I wanted.

Sound Man's Most Useful Tool!

The compressor is one of the sound man's most useful tools - yet I am always surprised how few people seem to understand and know how to effectively use this critical piece of gear. I would like to help a few more of you get comfortable using compressors.

So What is Compression Really?

The clearest definition of compression that I've ever seen is this: "Compression is the art of making louder parts of a composition appear softer, and conversely, the softer parts appear louder." That night, if I would have left the lead singer's fader in one spot for the entire night his volume alone would have ranged anywhere from 85 dB to 120 dB. Alright that might be an exaggeration but he got loud. When he was closer to the 85-95 dB volume he could barely be heard over the drums and guitars. Neither end of the spectrum is really acceptable in a good mix, so compression comes along and makes it possible to narrow down that volume range to make things more mixable.

Example of How an Audio Compressor Works

Let's say I have a 20 dB range between a vocalist's quiet singing versus their loudest singing. With a compressor I can take that 20 dB range and make it as small as a 1 dB range, but since I don't want to eliminate the artistic dynamic range that the singer is using to create the mood or feel of what they are singing, I can get that range down to a very manageable 5-8 dB that will make mixing significantly less complicated but still leave some of that dynamic in place. So how do we get our compressor to do that? With some understanding of the compressor's settings you can be on your way to a smoother sound and a less stressful time behind the board.

The Control Elements Found on an Audio Compressor!

Threshold

In simple terms the threshold is the point where the compressor starts to do its thing. Since there is a wide range of compressor and mixer brands I'm going to talk about these settings more generically as opposed to using the numbers on the knob. If the input meter on your console (let's say negative infinity to +15 dB) matches that of your compressor, things will be a little clearer as the numbers will match. You must first set the gain (or trim) of your channel on your mixer (on my regular console that is around +3, or typically where the green lights first turn to yellow or maybe the yellow light just starts to glow on the meter). Now if your numbers match, and your vocal meter is showing signal between -5 and +10 dB, I'd start with my threshold set close to 0 dB. If your numbers don't match, once your gain is set turn your threshold knob and find the area where the gain reduction knobs just come on. Begin with your threshold there and if you find it's not compressing frequently or soon enough you can lower the threshold from there to make it kick in sooner.

Ratio

This one is a relatively simple concept. The ratio simply says for every x dB the source goes up in volume, the compressor will only let the output go up y dB. For example, if you set a vocal mic with a 3:1 ratio, for every 3 dB the vocal increases coming into the board, the output will only increase 1 dB. You can think of the ratio as setting the size potential of the source. If you want it to be able to go bigger, you can leave your ratio smaller. If you want it to stay a little smaller, or perhaps be more under control, you can set your ratio higher. I tend to start with a ratio of 3:1 for most vocals and guitars, and often times I will go 4:1 or even 5:1 on drums or very dynamic guitars. My preference is to start low and if you need more compression (less range) you can always increase the ratio. The reason it is my preference is simply this, I don't want to take away any more control from the musicians than is absolutely necessary to make the mix work well. If I start it at 5:1 when 3:1 will do and don't adjust it down, I may be holding that source back. If I start low and it's still too big, I can easily adjust my ratio up.

Attack

The attack is how quickly the compressor responds to the volume change. A slower attack will sound a little smoother, rounding out the sound of your source a little bit and in essence making it sound a little "fatter". A slower attack will generally be less noticeable which can be good for vocals and some thin or scratchy guitars. Setting your attack to a faster setting can be great for instruments such as drums or any other very aggressive instruments. A faster attack will give an instrument more of an aggressive, pumping feel, and potentially bring out more of the high end edginess. The ultimate decider on where to set this is by listening. I tend to set vocals a little slower, guitars in the middle, and drums faster to start. From there, if you need a little more aggressiveness or snap you can speed up the attack, and if it needs to be a little smoother or fuller you can slow it down. As in all things with sound, let what you hear guide your settings and adjust until you are happy.

Release

The release is the back side of the attack, and sets how quickly you want to release the compression once that loud burst is over. As with the attack, a slower release will sound smoother and less noticeable but could end up taking some of the aggressiveness out of aggressive instruments by compressing them when they don't need to be. I again will tend to start a little slower for vocals, middle of the road for guitars, and faster for drums. You'll want to again experiment with where to set this by listening to the sound. If the source sounds like it is pumping a little bit, slow the release down to help even it out a bit. If it feels like you might be losing something on the next note/beat, you likely need to speed the release up a bit. Again, let the sound of the source guide you to where it should be set. Listen and adjust until it sounds right to you.

Output

Most compressors have an output to help boost the volume of the end result, and here's where I tend to see a lot of mistakes made. Now that you've taken that 85 to 105 dB vocal and compressed it down to a manageable 85 to 93 dB, you may need to increase the output a little to get it over those guitars and drums. Instead of reaching for the gain or trim knobs (which would then bring more signal into the compressor and would change how you've set your compressor), if you add 5 dB of gain to your output you just took that 85-93 dB and made it 90-98 dB.

Especially Useful in Worship Environments!

Compressors are a huge help to the sound man and used right they will help you get great sound out of your sources and give you the ability to get the mix where you want it. Compressors are especially useful in the worship environment where the voice of those leading the worship must always be present but not piercing, where more and more guitars are being used to lead the music but can't overtake the vocals, and where many churches use acoustic drums.

Wrap Up!

I truly believe that no one setting is right for any vocal or instrument. If you start with a lower basic setting and then adjust based off of what you are hearing, your compressors can give you a great edge to get your mix balanced and layered according to plan. Just remember, you don't want to compress something more than you need to. If you're having trouble keeping a source in it's place in the mix the compressor is the tool to help you make that happen.

Duke DeJong
Church Relations Director
CCI Solutions

A Detailed Guide To Constant-Voltage ("70v") Audio Systems

2011-06-24T11:20:00.000-07:00

Clarifying and defining key power aspects with constant-voltage (or high-impedance) systems

Electric power companies have a good idea that has been applied to audio engineering.

When they run power through miles of cable, they minimize resistive power loss by running the power as high voltage and low current.

To do this, they use a step-up transformer at the power station and a step-down transformer at each customer’s location. This reduces power loss due to the I2R heating of the power cables.

The same solution can be applied to audio communications in the form of a constant-voltage system (typically 70 volts in the U.S. and 100V overseas).

Such a system is often used when a single power amplifier drives many loudspeakers through long cable runs (over 50 feet). Some examples of this condition are distributed speaker systems for PA, paging, or low-SPL background music.

BACKGROUND
The label “constant voltage” has been confusing because the voltage is really not constant in an audio program. A better term might be “high impedance.”

A typical high-impedance system is shown in Figure 1. A transformer at the power amplifier output steps up the voltage to approximately 70 volts at full power.

Each loudspeaker has a step-down transformer that matches the 70-volt line to each loudspeaker’s impedance.

The primaries of all the loudspeaker transformers are paralleled across the transformer secondary on the power amplifier.

Figure 1. A typical high-impedance system using a step-up transformer on the amplifier output.

There are three options at the power-amp end for 70-volt operation:
• an external step-up transformer
• a built-in step-up transformer
• a high-voltage, transformerless output

These options are covered in detail later in this article.

The signal line to the loudspeakers is high voltage, low current, and usually high impedance. Typical line values for a 100-watt amplifier are 70 volt, 1.41 amperes, and 50 ohms.

How did the 70-volt line get its name? The intention was to have 100-volt peak on the line, which is 70.7 volts rms.

The technically correct value is 70.7 volt rms, but “70-volt (or “70V") is the common term. There are 70 volts on the line as maximum amplifier output with a sine wave signal. The actual voltage depends on the power amplifier wattage rating and the step-up ratio of the transformer. The audio program voltage in a 70V system might not even reach 70V. Conversely, peaks in the audio program might exceed 70V.

Other high-voltage systems might run at other voltages. Although rare, the 200V system has been used for cable length exceeding one mile.

ADVANTAGES OF 70V OPERATION
As stated before, a 70V line reduces power loss due to cable heating.

That’s because the loudspeaker cable carries the audio signal as a low current.

Consequently you can use smaller-gauge loudspeaker cable, or very long cable runs, without losing excessive power.

Another advantage of 70V operation is that you can more easily provide the amplifier with a matching load. Suppose you’re connecting hundred of loudspeakers to a single 8-ohm amplifier output. It can be difficult to wire the loudspeakers in a series-parallel combination having a total impedance of 8 ohms.

Also it’s bad practice to run loudspeakers in series because if one loudspeaker fails, all of the loudspeakers in series are lost. This changes the load impedance seen by the power amplifier.

With a 70V system you can hang hundreds of loudspeakers in parallel on a single amplifier output if you provide a matching load. Details of impedance matching are covered later. In addition, a 70V distributed system is relatively easy to design, and allows flexibility in power settings.

Let’s compare a standard low-impedance system to a constant-voltage system. Imagine that you want to provide PA for a runway at an airshow. A low-impedance system might employ 30 speaker clusters spaced 100 feet apart, each cluster powered by a 1000W amplifier for extra headroom. A high-impedance version of that system might use only one amplifier providing 140V. The cost savings is obvious.

DISADVANTAGES OF 70V OPERATION
One disadvantage of a 70V system is that the transformers add expense. Particularly if you use large transformers for extended low-frequency response, the cost per transformer may run $70 to $200. Low-power paging systems, or those with limited low-frequency response, can use small transformers costing around $4.95 each. Many loudspeakers are sold with 70V transformers included.

Another disadvantage is that transformers can degrade the frequency response and add distortion. In addition, a 70V line may require conduit to meet local building code.

TRANSFORMERS
The main component of a 70V system is the loudspeaker transformer.

Its secondary winding has taps at various impedances. You choose the tap that matches the loudspeaker impedance.

For example, if you’re using a 4-ohm loudspeaker, connect it between the 4-ohm tap and common.

The primary winding has taps at several power levels. These power taps indicate how much maximum power the loudspeaker receives. For example, suppose you have a 70V transformer with the primary tapped at 10W and the secondary tapped at 8 ohms. Then a loudspeaker rated at 8 ohms should receive 10W at its voice coil when the primary is connected to a 70V line.

Transformers have insertion loss mainly due to resistance. Precise system calculations should take insertion loss into account. These calculations are covered in the Appendix later in this article.

INSTALLATION
With this background in mind, let’s proceed to installation practices. Here’s a basic procedure that neglects transformer insertion loss:
1. Do NOT connect the 70V loudspeaker line to the power amplifier yet.
2. Install a transformer at each loudspeaker location, or use loudspeakers with built-in transformers.
3. Connect each loudspeaker to its transformer secondary tap. The tap impedance should equal the loudspeaker impedance.
4. Connect each transformer primary to the 70V line from the power amplifier. Choose the tap that will deliver the desired wattage to that loudspeaker.
5. Add the wattage ratings of all the primary taps. This sum must not exceed the amplifier’s wattage rating. If it does, change to a lower-wattage primary tap of one or more transformers, or use a higher power amplifier.
6. Connect the 70V loudspeaker line to the 70V output of the amplifier.

As an example, suppose you are setting up a 70V system with 8-ohm loudspeakers and a 60W power amp. Connect the 8-ohm secondary taps to each speaker. Suppose the total loudspeaker wattage is 55 watts. This is acceptable because it does not exceed the amplifier power rating of 60 watts.

Here’s a more detailed procedure that emphasizes impedance matching:
1. Compute the minimum safe load.

The minimum safe load impedance that can be connected to the amplifier is given by:

where
Z = minimum safe load impedance, in ohms.
E = loudspeaker line voltage (25V, 70.7V, 100V, etc.)
P = maximum continuous average power rating of power amplifier, in watts.

An example: For an amplifier rated at 100 watts continuous average power, the minimum load impedance that may be connected safely to the 70.7V output is:

2. Choose transformer taps.
Tap the primary at the desired power level for the loudspeaker, and tap the secondary at the impedance of the loudspeaker. The sum of all the power taps for all the loudspeakers should not exceed the power output of the amplifier.

Note: Changing the power tap also changes the load impedance seen by the amplifier. Raising the power tap lowers the load impedance, and vice versa.

Also, changing the power tap changes the SPL of the loudspeaker. Reducing the power tap by half reduces the SPL by 3 dB, which is a just-noticeable difference in speech sound level.

If a particular loudspeaker is too loud or too quiet, you can change its power tap. Just be careful that the total power drain does not exceed the power output of the amplifier.

3. Connect the loudspeakers together.
Connect all the loudspeaker-transformer primaries in parallel. Run a single cable, or redundant cables, back to the power-amplifier transformer secondary. But DON’T CONNECT IT YET.

4. Measure the load impedance.
Before connecting the load, first measure its impedance with an impedance bridge (a simple low-cost unit is adequate). Here’s why you must do this: If the load impedance is too low, the power amplifier will be loaded down and may overheat or distort. It’s a myth that your can connect an unlimited number of loudspeakers to a 70V line.

If the load impedance measures too low, re-tap all of the loudspeakers at the next-lower power tap. This raises the load impedance. Measure again.

Usually, it’s no problem if the load impedance measures higher than the matching value (the calculated minimum safe load impedance). The system will work, but at reduced efficiency. Typically there is more than enough power available, so efficiency is not a problem.

If for some reason power the power is limited, then the system should be wired for maximum power transfer. This occurs when the measured load impedance matches the calculated minimum safe load impedance. If the load impedance measures above this value, you can re-tap all the loudspeakers at the next-higher power tap and measure again. This tap change lowers the load impedance.

Many people don’t realize that a transformer labeled for use with a specific voltage will work just as well at other voltages. See the constant voltage calculator here. It determines the power delivered from a transformer tap when driven with other than the rated voltage.

PRECAUTIONS
Since a 70V line is relatively high-impedance, it is more sensitive to partial shorts than a low-impedance line. Consequently, you may want to avoid running 70V lines in underground conduit which may leak water.

Use high-quality transformers with low insertion loss. Otherwise, the power loss in the transformer itself may negate the value of the 70V system.

Avoid driving small transformers past their nominal input voltage rating. Otherwise, they will saturate, draw more than the indicated power (possibly overload the amplifier) and will distort the signal.
You may want to insert a high-pass filter ahead of the power amplifier to prevent strong low-frequency transients which can cause core saturation.

The CTs amplifiers include a high-pass filter that can be selected at 70 Hz, 35 Hz, or bypass. The CH amplifiers insert a 70 Hz high-pass filter when placed in high-impedance mode.

POWER-AMPLIFIER OPTIONS
As stated earlier, there are three power-amplifier options for 70V operation: The amplifier might have:
• an external step-up transformer
• a built-in step-up transformer
• a high-voltage, transformerless output

Let’s consider each option.

Amplifier with external transformer
This system is shown in Figure 1 (on page 1). If you use an external transformer, select one recommended or supplied by the amplifier manufacturer.

If you have a conventional amplifier with low-impedance outputs only, and you want 70V or 100V operation, Crown has the needed accessories. The TP-170V is a panel with four built-in autoformers that convert four low-impedance outputs to high impedance. The T-170V is a single autoformer for the same purpose.

Choose a transformer with a power rating equal to or exceeding the wattage of the power amplifier. The turns ratio should be adequate to provide 70.7V at the secondary when full sine-wave power is applied to the primary. Use the following formula for a 70.7V line:

where
T = turns ratio
70.7 = voltage of constant-voltage line
P = amplifier power output in watts
Z = amplifier rated impedance
SQR means square root

Better yet, measure the amplifier’s output voltage at full power into its rated load impedance, and use the formula:

where
T = turns ratio
70.7 = voltage of constant-voltage line
E = measured output voltage at full power into the rated impedance.
Amplifier with built-in transformer
If the transformer is already built into the power amplifier, simply look for the output terminal labeled “70V,” “25V,” “100V,” or “high impedance.”

Amplifier with transformerless, high-voltage output
Figure 2 shows how a power amplifier with a high output voltage can power a distributed system without a step-up transformer.

Figure 2. A constant-voltage system using a high-voltage power amplifier.

Many high-power amplifiers can drive 70V lines directly without an output transformer. For example, Crown CH amplifiers have an auto transformer (except CH 4). CTs amplifiers can provide direct constant-voltage (70V/100V/140V/200V) or low-impedance (2/4/8 ohm) operation.

In Dual Mode, the CTs 600/1200 can power 25/50/70V lines; the CTs 2000/3000 can power 25/50/70/100V lines. In Bridge-Mono mode, the CTs 600/1200 can power 140V lines; the CTs 2000/3000 can power 140V and 200V lines.

With CTs Series amps, one channel can drive low-impedance loudspeakers, while another channel drives loudspeakers with 70V transformers. This makes it easy to set up a system with large, low-Z loudspeakers for local coverage and distributed 70V loudspeakers for distant rooms—all with a single amplifier.

The Crown CTs 2000 adept at providing constant power levels into various loads. In dual mode, it delivers 1000 watts into 2/4/8 ohms and into a 70V line. In bridge-mono mono, it delivers 2000 watts into 4, 8, or 16 ohms, 2000 watts into a 140V line, and 2000 watts into a 200V line.

Crown Commercial Audio series of amplifiers and mixer-amps provide both low-Z and constant-voltage operation. For example, the 180MA and 280MA mixer-amps offer 4-ohm, 70V and 100V outputs.

Pros and cons of transformerless systems
The high-voltage, transformerless approach eliminates the drawbacks of amplifier transformers:
• cost
• weight
• limited bandwidth
• distortion
• core saturation at low frequencies.

On the other hand, transformers are useful to prevent ground loops, ultrasonic oscillations and RFI. Some local ordinances require transformer-isolated systems.

Let’s look at the core-saturation problem in more detail. Sound systems can generate unwanted low frequencies, due to, say, a dropped microphone or a phantom-powered mic pulled out of its connector.

Low frequencies at high power tend to saturate the core of a transformer. The less the amount of iron in the transformer, the more likely it is to saturate.

Saturation reduces the impedance of the transformer, which in turn may cause the amplifier to go into current limiting. When this occurs, negative voltage spikes are generated in the transformer that travel back to the amplifier—a phenomenon called flyback. The spikes cause a raspy, distorted sound. In addition, the extreme low-impedance load might cause the power amplifier to fail.

Some Crown amplifiers are designed with high-current capability to tolerate these low-frequency stresses.

Production amplifiers are given a “torture test.” Each amplifier must deliver a 15-Hz signal at full power into a saturated power transformer for 1 second without developing a hernia!

Many transformers are reactive, so their impedance varies with frequency. Some 8-ohm transformers measure as low as 1 ohm at low frequencies. That’s another reason for specifying an amplifier with high current capability.

CONCLUSION
Using a high-voltage system greatly simplifies the installation of multiple-loudspeaker PA systems. It also minimizes power loss in the loudspeaker cables. If you take care that your load does not exceed the power and impedance limits of your power amplifier, you’ll be rewarded with a safe, efficient system.

APPENDIX: HISTORY OF CONSTANT-VOLTAGE SYSTEMS
In early industrial sound systems, multiple loudspeakers were carefully configured to provide a matching impedance load to the amplifier. But as these systems grew in size, several problems arose: how to connect multiple loudspeakers to the same amplifier without loading it down, how to individually control the sound power level fed to those loudspeakers, and how to overcome the power loss associated with the typically long lines that ran between the power amp and loudspeakers.

By the late 1920s and early 1930s the “step-up, step-down” idea has been applied to loudspeaker lines in what has become known as “constant voltage” distributed systems. (Radio Physics Course 2nd Ed., Radio Technical Publishing co., N.Y., 1931).

Various voltages have been tried such as 25, 35, 50, 70, 100, 140, and 200 volts, but the 70V system has become the most widespread.

After World War II, we find constant-voltage systems depicted in such reference works as Radio Engineering 3rd Ed. (McGraw-Hill, N.Y., 1947). By the end of that decade, several standards had evolved to regulate 70V specifications for amplifiers and transformers. (Radio Manufacturer’s Association, SE-101-A And SE-106, both from July 1949). In the 1950’s we find the use of 70V systems very well established as evidenced by Radiotron Designer’s Handbook 4th Ed. (RCA, N.J., 1953 and Radio Engineering Handbook 5th Ed. (McGraw-Hill, N.Y., 1959).

As component design improved, 70V systems began to achieve high-fidelity status, but there were two weak links in the chain: the step-up and step-down transformers. Good broadband transformers that could resist core saturation and distortion were expensive.

Half of this problem was solved in 1967 When Crown International introduced the DC-300. It was most likely the first high-powered low-distortion solid-state power amplifier capable of directly driving a 70V line without a step-up transformer. And in June 1987, the Macro- Tech 2400 was introduced with the capability of directly driving a 100V line. Thus, today only loudspeaker needs a transformer to step down the voltage.

APPENDIX: TRANSFORMER INSERTION LOSS
Transformers have insertion loss (power loss due mainly to resistance). This loss should be included in system calculations for precision.

Converted to a power ratio, insertion loss can be expressed as

PR = 10 (L/10)

where
PR = power ratio
L = insertion loss in dB (always a positive number).

Some transformer manufacturers compensate for insertion loss by adding extra windings. In that case, the power delivered to the loudspeaker is the rated value of the tap. The primary draws the rated power times the power ratio of the insertion loss.

In this case, you can calculate the primary impedance as follows:

Pt = Ps + L

where
Pt = total power in dBm
Ps = power to the loudspeaker in dBm
L = insertion loss in dB

Pt = Ps * L

where
Pt = total power in watts
Ps = power to loudspeaker in watts
L = insertion loss (as a ratio).

Then the primary impedance is calculated as follows:

Z = (70.7)2/Pt = 5000/(Ps * 10(L/10))

where
Z = primary impedance in ohms
Pt = total power in watts
Ps = power to loudspeaker in watts
L = insertion loss in dB.

Other transformer manufacturers do not compensate for insertion loss. In this case, the primary impedance matches its rating. However, the power delivered to the loudspeaker is less than the power applied, due to the insertion loss.

Ps = Ptr/L

where
Ps = power to loudspeaker in watts
Ptr = power drawn by transformer in watts
L = insertion loss (as a ratio)

To determine whether a transformer is compensated, measure the power (E2/Z) delivered to the loudspeaker when connected to 70.7 volts. If it is less than the rated power, the transformer is not compensated for insertion loss.

When making loudspeaker SPL calculations based on sensitivity ratings, subtract the insertion loss in dB from the loudspeaker sensitivity rating (if the transformer is not compensated for insertion loss). In transformers that compensate for insertion loss, the speaker receives the power indicated. Consequently, each transformer draws a little more power from the line than is indicated. The final impedance will be too low if you add power equal to the amplifier power.

With non-compensated transformers, the labeled power is not the power received, so the loudspeaker SPL will be lower than calculated. The impedance will read correctly, but the acoustic output will be lower than expected.

APPENDIX: LINE LOSS
See the line loss calculator here.

REFERENCES
Daniels, Drew. Notes on 70-Volt and Distributed System Presentation, db, March/April 1988.

Davis, Don. Sound System Engineering, 2nd Ed., Indianapolis, Howard W. Sams Co., 1987, pp. 85-87, 402- 405. 138905-1 10-05

This article provided by Crown Audio, via ProSoundWeb.

Understanding Constant-Voltage ("70v") Audio Distribution Systems

2011-06-23T11:28:00.000-07:00

25, 70.7 & 100 Volts; U.S. Standards; Just What is "Constant" Anyway?; Voltage Variations -- Make Up Your Mind; Calculating Losses -- Chasing Your Tail

by Dennis A. Bohn

Constant-voltage is the common name given to a general practice begun in the late 1920s and early 1930s (becoming a U.S. standard in 1949) governing the interface between power amplifiers and loudspeakers used in distributed sound systems.

Installations employing ceiling-mounted loudspeakers, such as offices, restaurants and schools are examples of distributed sound systems.

Other examples include installations requiring long cable runs, such as stadiums, factories and convention centers.

The need to do it differently than you would in your living room arose the first time someone needed to route audio to several places over long distances. It became an economic and physical necessity. Copper was too expensive and large cable too cumbersome to do things the home hi-fi way.

Stemming from this need to minimize cost, maximize efficiency, and simplify the design of complex audio systems, thus was born constant-voltage. The key to the solution came from understanding the electric company cross-country power distribution practices. They elegantly solved the same distribution problems by understanding that what they were distributing was power, not voltage.

Further they knew that power was voltage times current, and that power was conserved. This meant that you could change the mix of voltage and current so long as you maintained the same ratio: 100 watts was 100 watts—whether you received it by having 10 volts and 10 amps, or 100 volts and 1 amp. The idea bulb was lit. By stepping-up the voltage, you stepped-down the current, and vice-versa.

Therefore to distribute 1 megawatt of power from the generator to the user, the power company steps the voltage up to 200,000 volts, runs just 5 amps through relatively small wire, and then steps it back down again at, say, 1000 different customer sites, giving each 1 kilowatt. In this manner large gauge cable is only necessary for the short direct run to each house. Very clever.

Applied to audio, this means using a transformer to step-up the power amplifier’s output voltage (gaining the corresponding decrease in output current), use this higher voltage to drive the (now smaller gauge wire due to smaller current) long lines to the loudspeakers, and then using another transformer to step-down the voltage at each loudspeaker. Nothing to it.

U.S. Standards—Who Says?
This scheme became known as the constant-voltage distribution method. Early mention is found in Radio Engineering, 3rd Ed. (McGraw-Hill, 1947), and it was standardized by the American Radio Manufacturer’s Association as SE-101-A & SE-106, issued in July 1949 [1]. Later it was adopted as a standard by the EIA (Electronic Industries Association), and today is covered also by the National Electric Code (NEC) [2].

Basics—Just What is “Constant” Anyway?
The term “constant-voltage” is quite misleading and causes much confusion until understood. In electronics, two terms exist to describe two very different power sources: “constant-current” and “constant-voltage.”

Constant-current is a power source that supplies a fixed amount of current regardless of the load; so the output voltage varies, but the current remains constant.

Constant-voltage is just the opposite: the voltage stays constant regardless of the load; so the output current varies but not the voltage. Applied to distributed sound systems, the term is used to describe the action of the system at full power only. This is the key point in understanding. At full power the voltage on the system is constant and does not vary as a function of the number of loudspeakers driven, that is, you may add or remove (subject to the maximum power limits) any number of loudspeakers and the voltage will remain the same, i.e., constant.

The other thing that is “constant” is the amplifier’s output voltage at rated power—and it is the same voltage for all power ratings. Several voltages are used, but the most common in the U.S. is 70.7 volts rms. The standard specifies that all power amplifiers put out 70.7 volts at their rated power. So, whether it is a 100 watt, or 500 watt or 10 watt power amplifier, the maximum output voltage of each must be the same (constant) value of 70.7 volts.

Figure 1 diagrams the alternative series-parallel method, where, for example, nine loudspeakers are wired such that the net impedance seen by the amplifier is 8 ohms. The wiring must be selected sufficiently large to drive this low-impedance value.

Applying constant-voltage principles results in Figure 2. Here is seen an output transformer connected to the power amplifier which steps-up the full-power output voltage to a value of 70.7 volts (or 100 volts for Europe), then each loudspeaker has integrally mounted step-down transformers, converting the 70.7 volts to the correct low-voltage (high current) level required by the actual 8 ohm speaker coil.

It is common, although not universal, to find power (think loudness) taps at each speaker driver. These are used to allow different loudness levels in different coverage zones. With this scheme, the wire size is reduced considerably from that required in Figure 1 for the 70.7 volt connections.

Becoming more popular are various direct-drive 70.7 volt options as depicted in Figure 3. The output transformer shown in Figure 2 is either mounted directly onto (or inside of) the power amplifier, or it is mounted externally.

In either case, its necessity adds cost, weight and bulk to the installation. An alternative is the direct-drive approach, where the power amplifier is designed from the get-go (I always wanted to use that phrase, and I sincerely apologize to all non-American readers from having done so) to put out 70.7 volts at full power. An amplifier designed in this manner does not have the current capacity to drive 8 ohm low-impedance loads; instead it has the high voltage output necessary for constant-voltage use—same power; different priorities.

Quite often direct-drive designs use bridge techniques which is why two amplifier sections are shown, although single-ended designs exist. The obvious advantage of direct-drive is that the cost, weight and bulk of the output transformer are gone. The one disadvantage is that also gone is the isolation offered by a real transformer. Some installations require this isolation.

Voltage Variations—Make Up Your Mind
The particular number of 70.7 volts originally came about from the second way that constant-voltage distribution reduced costs:

Back in the late ‘40s, UL safety code specified that all voltages above 100 volts peak ("max open-circuit value") created a “shock hazard,” and subsequently must be placed in conduit—expensive—bad.

Therefore working backward from a maximum of 100 volts peak (conduit not required), you get a maximum rms value of 70.7 volts (Vrms = 0.707 Vpeak). [It is common to see/hear/read “70.7 volts” shortened to just “70 volts”—it’s sloppy; it’s wrong; but it’s common—accept it.]

In Europe, and now in the U.S., 100 volts rms is popular. This allows use of even smaller wire. Some large U.S. installations have used as high as 210 volts rms, with wire runs of over one mile.

Remember: the higher the voltage, the lower the current, the smaller the cable, the longer the line. [For the very astute reader: The wire-gauge benefits of a reduction in current exceeds the power loss increases due to the higher impedance caused by the smaller wire, due to the current-squared nature of power.]

In some parts of the U.S. safety regulations regarding conduit use became stricter, forcing distributed systems to adopt a 25 volt rms standard. This saves conduit, but adds considerable copper cost (lower voltage = higher current = bigger wire), so its use is restricted to small installations.

Calculating Losses—Chasing Your Tail
As previously stated, modern constant-voltage amplifiers either integrate the step-up transformer into the same chassis, or employ a high voltage design to direct-drive the line. Similarly, constant-voltage loudspeakers have the step-down transformers built-in as diagrammed in Figures 2 and 3.

The constant-voltage concept specifies that amplifiers and loudspeakers need only be rated in watts. For example, an amplifier is rated for so many watts output at 70.7 volts, and a loudspeaker is rated for so many watts input (producing a certain SPL). Designing a system becomes a relatively simple matter of selecting speakers that will achieve the target SPL (quieter zones use lower wattage speakers, or ones with taps, etc.), and then adding up the total to obtain the required amplifier power.

For example, say you need (10) 25 watt, (5) 50 watt and (15) 10 watt loudspeakers to create the coverage and loudness required. Adding this up says you need 650 watts of amplifier power—simple enough—but alas, life in audioland is never easy. Because of real-world losses, you will need about 1000 watts.

Figure 4 shows the losses associated with each transformer in the system (another vote for direct-drive), plus the very real problem of line-losses. Insertion loss is the term used to describe the power dissipated or lost due to heat and voltage-drops across the internal transformer wiring. This lost power often is referred to as I2R losses, since power (in watts) is current-squared (abbreviated I2) times the wire resistance, R.

This same mechanism describes line-losses, since long lines add substantial total resistance and can be a significant source of power loss due to I2R effects. These losses occur physically as heat along the length of the wire.

You can go to a lot of trouble to calculate and/or measure each of these losses to determine exactly how much power is required [3], however there is a Catch-22 involved: Direct calculation turns out to be extremely difficult and unreliable due to the lack of published insertion loss information, thus measurement is the only truly reliable source of data.

The Catch-22 is that in order to measure it, you must wait until you have built it, but in order to build it, you must have your amplifiers, which you cannot order until you measure it, after you have built it!

The alternative is to apply a very seasoned rule of thumb: Use 1.5 times the value found by summing all of the loudspeaker powers. Thus for our example, 1.5 times 650 watts tells us we need around 975 watts.

Wire Size—How Big Is Big Enough?
Since the whole point of using constant-voltage distribution techniques is to optimize installation costs, proper wire sizing becomes a major factor. Due to wire resistance (usually expressed as ohms per foot, or meter) there can be a great deal of engineering involved to calculate the correct wire size.

The major factors considered are the maximum current flowing through the wire, the distance covered by the wire, and the resistance of the wire. The type of wire also must be selected. Generally, constant-voltage wiring consists of a twisted pair of solid or stranded conductors with or without a jacket.

For those who like to keep it simple, the job is relatively easy. For example, say the installation requires delivering 1000 watts to 100 loudspeakers. Calculating that 1000 watts at 70.7 volts is 14.14 amps, you then select a wire gauge that will carry 14.14 amps (plus some headroom for I2R wire losses) and wire up all 100 loudspeakers. This works, but it may be unnecessarily expensive and wasteful.

Really meticulous calculators make the job of selecting wire size a lot more interesting. For the above example, looked at another way, the task is not to deliver 1000 watts to 100 loudspeakers, but rather to distribute 10 watts each to 100 loudspeakers. These are different things. Wire size now becomes a function of the geometry involved.

For example, if all 100 loudspeakers are connected up daisy-chain fashion in a continuous line, then 14.14 amps flows to the first speaker where only 0.1414 amps are used to create the necessary 10 watts; from here 14.00 amps flows on to the next speaker where another 0.1414 amps are used; then 13.86 amps continues on to the next loudspeaker, and so on, until the final 0.1414 amps is delivered to the last speaker.

Well, obviously the wire size necessary to connect the last speaker doesn’t need to be rated for 14.14 amps. For this example, the fanatical installer would use a different wire size for each speaker, narrowing the gauge as he went. And the problem gets ever more complicated if the speakers are arranged in an array of, say, 10 x 10, for instance.

Luckily tables exist to make our lives easier. Some of the most useful appear in Giddings [3] as Tables 14-1 and Table 14-2 on pp. 332-333. These provide cable lengths and gauges for 0.5 dB and 1.5 dB power loss, along with power, ohms, and current info. Great book. Table 1 below reproduces much of Gidding’s Table 14-2 [4].

References
1. Langford-Smith, F., Ed. Radiotron Designer’s Handbook, 4th Ed. (RCA, 1953), p. 21.2.
2. Earley, Sheehan & Caloggero, Eds. National Electrical Code Handbook, 5th Ed. (NFPA, 1999).
3. See: Giddings, Phillip Audio System Design and Installation (Sams, 1990) for an excellent treatment of constant-voltage system designs criteria; also Davis, D. & C. Sound System Engineering, 2nd Ed. (Sams, 1987) provides a through treatment of the potential interface problems.
4. Reproduced by permission of the author and Howard W. Sams & Co.

Supplied by Rane via ProSoundWeb.

What You Need To Know About Wireless Systems

2011-06-10T20:17:00.000-07:00

An in-depth yet easy-to-understand discussion of wireless systems, how they operate, issues that can plague performance, and solutions that do the trick in the vast majority of situations.

Editor’s Note: This article provides straightforward explanations of the primary issues that account for a full 80 to 90 percent of all wireless microphone system problems, while also presenting solutions that will do the trick in most cases.

However, keep in mind that the best solution is avoiding these problems from the outset. Certainly this won’t guarantee completely trouble-free operation, but the odds dramatically improve.
This compilation of wireless system knowledge is provided by several highly qualified professionals, with Gary Stanfill, who has worked with wireless and related technologies for more than 40 years, topping this list.

Part 1: PSW Wireless Primer

Getting Started
Anyone who has used wireless microphone systems for even a short time doesn’t need to be sold on their advantages. “Going wireless” allows concentration on the message rather than on the mechanics of delivering the message. (No more pesky mic cables!)

Yet wireless systems can be slightly mysterious, prompting suspicion among some users - particularly if they’ve experienced problems for unclear reasons.

The easiest way to understand wireless systems is to think of them as small-scale radio and TV broadcast stations – a transmitter sends out a signal that is picked up by a receiver.

For a number of reasons, including size, weight, battery life and government regulations, wireless systems operate at quite low power and thus have limited range.

The wireless microphone (or bodypack) is the transmitter, complete with a mic capsule, some audio circuitry, and an antenna (usually built into the case). It sends radio signals to its companion wireless receiver, which also has an antenna and some circuitry to select and process the signal, which is then sent via a cable to the sound system.

The transmitter and receiver of each wireless system must share the same frequency. Any other wireless systems in use in the same area must have their own frequencies as well. Ugly noise is produced if two wireless systems are using the same frequency in the same area.

The same goes for other transmitters, especially those of TV stations.

And because these transmitters send out very powerful signals, they are a common cause of interference for wireless systems.

Even though a wireless system needs a clear frequency for the area where it’s going to be used, every frequency is used again and again across the nation.

Again, this is because the power of the output signal of wireless systems is very low.

Keep in mind, however, that there is no absolute guarantee that a clear frequency in one area will be clear elsewhere, even just across town.

This is an aspect about wireless systems that sometimes puzzles users; the government takes care of the problem for the high-power signals of commercial broadcasting, but wireless system users are responsible for avoiding this problem on their own.

Fortunately, most modern wireless systems (developed in the past 15 years or so) offer some degree of frequency agility (also called frequency synthesis). This means that the user is able to select an operating frequency from a number of possible choices, ranging from as few as four frequencies to 1,400 or more, depending upon the model.

The more frequencies offered by a wireless system, the better the chance of finding a clear frequency that is not being used by someone else in the area. Further, in larger cities, where there are more frequencies occupied by numerous users, the ability to choose from a larger number of frequencies is especially important.

Having plenty of open frequencies also helps wireless system users get around another potential problem: intermodulation (or intermod for short). This can occur where the frequencies of two transmitters (of any type) “combine” in a wireless system receiver, resulting in noise and interference.

Most often, intermod is caused by a combination of the frequencies from two TV transmitters, or by the frequency of a TV transmitter combined with the frequency of a wireless system transmitter.

Because the source of intermod is usually not under the control of the wireless user, there is usually little choice except to change the frequency of the wireless system. This is yet another reason for choosing a wireless system outfitted with a wide range of frequency selections.

By law in the U.S., wireless systems are supposed to operate only on TV channels not in local use. If a wireless system happens to cause interference to TV viewers in the area of its use (and this can happen even with their lower output level), the interference is likely to be reported, resulting in the user drawing unwanted attention from law enforcement.

Thus it’s vital for the wireless system user to keep handy a list of local TV frequencies in use (available online at www.antennaweb.org/aw/Address.aspx), and to avoid those frequencies.

Although many wireless systems can “automatically” select frequencies or scan to see local RF activity, it is still possible to select the frequency of a local TV channel and get the innocent user into trouble.

Wireless systems are available for “VHF” and “UHF” frequency ranges (also called bands), roughly corresponding to VHF TV channels 7 though 13 and the UHF TV channels 14 through 69.

The question as to which range is “best” has pretty much been settled by the wireless manufacturers, who generally only offer systems with numerous frequency choices in the UHF band.

Additional bands used by wireless microphones include the “944 MHz” band between 944 - 952 Mhz. This is a band reserved for use exclusively for broadcasters.

Also, the “ISM” band between 902 - 928 MHz is an unlicensed band used by several wireless microphone products. Finally, the 2.4GHz band is another unlicensed area used by wireless manufacturers.

Although the UHF TV band classically extended up to channel 69, channels 52 to 69 (698 MHz to 806 MHz) has been converted to non-TV use - divided up by the U.S. government/FCC and auctioned to various companies for wireless devices available on the consumer market.

Accordingly, it is now against the law to use wireless microphone systems in this band. Even though a system has operated in this range without problems for years, it is illegal.

With all these competing signals in the air throughout the VHF and UHF bands, even high-quality wireless systems can run into problems when operating at distances of 100 feet or less between the transmitters and receivers.

Range problems usually appear as “fizzing” or “swishing” noises, perhaps followed by the complete loss of the audio signal. (This is called dropout.)

In addition to the low transmitter power, two other problems can limit the range of wireless systems. The first is signal absorption due to building construction and internal equipment, or shielding by metallic objects such as electrical wiring, air conditioning ducts, storage cabinets and the like between the transmitter and the receiver.

Note the dual antennas on this wireless receiver, indicating it uses diversity.

The term “line of sight” is often used to express the idea that the signal path from the transmitter to the receiver should be open and clear of obstructions.

This simply means that if the wireless user can physically observe the receiver antenna, RF signal absorption is likely to be low.

The second problem is called multipath. It’s a phenomenon that results in numerous small areas where little or no wireless signal is present because of reflections and the resulting phase cancellations, and it often tends to occur within a fairly short distance between transmitter and receiver.

To overcome the problem, a majority of modern wireless receivers now use a technique called diversity. With diversity, two slightly separated receiver antennas are used, making it very unlikely that both will simultaneously be in one of the low signal (multipath) areas.

The receiver automatically selects the antenna with the strongest signal, not only solving multipath, but also increasing the reliable range of a wireless system.

A final note: most users are surprised to learn - despite urban myths to the contrary – that the U.S. government requires wireless systems to be properly licensed prior to use.

Unfortunately, the agency in change of issuing these licenses (Federal Communications Commission, or FCC) makes it very difficult for conscientious users to actually comply.

As a result, the vast majority of users don’t go to the trouble. But keep in mind that unlicensed wireless systems are in technical violation of FCC rules, and therefore are theoretically subject to fines.

As a practical matter, the FCC has neither the resources nor the inclination to go after the “average” wireless user, so the risk is low. But not zero. Due to the recent changes in spectrum allocation, this issue is being re-visited.

It appears that the FCC may make it easier for typical wireless microphone users such as churches, theaters, musicians, etc. to register their products.

This would also be beneficial in the event that additional types of consumer devices appear and complete for the same spectrum we are currently using.

Part 2: PSW Wireless Primer

Avoiding Wireless System “Issues”
Although the popularity of wireless microphones continues to grow, there’s no denying that they present more opportunities for problems than their wired counterparts.

In addition to the normal acoustic concerns that come with any mic are the complications of RF (radio frequency) transmission, interference, frequency selection, batteries and several other issues.

And technical improvements in wireless systems have not entirely kept pace with increasing frequency congestion, digital television and other recent complications.

Still, the hundreds of thousands of wireless systems employed in the U.S. is compelling evidence that the majority of users will live with the added challenges. Besides, many of the problems encountered by wireless users are largely avoidable, and happen primarily due to oversights, mistakes and misunderstandings.

Addressing the following common issues greatly improves the reliability of wireless systems and goes a long way toward ensuring trouble-free operation.

Issue: Frequency planning and coordination. Wireless systems share the RF spectrum with TV stations and several other types of authorized users. As a result, interference is very likely unless appropriate precautions are taken.

Solution: The first step is to determine the TV channels that broadcast over the air in your area.

When the local TV channels are known, they can be compared to the frequencies of the wireless systems. If there’s a conflict, the wireless frequencies must be changed. This is relatively simple for synthesized systems as well as ones that search for vacant frequencies, but the solution is more difficult for fixed-frequency wireless.

Despite the inconvenience, wireless systems should not be used on occupied TV channels. Not only is interference almost certain, the practice is illegal.

Issue: Intermodulation. Wireless systems can also experience severe interference even when operating on “vacant” frequencies. This is created by intermodulation distortion - basically two strong signals on other frequencies combining in the wireless receiver to create an interfering signal.

In one variation of intermod shown here, the frequencies of two wireless systems can combine to “gang up” on a third system.

Called “intermod” for short, generally this type of interference is more common than direct on-frequency interference from other transmitters.

Intermod is typically caused by other wireless systems, or by other wireless in conjunction with local TV signals.

Even single systems can be affected, but the probability of problems grows roughly proportionally to the square of the number of systems in simultaneous use, plus the number of active analog TV channels present.

By the time eight or more wireless systems and six or more TV channels are involved, it can become quite challenging to find usable frequencies.

Solution: One or more wireless frequencies will have to change. There is generally no other practical solution.

Again, synthesized systems and “auto-search” frequency finding can be very helpful.

However, any frequency can potentially interact with any other, so changing one frequency can solve one problem can create another - or several others.

When changing frequencies or searching, it’s absolutely critical that all RF systems of any type at the location be turned on and operating.

As one clear wireless frequency is found, that system must be left on, and the next system tested until all are operational. Otherwise, the situation can quickly become a snarl of changes and more changes, “phantom” problems, confusion and frustration.

Some manufacturers offer assistance in selecting usable frequencies, and as always, don’t hesitate to get your sound contractor involved.

In addition, there are a number of readily available software packages that are designed to aid in calculating your frequencies so that intermod problems are avoided.

Several manufacturers of wireless microphones offer this kind of software, and there are third-party options as well. Often, the third-party solutions are the most flexible – offering coordination of many types of systems by most manufacturers.

Issue: Shielding or covering antennas. In order to properly launch a radio wave, a sizeable volume of free space is required around an antenna, and in general, they must be unobstructed.

Solution: For efficient operation, all wireless system antennas must be kept clear of metallic objects that can weaken and distort signals in addition to reducing range. With bodypack transmitters, the antenna must be kept away from the mic cable, the bodypack case and ideally, the wearer’s body.

Securing antennas to the transmitter case and tying antennas to cables, as is sometimes done, can be absolutely deadly to range. Skin and flesh can absorb RF energy, so it is best to have the transmitter case and antenna away from the body.

Further, receiver antennas must extend away for the receiver case, as well as away from other antennas, equipment racks, other equipment, cabling and, again, metallic objects.

Large metal structures like ductwork can create serious multipath issues.

It’s best to mount receivers at the top of the rack so that the antennas extend above and away from the rack and other equipment. Using rear-mounted antennas inside a metal rack will almost always result in very poor reception.

For multiple receiver installations, the common practice of positioning front-mounted antennas in a “V” configuration, with all the antennas parallel, will also reduce range. It causes them to function together somewhat like a TV antenna that’s pointed upwards.

Even worse is when antennas from two different receivers touch. Not only will range be seriously compromised, interference becomes much more likely. In such a situation, it is much better to incorporate a single pair of antennas and then an antenna splitter to distribute the signals to the receivers in the rack.

Issue: RF path. A clear path between the receiver and the transmitter is also required. This is sometimes called a “clear line-of-sight,” but remember, light will pass in a straight line through a small hole while radio waves will not.

Solution: Similar to the free space needed around an antenna, radio waves require a sizeable space in which to travel.

The amount of space necessary depends upon frequency - the lower the frequency, the more space needed.

Create an imaginary tunnel of open air between the transmitter and the receiver antennas.

For UHF systems, a tunnel diameter of 3 feet or so is usually adequate, but for VHF systems, it should be at least twice as large. There also should be no metallic objects - scaffolding, iron beams, cables, cabinets, pipes, etc. - within this space.

In particular, large flat metal objects such large ducts, rows of cabinets, truck bodies and the like that are parallel to the path should also be avoided.

Even though they might not be in the direct path, they can still act similar to a mirror, reflecting RF energy away from the direct path. Systems with diversity reception help avoid dropouts in these situations, but range still can be reduced considerably.

Issue: Long antenna cables. Sometimes it’s necessary or desirable to locate antennas at a farther distance from a receiver. RF coaxial cables can be used to connect the remote antennas to the receiver inputs.

However, they typically have considerable losses that will reduce operating range. The amount of loss depends upon the size, construction and quality of the cable, and upon the operating frequency.).

Even high-quality RG-58 cable will have a loss of about 8 dB per 100 feet at 200 MHz, and about 17 dB at 700 MHz. Since every 6 dB of loss cuts range by half, the working range with 100 feet of this cable will be only 40 percent of normal at 200 MHz, and a mere 14 percent of normal at 700 MHz.

Premium RG-58 type cables, such as Belden 7806R, are better, offering about 4.7 dB loss at 200 MHz and 8.9 dB at 700 MHz. Still, at 700 MHz, only 68 feet of this cable will cut range in half.

Solution: If long cable runs are s necessary for your wireless systems to work properly, skimping on the cost of the highest quality cables available is a bad decision. For the best results, a premium foam-dielectric cable such as Belden 9913 should be used. This cable has only 1.8 dB of loss per 100 feet at 200 MHz, and 3.6 dB at 700 MHz.

Generally, it’s preferable to run audio cables out to remote receivers, keeping RF cables short. This is particularly true with runs longer than 75 feet or so. If remote location of the receivers is not feasible, go with the high-quality, low-loss cable noted above.

In-line RF amplifiers can also be used to boost the signal before the long cable run. These devices require power, and add cost. So before thinking that RF amps are the way to go, consider how the system can be configured to avoid using them and still keep your cable loss to a minimum.

Issue: Batteries. Simple but true and most certainly the number-one cause of wireless problems the world over!

Fortunately, it’s the one that’s easiest to fix.

The most common cause of short battery life is poor quality or old age, along with mixing used batteries with new ones and simply losing track of how long a battery has been in use.

Some sound operators also fail to understand that, when turned on, wireless transmitters draw power even if not being used, and that the “mute” switch does not affect the current drain.

Solution: Check transmitter batteries prior to every use. Get a battery tester to help you determine a good battery from a bad one. And when in doubt, change to a new battery!

Name-brand alkaline batteries such as Duracell and Eveready are the best bet. While private label batteries are often nearly as good, their useful life can vary considerably from purchase to purchase.

Make sure that to buy batteries that are date coded, and don’t accept any whose expiration date is less than three years away. And never use zinc carbon or toy batteries; most can’t even properly power up a modern wireless transmitter.

Classically, many techs recommend against use of rechargeable batteries, and for good reason. Rechargeable batteries used to have much lower capacity than alkalines, and the useful life was usually short. This was particularly true of 9-volt units, whose operating life was a fraction of that of an alkaline.

In the past five years, the technology for rechargeable batteries has improved dramatically. Now, NimH and LiPoly batteries are every bit as good as alkalines, and in some cases even better.

Still, it is important to recognize the added complexity of using rechargeable batteries – a clear strategy will be needed for keeping them charged, tested, and removed from the pool when the time comes. By doing this, you can save considerable costs and it’s also better for the environment.

Even more issues that are relatively simple to address can impact wireless performance.

Part 3: PSW Wireless Primer

Downsides Of Digital
Issue: Digital interference. Modern digital audio equipment, including processors, equalizers, controllers and other gear, operate at high clock frequencies that generate considerable radio frequency (RF) noise. (By the way, this noise is often termed RFI.)

As a result, it’s not at all unusual for such equipment to interfere with wireless systems.

Symptoms include low-level spurious tones, buzzing sounds, hissing and a varying noise floor.

Digital interference can also cause an unexplained loss of range and other problems.

Although FCC rules require that such equipment be tested to meet spurious emission standards, it’s a fact that not all units are indeed tested.

In addition, loose covers and casings, warped metalwork, lax grounding and other mechanical shortcomings can greatly increase spurious RF emissions.

Even properly approved digital equipment, in good working order, may generate enough RFI to affect wireless receivers located nearby.

Digital audio equipment in close proximity to wireless systems can sometimes result in interference.

When wireless interference occurs, one of the first things to do is to temporarily turn off digital devices to see if they are the source of the problem.

Solution: As a general precaution wireless receivers should be located as far as possible from digital gear. Often just moving the equipment a few rack spaces apart is enough to solve a problem.

More severe cases may require separating the wireless power, signal and RF cables from those going to the digital equipment.

Using remote antennas with the wireless systems may also be helpful.

And finally, try tightening up the covers on any offending digital gear and also adding a ground strap to the cabinet or other local ground point.

Issue: Lapel (or lavalier) (microphone sound quality. Lapel mics can cause a number of different problems. A common complaint is thin sound quality, which often occurs when the user has previously used only mics intended primarily for vocal applications.

These mics generally boost low frequencies to make the voice sound warmer and fuller, but the omnidirectional mics normally used with wireless bodypack transmitter systems don’t have this boost and thus can sound noticeably different.

Another cause of “thin audio” from lapel mics is interference. RF energy can “couple” into the mic cable and affect the preamplifier circuitry in the mic capsule. A high percentage of all lapel mics exhibit this problem under at least some circumstances.

If the voice quality and level varies when the mic and cable are moved around in close proximity to the wireless transmitter antenna and body, it is almost certain that RF interference is present.

Solution: In all cases, the manufacturer of the wireless system exhibiting this problem should be first contacted for specific recommendations. However, the problem is often solved with the addition of small RF bypass capacitors to the mic connector. Note that this should only be done by a qualified service professional only.

Issue: Lapel mic feedback. Users new to wireless often complain that a system is defective because feedback occurs where none was present before. Part of the problem is that the lapel mics typically used with wireless are not directional and thus provide little feedback protection.

However, the larger problem is usually that the mobility of wireless allows users to walk into zones more likely to cause feedback.

Solution: Use lapel mics with a unidirectional pattern, or use headset mics. Moving the mic closer to the mouth and lowering gain is also helpful. Many users think headset mics are unsightly, but unidirectional mics can suffer from sudden drops in level when wearers turn their heads.

The better solutions are acoustic, either by training users to avoid feedback zones, or by modifying the loudspeaker configuration to put feedback zones out of reach.

Issue: Lapel mic mechanical problems. This is common to lapel mics, in particular because their cables are small, often delicate and typically get considerable abuse.

Even if not damaged outright (i.e., the cable pulled out of the mic connector), lapel mic cables eventually wear out.

Most often this wear occurs first at the connector end, but keep in mind that it can also happen at the capsule end. Usually the cable shield fails first due to constant bending in the area where a cable leaves the connector’s strain relief.

A headworn mic can be an option in some cases, and there are a wide variety of lapel mics to choose from. (Upper photo couresy of Electro-Voice, showing the company’s RE97 headworn mic.

When this happens, clicks, pops, other noise and “lost audio” are experienced. Even before there’s a complete break in the shield, pops and clicks due to RF disturbances can happen.

Therefore, it’s always prudent to check the cables when experiencing lapel mic noise of any type. Breaks at the connector end can usually be repaired (and don’t forget the bypass capacitors), but a break at the capsule end may not be fixable.

Mechanical noise due to lapel mic capsules rubbing on clothing is relatively common and can usually be eliminated by using the right type of mic clip, one that holds the capsule away from the fabric.

It may also be necessary to carefully secure the cable near the mic capsule. Static electricity sometimes creates audio noise, especially with certain types of fabric. Clothing anti-static spray usually solves this problem.

Issue: System quality. It may seem strange to list “system quality” as a wireless problem, but a great many wireless difficulties start with inferior equipment. Inexpensive systems can often work well in rural areas and/or in relatively undemanding applications.

But in larger cities and their surrounding suburbs plagued by typical frequency congestion and myriad interference sources, something better may be required.

The same is usually true when more than a few systems must be operated at the same site. And, this situation is going to worsen, with more and more digital signal sources going on the air almost daily.

The adoption of digital technology has greatly lowered the price of many audio products, but the impact of these advantages on wireless systems has been relatively small to this point. Wireless systems are still largely analog-based, and their manufacture is more labor intensive due to the requirement of considerable tuning, testing and tweaking.

Quality components also tend to be expensive in comparison to digital components and are less adaptable to low-cost automated assembly.

Unfortunately, there is yet no new magic technology that can cut the cost of a quality wireless system significantly - say 30 to 40 percent. Right now, if cost goes down, so do quality and performance. And it’s easier and cheaper for manufacturers to promote their mic capsules and “features” rather than build in better performance.

Consequently there is a growing tendency to regard the RF portion of a wireless system as being relatively unimportant. This is a serious mistake.

Solution: If a wireless system doesn’t have the selectivity and interference rejection to cut through all of the “junk” in the air, it doesn’t matter which mic elements it has, how neat the feature set, or how much money was “saved”. You’re simply left with something that doesn’t work like it should.

The recommendation is to pay a little more and go for performance over features. High-quality wireless systems cost less than half of what they did 10 years ago, and they work better in virtually all cases.

Final Thoughts: All in all, wireless microphone and in-ear monitoring systems can significantly enhance the experience for audiences and performers alike. Freedom of movement for actors, musicians, minsters, orators and politicians is a major benefit.

However, the complexity, cost and potential problems are the risks of using microphones. By following the guidelines presented in this series of articles, you should be well on the way to flawless operation from wireless systems.

Don’t forget that this is a changing world with respect to the RF spectrum and thus the operation of wireless mic systems. What works today may not work tomorrow.

Your best bet is to stay informed and educated. Watch for announcements about RF issues related to the FCC and potential other users of the spectrum. Keep up with the technology as manufacturers introduce new systems.

And most of all, stay up on troubleshooting skills so you can identify where the problems originate. Sometimes the wireless will be at fault, and sometimes not. It’s best to know the difference.

How to Avoid the Seven Characteristics of an Amateur Mix

2011-05-02T08:45:00.000-07:00

Bobby Owsinski, author of the The Mixing Engineer's Handbook, Second Edition, identified the seven characteristics of an Amateur mix in this article. Now that those seven have been identified, let's look at what we can do to avoid them.

1. Inconsistent Levels

"Instrument levels that vary from balanced to too soft or too loud or lyrics that can’t be distinguished. Once again, a newbie mixer usually sets the faders and forgets them, but mixing is just as dynamic as the music. Every note of every solo and every word of the vocal must be heard. Even with automation as sophisticated as it is these days, it still takes some time and a critical ear to be sure that everything is heard."

Of all the mixes I've ever heard, this is the one that drives me crazy.

The FGM-148 Javelin is a shoulder-mounted fire-and-forget missile system with "lock-on before launch" and automatic self-guidance. Pick your target, pull the trigger, and the missile does all the work. This is not how church audio mixing is to be done. There is no setting and forgetting. How do you avoid this? Easy - see points two through seven.

2. No Contrast

"The same musical textures are used throughout the entire song. This is generally an arrangement issue, which the mixer can affect somewhat since mixing is so much more than balancing. It’s influencing the arrangement by what you mute, emphasize or lower in the mix."

Adding contrast means you need to be an active mixer. For example, during the bridge of a song, add some high EQ boost to the cymbals to make them pop out in the mix. Another example is bringing the backup singers closer to the front of the mix in the chorus. This could be done with volume and effects changes.

3. Frequent Lack of a Focal Point

"There are holes between lyrics where nothing is brought forward in the mix to hold the listener’s attention. Granted, this is an arrangement issue too, but it’s your job as a mixer to find some point of interest and emphasize it."

Bring up the volume of the lead guitarist between verses or definitely in any instrumental passage. Think of a song as a continuous story. You don't want the story to stop at any time, so keep it going by bringing in instruments to take over when the singing stops for a significant period of time.

4. Mixes That Are Noisy

"Clicks, hums, extraneous noises, count-offs, and sometimes lip-smacks and breaths are all things that the listener finds distracting. It may be a pain to eliminate these distractions but you’ve got to do it to take the mix to where it has to be."

This is where you really have to know how the band performs a song. The audience doesn't need to hear a singer count off the beginning of the song to the rest of the band. Ride the fader at the beginning of the song so after the count, you can bring up the singers volume.
As far as hums and some of the other points mentioned, it's a matter of proper equipment usage and setup. Track down the hums during practice. Set the proper gain structure, and use close-mic'ing techniques. Your goal is a crisp clear sound.

5. Mixes That Lack Clarity And Punch

"Instruments aren’t distinct, and low-end frequencies are either too weak or too big. This is really the number one indication of an amateur mix, especially in the low end. It’s either way too heavy or way too light. The way around this is to listen to other records that you think sound great and try to emulate the sound. Sure it takes time, but it will get you in the ballpark."

Bleech, yuck, this is terrible...oh sorry, I just took a drink of unexpectedly cold coffee.

Bobby said it quite well, "listen to other records that you think sound great and try to emulate the sound." Separate out instruments/vocals in your mix by bringing out their natural frequencies. Blend them together so they complement each other but remain distinct in the mix. It's like the classic movie line from Jerry Maguire, "you complete me." Each instrument is its own but when they are brought together, the overall sound benefits.

6. Mixes That Sound Distant and Are Devoid Of Any Feeling of Intimacy

"The mix sounds distant because too much reverb or overuse of other effects. This is another common trait since a newbie mixer thinks the plug-in effects are so cool (because they are!) that they want to use them all on everything all the way through the song. You’d be surprised just how many effects are used in a great mix sometimes, but the results are so subtle that you can’t really tell unless you had the original non-effected sound to compare with. In an amateur mix, you hear them all screaming at you all the time. If you can make it sound great without effects first, you’ll automatically moderate their use."

I recall a sound engineer talking about his early days in the biz. He said he mixed with all the fancy effects during a rehearsal and then the lead FOH guy stepped in, turned off 99% of the effects and set a mix with just the EQ. The mix was so much better than what he had done.
Antoine de Saint-Exupery said, "Perfection is achieved, not when there is nothing more to add, but when there is nothing left to take away." When it comes to mixing, this is a great quote to keep in mind.

7. Dull and Uninteresting Sounds

"Generic, dated or frequently-heard sounds are used. There’s a difference between using something because it’s hip and new and using it because everyone else is using it."

There are a few ways you can get dull and uninteresting sounds; don't touch the eq/effects, use what you've always used, and use what everyone else is using. You can avoid this by following a few simple rules;

Use the EQ and effects, they are there for a reason. Odds are if you are reading this then you're already using these.
Change up your sound. A modified mix can be like a new color of paint in the same room. Listen to your music collections, go to live concerts, listen to how other people are mixing and the sound they are getting. Then during practice, see what you can do. I suggest picking one instrument to focus on at a time (per service).
Rather in contrast to "b," don't use what everyone else is using. This deals more with effects. If the latest popular effect is a distorted acoustic guitar sound, don't think you have to copy it. Likely, the first person to use it found it perfect for a particular song. Then others copy it because it sounds cool. Now, it's no longer a cutting-edge sound and everyone is doing it.

By Chris Huff, Behind the Mixer, used by permission.

image source = http://en.wikipedia.org/wiki/File:Army-fgm148.jpg

Advice For New Technical Artists

2011-03-18T12:19:00.000-07:00

The life of a church tech is crazy. You’re the first to arrive and last to leave, get few days off and for less money than your secular counterparts. Despite that, I believe tech ministry is one of the most amazing ministries you can serve in. I’ve recently been asked for advice on starting a career as a church tech. Those who’ve asked have had varying skills, personalities, specialties and areas needing improvement, but all of them got the same advice from me.

First, church techs must become proficient in multiple, if not all of the tech disciplines of audio, video and lighting. Every tech has specialties and some are blessed with multiple specialties. Most churches however only have the budget to hire one tech and that person has to lead them all. Even in churches that can afford multiple, more specialized techs, being well versed in all disciplines makes you more effective, more valuable and better equipped to handle possible issues that could come your team’s way.

Second, be open to learning from those more experienced or knowledgeable. Many young artists struggle with being teachable. There are some seasoned artists who struggle with this too. Often we get a little bit of knowledge and we think we know it all. I’ve certainly had prideful moments, but when I’ve taken the opportunity to learn from those who know more than me, I benefit greatly and so does everyone around me. The best techs I’ve met have this trait. The other day I spoke with a well respected and seasoned sound guy who was experimenting with a new technique he learned from someone else. There is always something more or new to learn in the tech field, the trick is to stay open to learning it.

Last, create boundaries that will guard the hearts of you and your family. This may ruffle feathers, but it’s easy for ministry to overtake your life, mess with your family and kill your zeal for serving. One of the hardest things for me to learn was that I had to create boundaries to protect myself and family. For every church that has amazing leaders who are protective of their people, there are more that are just trying to get by and ask too much of their staff. Churches don’t burn people out on purpose, but ultimately it’s not the church’s responsibility to protect you and your family. A church’s top priority must be the whole ministry before each person. Your priority must first be you and your family and then your ministry.

Learn every discipline you can, take advantage of opportunities to learn more, and have healthy boundaries. For nearly 15 years now I’ve loved both serving in and leading Technical Arts ministries. I believe it’s a very noble calling, one that is increasingly critical in the church today. If you’ve been called to a ministry in Technical Arts, I believe and have experienced how these three things will help you be happy and successful as you serve your church and community.

Reposted with permission

Duke DeJong has been involved in live production for over 15 years and spent 10+ years in full time ministry, and in 2011 began serving as the Church Relations Director for CCI Solutions. You can find him online at www.dukedejong.com or on Twitter @dukedejong.

Common Acoustical Problems for Houses of Worship

2011-03-02T07:35:00.000-08:00

Acoustics is arguably the single most important consideration for new and existing houses of worship. In order to effectively communicate the message through spoken word or music, a good acoustical environment is critical. Poor speech intelligibility or inadequate music reproduction can make it difficult for members of your congregation to receive and understand the message as it was intended to be heard.

Acoustics is a comfort factor. The acoustics of your space should be as important as, or even more important than, the lighting, heating, cooling, seat cushions, or any other design factor that affects comfort.

Oftentimes, the sound system is blamed for a poor sounding sanctuary. One might think that things will be improved by purchasing a bigger and/or newer sound system if they are not happy with the sound quality of their room. This is rarely the case. The equipment in your sanctuary will only be as good as the acoustic environment in which it is placed. You must address the acoustical properties of the room itself in order to solve the root of the problem.

There are some common acoustical problems that many houses of worship face. It is important to be aware of these problems and to understand how they can be avoided or remedied in your sanctuary.

Sound Isolation and Noise
Isolation can be defined as keeping outside sounds out and inside sounds in. Sound from the outside – such as from nearby roads, airports or train tracks – must be kept outside. If the house of worship is located in a residential neighborhood, you might also wish to keep your noise – from the service itself or from outdoor building equipment – from bothering the community.

Sources of noise include air conditioning, fans, lighting, highway traffic, air traffic, trains and footfall noise. Sound travels like water. Any little leak in the system and sound will find a way to escape. Typically, a construction approach is needed in order to add mass to the existing structure to isolate it.

Noise is best controlled by using sound transmission (sometimes called “soundproofing”) products during the construction phase. There are fairly common building materials that can isolate the room without adding a lot to the cost of construction. For example, simply doubling up on drywall on both sides of a wall or insulating interior walls can help with sound transmission issues. Sound isolation can be a tricky subject and acoustical professionals should be involved to determine the best course of action.

Reflected Sound
It is important to make sure that you strike an acoustical balance between the various applications used to communicate the message. For example, if a room is too reverberant, reflected sound will interfere with the spoken word – even though music might sound wonderful. If a room is not reverberant enough, musical factors such as “ensemble” of the performers and “envelopment” of the participants may suffer – even though speech is very well understood.

Controlling reflected sound is the key to making a worship setting sound good. Absorption of sound waves can be accomplished through common room features like curtains, carpet and even the people in the pews. Diffusion is provided by anything that breaks up a parallel surface and directs sound waves in different directions. The placement of wall décor like window trim and statuettes are sometimes enough to provide adequate diffusion, depending on the style of worship. While these common room features help control acoustic anomalies, they are typically not enough and acoustical panels are usually needed to properly treat a house of worship.

Absorbers make a room more “dead” acoustically by absorbing and trapping sound energy in high-density fiberglass or other fibers. Absorption is an important element of speech intelligibility because it reduces the reflected energy that can mask spoken word.

Redirectors interrupt parallel surfaces to prevent reflection. Redirection is often helpful in the choir area to allow the singers to hear themselves and each other. Redirectors range from angled sheets of painted plywood mounted to the wall to the popular convex half-barrels and pyramidal prefabricated panels.

Diffusors accomplish accurate redirection of sound energy with complex surfaces. These highly engineered surfaces evenly distribute sound and create a sense of warmth or envelopment to the room. Diffusion can also make a space sound larger and provide the same degree of high-quality sound to every person in the sanctuary. This contributes to eliminating hot spots or nulls.

Specifying the quantity, size and placement of these materials is a science and acoustical professionals should be contacted in order to obtain the best treatment plan for your space.

Stage Resonance and Stage Volume
If your church has a live band, you may have heard this one before. A typical scenario is when the guitar player says that he cannot hear himself, so he turns his volume up. This is followed by the bass player turning up, the drummer playing louder, and so on. The result is drastically increased stage volume that can compete and interfere with the main sound system. Also, the audience suffers because the sound from the stage is excessive and can be annoying to some members of the congregation.

A large number of stages are built as hollow cavities. These cavities have a tendency to resonate (similar to a drum) and produce an audible tone. Hollow stages can cause a “muddy” sound where low frequency energy is dominating the room. This lingering low frequency energy can have a negative impact on speech intelligibility and cause music to seem unbalanced and unpleasant to listen to. Also, the walls and ceiling around the stage area are often neglected when it comes to acoustical treatment and are highly reflective surfaces.

Ways to improve this situation is to fill in the hollow cavity below the stage with standard insulation. Attacking the vibration at the source is also a great option. There are commercially available products that will “decouple” (similar to the shocks in your car) the vibrations caused by guitar amplifiers, subwoofers, and stage monitors that are directly connected to the stage.

Finally, selecting the proper amount and placement of absorption and diffusion panels for the stage area will also bring down the stage volume, resulting in a more pleasing acoustical environment.

Understanding these common acoustical problems will help you to make decisions for your church that will dramatically improve the experience for your congregation and allow you to effectively deliver the message.

Gavin Haverstick is the owner of Haverstick Designs, a full-service acoustical consulting firm specializing in architectural acoustics. From Religious Product News. Used by permission.

The Basics Of Lavalier & Headworn Microphones

2011-02-24T13:39:00.000-08:00

Two widely used microphone types, and tips on how to use them with maximum effectiveness

Lavalier Microphones
The desired sound source for a lavalier microphone is a speaking (or occasionally singing) voice.

Undesired sources include other speaking voices, clothing or movement noise, ambient sound, and loudspeakers.

Balanced low-impedance output is preferred as usual. Adequate sensitivity can be achieved by both dynamic and condenser types, due to the relatively close placement of the microphone.

However, a condenser is generally preferred. The physical design is optimized for body-worn use. This may be done by means of a clip, a pin, or a neck cord. Small size is very desirable.

For a condenser, the necessary electronics are often housed in a separate small pack, also capable of being worn or placed in a pocket. Some condensers incorporate the electronics directly into the microphone connector.

Provision must also be made for attaching or routing the cable to allow mobility for the user.

Placement of lavalier microphones should be as close to the mouth as is practical, usually just below the neckline on a lapel, a tie, or a lanyard, or at the neckline in the case of robes or other vestments.

Omnidirectional types may be oriented in any convenient way, but a unidirectional type must be aimed in the direction of the mouth.

Avoid placing the microphone underneath layers of clothing or in a location where clothing or other objects may touch or rub against it. This is especially critical with unidirectional types.

Locate and attach the cable to minimize pull on the microphone and to allow walking without stepping or tripping on it. A wireless lavalier system eliminates this problem and provides complete freedom of movement.

Again, use only high-quality cables and connectors, and provide phantom power if required.

A condenser lavalier microphone will give excellent performance in a very small package, though a dynamic may be used if phantom power is not available or if the size is not critical.

Lavalier microphones have a specially shaped frequency response to compensate for off-axis placement (loss of high frequencies), and sometimes for chest “resonance” (boost of middle frequencies).

The most common polar pattern is omnidirectional, though unidirectional types may be used to control excessive ambient noise or severe feedback problems.

However, unidirectional types have inherently greater sensitivity to breath and handling noise. In particular, the consonants “d”, “t”, and “k” create strong downward breath blasts that can result in severe “popping” of unidirectional lavalier microphones.

Placing the microphone slightly off to the side (but still aimed up at the mouth) can greatly reduce this effect.

Good techniques for lavalier microphone usage include:
• Do observe proper placement and orientation.
• Do use pop filter if needed, especially with unidirectional.
• Don’t breathe on or touch microphone or cable.
• Don’t turn head away from microphone.
• Do mute lavalier when using lectern or altar microphone.
• Do speak in a clear and distinct voice.

Headworn Microphones
Again, the desired sound source for a headworn microphone is a speaking or singing voice.

Undesired sources include other voices, instruments, ambient sound and sound system loudspeakers.

Most headworn microphones are of the condenser type because of their small size and superior sound quality. A dynamic type can be used for speech-only applications or if larger size is not an issue.

For either type, the frequency response is shaped for closeup vocal with some presence rise.

An omnidirectional polar pattern is suitable for most applications, especially if the microphone does not reach all the way in front of the mouth.

A unidirectional pickup is preferred in very high ambient noise applications or to control feedback from high volume monitor loudspeakers.

For proper operation, unidirectional types should be positioned in front of or directly at the side of the mouth and aimed at the mouth. A windscreen is a necessity for a unidirectional headworn microphone.

Headworn designs put the mic element very close to the voice.

Balanced low-impedance output is preferred for hardwired setups but headworn types are often used in wireless applications. In that case, the impedance and wiring are made suitable for the wireless system.

For condenser types, the bodypack transmitter provides the necessary bias voltage for the microphone element.

There are many different headworn mounting designs. Most have a headband or wireframe that goes behind the head, while a few are small enough that they merely clip over the ear.

In all cases, the microphone element is at the end of a miniature “boom” or flexible arm that allows positioning close to the mouth. Again, an omnidirectional element can be positioned slightly behind or at the side of the mouth while the unidirectional type should be at the side or in front and aimed toward the mouth.

The main advantages of the headworn microphone over the lavalier are greatly improved gain before feedback and a more consistent sound level.

The increase in gain before feedback can be as much as 15-20 dB. This is completely due to the much shorter microphone-to-mouth distance compared to lavalier placement. The headworn can nearly rival a handheld type in this regard.

In addition, the sound level is more consistent than with the lavalier because the headworn microphone is always at the same distance to the mouth no matter which way the user may turn his head.

Good techniques for headworn microphone usage include:
• Do observe proper placement and orientation.
• Do adjust for secure and comfortable fit.
• Don’t allow microphone element to touch face.
• Do use pop filter as needed, especially for unidirectional.
• Do adjust vocal “dynamics” to compensate for fixed mouth-to-microphone distance.

Layering the Mix

2011-02-16T14:25:00.000-08:00

By Jonathan Malm

Editor's Note: This is an introduction that Jonathan wrote for his church's audio engineering team. It's a basic introduction, but in those basics, it's going to be helpful, particularly to volunteer sound guys or sound guys at smaller churches. Thanks, Jonathan, for the good material, and for permission to use it!

This is one of the most important parts of running sound. A good mix has basically three sound level layers. The top is the loudest, and the bottom is the softest.

• Top:
(1) Lead vocals,
(2) Instrument solos,
(3) Featured loop elements

• Middle:
(1) Drums,
(2) Electric guitar,
(3) Bass,
(4) BG Vocals,
(5) Loops

• Bottom:
(1) Acoustic guitar,
(2) Pad/Organ

The Theory of Layering
When you listen to a CD that represents our musical style, the main instruments you hear are drums, electric guitar, and bass. If you listen closely you will hear acoustic guitar and pad/organ...but those are mainly to fill in the gaps. You wonʼt really hear them that well...only when everything else has cut out. Then the main vocal or any other instrument solo should be clearly heard and understood above the rest of the layers.

Guidelines to Layering

• Every few minutes you should evaluate your layering.
• Is the top layer clear and distinguishable?
• Are the middle layer elements properly balanced.
• Are the bottom elements barely audible?
• Then adjust as necessary.

Volume of the Spoken Word

15 dB above volume of the room. Always changes

Bass Guitar versus Kick Drum

When youʼre running sound youʼll want that sweet low end coming through the system. The proper balance between the kick (bass) drum and the bass guitar is critical to this. Typically you want the bass guitar just under the kick drum. The bass guitar provides a critical musical element while the kick drum provides the rhythm that people feel.

Without the bass guitar properly mixed in, the music will feel empty no matter how much kick drum you put into the mix.

Itʼs also important to remember that some times people will say there is “too much bass”...but that does not mean bass guitar. Most people donʼt realize the difference between kick drum and bass, or they canʼt distinguish between the sounds. Typically lowering the kick drum will satisfy bass complainers.

Snare/Kick

Never let the snare or kick get lost in the mix. They should be clearly heard. The snare gives people something to clap to and the kick gives them something to tap their foot to.

Myths About Running Sound

Myth 1: Sound is subjective. Each person can run it differently.

To a small degree this is true...but the style of music that the worship leader is producing dictates the way the sound is run more than the sound technician. Rock and roll has a different sound than Gospel music...but the musician determines the sound. A good sound guy makes it sound like the musician wants it to.

Myth 2: A sound guy is supposed to make the band sound good.

The sound guy is supposed to accurately reflect what is happening on the stage. Because different people have different preferences, what sounds good to the sound guy is not necessarily what sounds good for the musicians.

Myth 3: A sound guy should constantly be adjusting the levels.

If youʼre constantly adjusting the levels, you are micro-managing the band. The only time you should constantly adjust levels is when you have musicians or vocalist who are not consistent with their volume.

By Jonathan Malm

Effective Microphone Strategies That Produce Great Results With Church Choirs

2011-02-02T09:07:00.000-08:00

What mics work well for the choir? Where should the mics go, and how many are needed in each situation? Suggestions that point you in the right direction

by Bruce Bartlett

In house-of-worship sound system installations, one of the biggest challenges is miking the choir.

We want to achieve a good balance, a natural sound, and high gain before feedback.

Another goal is to make sure that the microphones are invisible! It’s a tough assignment.

What mics work well for the choir? Where should the mics go, and how many are needed in each situation? The suggestions that follow should point you in the right direction..

Mics for Choir
The most popular type of choir mic is a small hanging mic. A few of these tiny microphones can be hung over the choir from the ceiling, from rafters, or on stands. They’re almost invisible when viewed from the congregation.

Choir mics are condenser types with a cardioid or supercardioid polar pattern. These patterns reject feedback yet have a wide enough pickup for good coverage of the singers.

Condenser mics can be made much smaller than dynamics of equivalent bass response.

An example of a miniature choir mic.

Choir mics are built in three parts: mic head, cable, and power module. The mic head puts out an unbalanced, medium impedance signal which travels through the long cable.

At the far end of the cable is a power module with an XLR connector or a terminal block. The module accepts phantom power and sends DC to an FET near the condenser mic capsule.

Also, the module equalizes the mic signal and converts it to low-Z balanced.

In some choir mics, the power module takes the form of a flat plate that is mounted in the ceiling. In other mics, the module is a tube with an XLR-type connector.

Mic Placement for Sound Reinforcement
When placing mics to pick up the choir, a critical factor is gain before feedback. To get enough gain, you must to mike the choir much closer than you would for recording.

Place the mics about 18 inches in front of the first row of singers, and about 18 inches above the head height of the back row (See Figure 1).

The mics are raised to prevent overly loud pickup of the front row, relative to the back row. The rows are equidistant from the raised mics, giving a well-balanced sound.

Figure 1: Typical choir mic placement

To achieve uniform coverage, use one microphone in the center of every 20-foot span of singers.

A choir of 30 to 45 voices should need only two or three mics. Given a fixed miking distance, you’ll get less feedback with fewer microphones.

You might want to mount the choir mics on tall boom stands to experiment with placement during choir rehearsals. Once this is done, hang the mics permanently.

In miking a choir, it might seem important to consider the 3:1 rule. When multiple mics are mixed to the same channel, the distance between mics should be at least three times the mic-to-source distance.

This prevents phase interference between mics (comb filtering), which is a series of peaks and dips in the frequency response - a colored, hollow sound.

The 3:1 rule cannot be applied to miking a choir with a few mics. Why? Most of the singers are somewhere between the mics, and those singers will be picked up with some phase interference.

However, since each singer is in a different position relative to the mics, each singer is heard with a different coloration. The effect averages out over all the singers and so is not very audible.

Once the mics are placed, you need a way to hold them in position. Mic cables can lose their orientation as the mic cable uncoils over time, or the mics can swing back and forth in a breeze.

Some choir mics have a built-in hanger which comes with a tiny crossbar or pipe.

You thread a fish line through this pipe and attach the line to screw hooks in the side walls (See the example photo of a miniature choir mic above to view how this should look.). The guy wire keeps the mics oriented toward the choir.

If hanging mics is not an option, you might try making some clear plastic mic stands of Lexan corner molding. The stands can be cut to the desired height and mounted to or near the choir rail.

What if the choir is under a balcony? Try mounting some supercardioid boundary mics to the bottom surface of the balcony, near its front edge.

Other Considerations
Monitor loudspeakers can easily feed back into the choir mics. To keep feedback under control, try not to use monitors near the choir.

Instead, turn up the house loudspeakers. If the choir insists on monitor loudspeakers, don’t feed a monitor signal of the choir back to them because it will cause feedback with the choir mics. Rather, just feed them some music for accompaniment.

If the choir members complain they can’t hear themselves, maybe the piano, organ, or music tracks are too loud in the choir monitor speakers.

Have the choir sing a capella, with the air conditioning turned off. Can they hear themselves? Now turn on the air conditioning. Can they still hear?

Turn up the piano or organ in the choir monitors. Then turn up the tape tracks. At what point can the choir no longer hear their voices? Turn down the offending sound source.

In some venues the choir mics pick up too much of the organ’s sound. In this case, use supercardioid mics and aim them toward the middle row of the choir.

Because the mics partly reject sound from the side, they will pick up less of the organ in this configuration. If the organ is still too loud, aim the mics straight down over the choir and filter out frequencies below 100 Hz in the choir mics.

The Smaller Choir
Suppose you want to mike a small choir of about 12 people.

Of course, you want the most gain-before-feedback possible, but you also want to keep the number of mics to a minimum, both to reduce cost and to simplify mixer operation.

Should you use one close-up mic per person? One mic on every two people, or one every four people? How about one mic on the entire group?

There’s a scientific way to answer these questions. It’s possible to calculate the gain before feedback (GBF) of a chosen mic setup, given these variables:
-- Number of open mics
-- Miking distance
-- Mic polar pattern
-- Angle of sound incidence to the mic

GBF drops 3 dB every time you double the number of open mics. GBF drops 6 dB every time you double the miking distance (not including proximity effect).

Directional mics have more GBF than equivalent omni mics. With directional mics, on-axis sound sources produce more GBF than off-axis sources.

The results of these calculations might surprise you.

Figure 2 below shows several mic techniques and their calculated GBF, from best to worst. Note: The GBF of 12 close up mics was normalized to 0 dB.

Figure 2: A variety of mic techniques
and GBF

What do the results show? If a facility can afford 12 mics, and can handle that many on the mixing console, go for it - 12 close-up mics give the best GBF, by far.

If six mics make more sense, you save cost and complexity, but give up about 14 dB GBF. That’s not necessarily a problem, depending on the venue, mics, and loudspeaker placement.

When using six mics to pick up 12 people, the singers must be relatively far from the mics, and about 45 degrees off axis in relation to them. So six distant mics have much less GBF than 12 close-up mics.

If the customer wants the simplicity of a single mic on the choir, try either one hanging supercardioid mic or one floor-mounted supercardioid mic.

Either should provide about 23 dB less GBF than 12 handheld mics, but this approach could work suitably anyway.

All of this is theoretical, and assumes that the mics have textbook polar patterns. Your real-world results may vary! But these findings might suggest some tendencies helpful to know.

Mics for Recording/Broadcast
For recording and/or broadcasting a choir in a live setting, three types of mics have proven to work well: cardioid condenser, omni condenser, and stereo condenser types.

A stereo pair of cardioid condenser mics is shown in Figure 3. This array reduces pickup of audience noise and room acoustics, and tends to provide sharp imaging.

If mono compatibility is important, use two hypercardioid mics, angled 90 degrees apart, with their grilles almost touching and aligned vertically. This will prevent phase interference between the mics.

A spaced pair of omni condenser mics is a good choice if you want stereo images to be blended rather than pinpointed. Omni condensers have superior low-frequency response, so they also excel at reproducing a pipe organ.

A stereo condenser mic combines two mic capsules in a single housing, for convenience. One stereo mic generally costs more than two mics of comparable quality.

A special type of stereo mic is called “mid-side.” It has one mic capsule aiming straight ahead toward the middle of the choir, and another capsule aiming to the sides. In a mid-side stereo mic, you can adjust the stereo spread by remote control.

Placement for Recording/Broadcast
Mic placement for recording must be determined by ear as the choir rehearses. Experiment with a pair of mics, or a stereo mic, on a tall stand. Once you’ve found a good-sounding mic position, hang the mics there.

Start by placing the mics about 12 feet from the choir. Raise them a few feet above the heads of the back-row singers, and aim them down at the choir.

The setup shown in Figure 3 provides excellent stereo. For convenience, you might want to mount both mics on a stereo bar or stereo mic adapter. This device mounts two microphones on a single stand.

A good starting position for two spaced mics is about 3 feet apart and 12 feet back. The wider the spacing between mics, the greater the stereo spread. But too-wide spacing results in an exaggerated left-right effect.

Figure 3: A near-coincident stereo mic technique.

After setting up the mics, it’s time to fine-tune miking distance. The farther the mics are from the choir, the more room acoustics you’ll hear in the recording.

Listen to the mics’ signals, using either headphones or loudspeakers in a separate room. If the choir sounds too distant and muddy, move the mics about a foot closer and listen again. f the choir sounds too close, without much room sound, move the mics farther away.

Note: if the hall is acoustically “dead” (lacking reverberation), you might prefer to add artificial reverb. Use a digital reverb unit patched into the effects loop of your mixing console.

If the organ overpowers the choir, or if air handling is noisy, you’ll have to close-mike the choir and add digital reverb.

Simultaneous PA and Recording
Ideally, you’d use distant mics for recording and close mics for PA. But suppose you’re limited to just the PA mics. Since PA mics are closer to the choir, their recorded sound will lack ambience unless you add some artificial reverb.

You’ll want to send the reverb only to the recorder, not to the house loudspeakers. Here’s one way to do it:

1. Connect buses 1 & 2 to the PA power amp inputs (or to the graphic EQ inputs). Connect busses 3 & 4 to the recorder line inputs.
2. Assign all mics to buses 1, 2, 3 and 4.
3. Connect the reverb returns to buses 3 & 4 (input).
4. Turn up the choir mics’ effects sends while monitoring the recorder output.

You also could record “dry” to multitrack, then add reverb during mixdown.

Try out these techniques, and feel free to experiment with your own techniques too. In time, you’ll enjoy a beautiful choir sound for both recording and PA applications.

From ProSoundWeb, used by permission. AES and Syn Aud Con member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.

Digital Wireless and The Future of Wireless Microphones

2011-01-19T06:33:00.000-08:00

The last few years have not been kind to the users of wireless microphones. First, the FCC sold off a great deal of our wireless frequencies for a few billion dollars. We had to abandon all of the 700 MHz frequencies.

Then they issued the “Second Memorandum Opinion and Order” regarding the rest of the UHF bandwidth (“White Spaces”). Basically, we have to share those frequencies (with a few exceptions) with a bazillion cell phones in the next year or two.

But they’re not through with us. The next to go will reportedly be either the 500 MHz or the 600 MHz bandwidth, though not for “a few years.” And who knows what’s coming after that?

There are a number of possible strategies for responding to these threats.

1) Buy disposable wireless. Buy competent electronics, but at the bottom end of the price range. The danger, of course, is that it’s easy to not just get “inexpensive” wireless, but to end up with “cheap” wireless, and there’s a world of difference. In the youth room, it’s not so bad; in the main room, it’s a real problem.

2) Buy wireless with a very high number of usable frequencies and very tight control of those frequencies: there has got to be at least one clear frequency when all the dust settles! This has the added advantage of allowing you to actually trust your wireless, which makes it all that much more useful! But these can be fairly costly.

3) Go without wireless. Let’s face it: the back line is sometimes better served by wired mics, but the “money channels” are probably never going to tolerate a wire tethering them into place.

4) Predict what the FCC is going to do next. Currently, they have announced a few clear frequencies, but they’re not the same in every region, so… Good luck with that prediction! Let me know how it works out for you, OK? (Update: Shure seems to be doing this, and in a way that looks very promising!)

5) Develop a technology that will survive whatever the FCC throws at us. Do something that completely steps outside the way we’ve always done wireless mics, and trust that whatever the FCC does won’t affect it too much.

This last option is the thinking behind a new series of microphones from Line6. Yes, I said Line6: the guitar effects and amplifier people. It sounds strange to me too. They’ve introduced a line of wireless microphones that do exactly that: they do wireless differently. Here’s what’s they do:

First, they use a frequency that is under the FCC’s radar: 2.4 GHz. I was concerned that they’d pick up interference from cell phones or other high-powered things, but those are on other nearby frequencies. It’s true, WiFi (a very low-powered technology) operates in that range, but they avoid those problems with other tools. And yes, it’s license-free (and that’s growing in importance). A number of companies have tried to use cell-phone bands (I remember some in the 1.9 and 2.1 GHz range), but none of those have succeeded in avoiding interference. This 2.4 bandwidth appears (key word there) to be a safe place for wireless to live, if you can handle WiFi.

So, to handle other RF signals (like WiFi), Line6 uses something that is not spread-spectrum technology, but acts kind of like it. I don’t completely understand how they handle the frequencies, but I really appreciate how you can set up a dozen wireless in one room, and not bother with any frequency settings, but all twelve will work really well.

They use honest-to-gosh digital wireless technology for their RF section: “Fourth generation digital wireless technology,” they call it. Some of the benefits of digital include amazingly simple setup, excellent audio quality.

Because of the very high bandwidth (2.4 GHz is a very small; 700 MHz, by comparison, is much larger wavelength!), the Line6 RF signal doesn’t travel through walls. So I can have twelve frequencies in this room and another twelve in the room right next door. Try that with 700 MHz!

In my opinion, this is a difficult season for those of us with limited budgets, and who rely on wireless microphones. My personal approach is a combination: I’m putting back-line vocalists on wired mics and wired in-ear monitors to reduce the number of wireless systems, but I’m not shying away: I’m using competent wireless, centered around the FCC’s clear channels. I’ve been using wireless with high channel counts, but I’m testing the Line6 product as an alternative, and so far, I’m really impressed.

The Line6 bodypack system uses a good connector (it’s a Switchcraft TA3F), and there are lapel mics and headset mics aplenty for them. The handheld integrates Line6’s renowned modeling experience: the single dynamic element will give you the audio based on (modeled after) six top live-sound mics including a Shure® SM58®, a Shure® Beta 58A®, a Sennheiser® e 835 and more. If that weren’t enough, you can actually put most Shure capsules on the Line6 transmitter. They have an “entry level” system at around $350, but I find myself most impressed with the stronger, more durable XD-V70 systems at under $500.

One story: I used the Line6 handheld in a public festival recently. More accurately, I let the lead singer use it in a tiny little gazebo that was functioning as a stage in the city park where the concert was taking place. I’d set the handheld to work like an SM58, thinking “We need an awfully forgiving mic for an uncertain gig.” It failed: immediate feedback; the vocalist scowled and handed me the mic back. I re-set the mic to emulate an ElectroVoice ND767B® and returned the mic to him, "Try this one." He used it for the whole concert, and raved about it afterwards: “It’s so much better than the first mic you gave me!” I laughed.

You know: Line6 has never been named among pro audio brands. I’m thinking that it’s time it was. There's change in the wind.

How to Remove Audio Feedback through Equalization

2010-12-20T08:06:00.000-08:00

By Chris Huff

Not all feedback is eliminated in the same way. Feedback typically occurs when a microphone and a loudspeaker are too close together thanks to a singer dropping the mic to their side. In this case, let's call it user error. But have you ever had multiple people use the same microphone and suddenly you hear the ringing of feedback? Have you ever created feedback by altering the EQ of a channel? Let's find out why.

What is feedback?

Audio feedback is the sound created when a sound loops between an audio input and an audio output. A simple example is a microphone and a monitor. The monitor is broadcasting sound the microphone then picks up. The monitor then is amplifying that sound and broadcasting it back out where the microphone picks it up again. Eventually, when the volume going into the microphone is the same as the volume coming out of the monitor, feedback begins.

The first frequency that feeds back is the one that requires the least amount of energy to excite resonance. Resonance is a vibration of large amplitude caused by a relatively small stimulus of the same or nearly the same period as the natural vibration period of the system.

What are the common reasons for audio feedback?

Microphone located too close to a monitor
Gain structure set too high so as frequencies primed for feedback

What can be done to stop audio feedback in these cases?

Move the microphone
Move the monitor
Use a more directional microphone to meet your mic'ing need.
Turn down the monitor volume
Turn down microphone gain (likely the person needs to hold the mic to their lips if singing)
Watch for reflective surfaces that might be bouncing the monitor sound to a microphone not directly in line with the monitor. Then, make changes then using one of the above.
Simple but common...turn off microphones when not in use. A stage arrangement can change for an event and create the right conditions for an open mic to cause feedback.
Equalize the microphone channel signal, lowering the frequencies which are causing the feedback.

How does the equalization-for-feedback process work?

In the first part of the article, I mentioned the frequency that required the least amount of energy to excite resonance. Let's lasso that one to the ground!

Frequencies by their sound:

Hoots and howls: Likely caused by a feedback frequency in the 250 to 500 Hz range.
Singing: The range is in-line with 1kHz.
Whistles and screeches: most likely above 2 kHz.

Determine the likely frequency range and then apply a cut to that range by 3dB.

What about creating feedback when EQ'ing a channel?

It's that very EQ process where we can cause feedback ourselves. For example, one time I had choir mic's all set and EQ'ed to my liking. During a specific song, I decided to try boosting the mid-range EQ a bit more (that 1kHz range). That's when the feedback started. I quickly cut that mid-range frequency back before anyone (except my sound guy, Jeff) noticed.

The keys to feedback control

Eliminate the conditions in which it can appear. Teach singers to hold the mic right up to their lips...and never drop down next to a monitor, establish proper gain structure, and turn off unused mic's.

When it does appear, know that you have an immediate alternative to turning down volumes...you might just be able to EQ it out.

Question(s): What have you done to control feedback in your environment?

Fitting a Countryman E6 Mic

2010-12-14T12:58:00.000-08:00

The Countryman E6 and E6i mics are among the most popular mics in the world today. But fitting them is a little tricky. Here's how it works best.

When Old Becomes New: Hearing Loop Assisted Listening Systems

2010-12-01T09:38:00.000-08:00

As more of our nation is diagnosed with hearing loss, it's important that we offer our parishioners the best assisted listening options possible.

It’s always interesting to see old technology become new again.

We see it daily in our lives, but I’m referring to a specific type of old tech; hearing loops.

The technology, which is correctly known by the name of inductive loops, has been in existence for decades.

Based on Faraday’s law of induction, a magnetic field is created and individuals that have a Telecoil (t-coil) equipped hearing aid, can receive audio signals directly in their hearing aid.

A “loop” is a relatively low-tech solution. In simple terms an area is surrounded “looped” with a piece of copper wire.

An amplifier drives signal down that wire creating a magnetic field inside the “loop”.

To receive the magnetic signal a person inside of the field simply has to switch their hearing aid to the “T” (t-coil) position.

With the current ADA (American Disabilities Act) requirements and an aging population, providing a loop based hearing assistance system can be beneficial not only for compliance but also for the convenience of our parishioners.

“Hearing loss is a significant issue. As the baby boomers come of age, the years of industrial sounds and/or noise pollution coupled with exposure to loud music has accelerated the problem,” says Todd Billin of Hearing Loop Systems.

With 36 million of Americans reporting hearing loss and over 8 million already outfitted with hearing aids, the need for hearing loops continues to grow.

There has been no stronger advocate than David G. Myers, PhD, Professor and Social Psychologist at Hope College.

When visiting Europe and attending a service at an old cathedral, Dr. Myers noticed a sign on the wall and remembered the words of the audiologist he purchased the hearing aids.

He knows upon seeing the “Hearing Loop Installed” sign that if he flipped a switch on his hearing aid he’d be tapping directly into the house audio.

Sure enough, from the moment Dr. Myers flipped that switch he has been and continues to be a huge supporter of the technology.

He is the creator of hearingloop.org, a website that provides education and information on hearing loops.

“I activate my T-Coils and instantly the speaker’s voice comes to me not from some distant loudspeaker but seemingly from the center of my head, says Myers.

“My hearing aids now serve me as customized wireless loudspeakers”

There can, of course, be challenges when installing a loop system.

The most obvious being how to route the cable in an existing space.

“In all of the existing buildings in which I have designed and installed loops, including very old churches, airport terminals and sporting venues, like the Michigan State Breslin Center, I have always found somewhere to hide the wire for the loop system,” says Tim Vander Meer of Hearing Loop systems.

A loop can cause a high pitched noise through some instruments pickups such as guitar, piano, strings.

Generally speaking this usually happens on older pickups or installations where there is not a solid ground present.

Spillover into adjacent rooms can also be an issue; this also can be overcome by using a properly designed phased array system.

The advantage of a loop is that there is no need for portable receivers and batteries as the t-coil in the hearing aid requires no power.

Additionally the hygiene issue goes away as earphones do not need to be cleaned and shared.

However, for those individuals not using t-coil equipped hearing aids, portable receivers are still, of course, available.

There is also a cost/benefit savings compared to traditional RF and Infrared systems, as the only limit on users of loop systems is physically how many people can fit inside of a loop.

As you consider assisted listening systems for parishioners in the future, whether for compliance or convenience, I would certainly consider a loop system.

What are your thoughts on assisted listening systems? Let me know in the comments below!

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

From ProSoundWeb, used with permission.

Channel Inserts

2010-11-14T09:07:00.000-08:00

The Signal Route
The Ins and Outs of Channel Inserts

One of the features often found on the rear panel of a mixing console is the Channel Insert. The insert serves simultaneously as both an input and an output for either a single channel or for some other signal path, such as a submix or main output bus. It is a point in the signal path at which the signal can be detoured — sent out of the mixer — and then returned to its normally scheduled programming, creating what is called an effects loop. In other words, it allows you to “insert” an outboard device into the signal path. On most mixers, a single ¼” three-conductor jack provides connections for both an input and an output.

What would you do with such a strange jack?
1. Apply effects to a channel or submix. Because an insert is both an input and an output, you can route the signal from the channel out to a reverb, compressor, limiter, etc., and then back into the channel. You might send the signal to a noise gate to automatically “turn off” a mic when it's not in use. Reducing the number of mics that are on, or “open”, reduces the risk of feedback and improves your signal-to-noise ratio.

2. Use it as a direct output, like a post-mic preamp, but pre low-cut filter, mute, EQ, fader, etc. Just because you're sending something out doesn't mean you have to bring it back. You can use each insert to send a “direct out” signal to a line-level input of a tape recorder, or to another mixer for a broadcast or recording feed. At the mixer end of your direct out cable, you'll want a standard 1/4" mono (or TS, tip/sleeve) phone plug. Push the phone plug part way into the insert jack, just to the first click. This will route the direct out signal via the cable, without interrupting the signal flow in the mixer. If you insert the plug all the way to the second click, you will still get a direct out signal, but the signal in the channel will be interrupted at that point — removed from the mix.

3. Insert a signal through a “Y” cable — using the insert as both a direct out and an effects loop. As an alternate approach, create your effects loop as described earlier, then insert a “Y” adapter after the processor to affect (compress, for example) both the direct out and the individual channel in the mix. A good application for this might be to compress a pastor’s lapel mic or a pulpit mic, in both the house mix and a recording or broadcast.

Whether you use them as part of your normal setup every week, or just to solve an occasional routing problem, inserts add tremendously to the versatility of your mixing console.

Information provided by kind permission, Mackie Designs, Woodinville, Washington. Reprinted from Mackie Church Sound Notebook

Sound System Interconnection

2010-10-25T13:06:00.000-07:00

Or: Dealing With Ground Loop Hums

Rane Technical Staff

RaneNote 110 written 1985; last revised 12/2009

Introduction

This note, originally written in 1985, continues to be one of our most useful references. Its popularity stems from the continual and perpetual difficulty of hooking up audio equipment without suffering through all sorts of bizarre noises, hums, buzzes, whistles, etc.-- not to mention the extreme financial, physical and psychological price. As technology progresses it is inevitable that electronic equipment and its wiring should be subject to constant improvement. Many things have improved in the audio industry since 1985, but unfortunately wiring isn't one of them. However, finally the Audio Engineering Society (AES) has issued a standards document for interconnection of pro audio equipment. It is AES48, titled "AES48-2005: AES standard on interconnections -- Grounding and EMC practices -- Shields of connectors in audio equipment containing active circuitry."

Rane's policy is to accommodate rather than dictate. However, this document contains suggestions for external wiring changes that should ideally only be implemented by trained technical personnel. Safety regulations require that all original grounding means provided from the factory be left intact for safe operation. No guarantee of responsibility for incidental or consequential damages can be provided. (In other words, don't modify cables, or try your own version of grounding unless you really understand exactly what type of output and input you have to connect.)

Ground Loops

Almost all cases of noise can be traced directly to ground loops, grounding or lack thereof. It is important to understand the mechanism that causes grounding noise in order to effectively eliminate it. Each component of a sound system produces its own ground internally. This ground is usually called the audio signal ground. Connecting devices together with the interconnecting cables can tie the signal grounds of the two units together in one place through the conductors in the cable. Ground loops occur when the grounds of the two units are also tied together in another place: via the third wire in the line cord, by tying the metal chassis together through the rack rails, etc. These situations create a circuit through which current may flow in a closed "loop" from one unit's ground out to a second unit and back to the first. It is not simply the presence of this current that creates the hum -- it is when this current flows through a unit's audio signal ground that creates the hum. In fact, even without a ground loop, a little noise current always flows through every interconnecting cable (i.e., it is impossible to eliminate these currents entirely). The mere presence of this ground loop current is no cause for alarm if your system uses properly implemented and completely balanced interconnects, which are excellent at rejecting ground loop and other noise currents. Balanced interconnect was developed to be immune to these noise currents, which can never be entirely eliminated. What makes a ground loop current annoying is when the audio signal is affected. Unfortunately, many manufacturers of balanced audio equipment design the internal grounding system improperly, thus creating balanced equipment that is not immune to the cabling's noise currents. This is one reason for the bad reputation sometimes given to balanced interconnect.

A second reason for balanced interconnect's bad reputation comes from those who think connecting unbalanced equipment into "superior" balanced equipment should improve things. Sorry. Balanced interconnect is not compatible with unbalanced. The small physical nature and short cable runs of completely unbalanced systems (home audio) also contain these ground loop noise currents. However, the currents in unbalanced systems never get large enough to affect the audio to the point where it is a nuisance. Mixing balanced and unbalanced equipment, however, is an entirely different story, since balanced and unbalanced interconnect are truly not compatible. The rest of this note shows several recommended implementations for all of these interconnection schemes.

The potential or voltage which pushes these noise currents through the circuit is developed between the independent grounds of the two or more units in the system. The impedance of this circuit is low, and even though the voltage is low, the current is high, thanks to Mr. Ohm, without whose help we wouldn't have these problems. It would take a very high resolution ohm meter to measure the impedance of the steel chassis or the rack rails. We're talking thousandths of an ohm. So trying to measure this stuff won't necessarily help you. We just thought we'd warn you.

The Absolute Best Right Way To Do It

The method specified by AES48 is to use balanced lines and tie the cable shield to the metal chassis (right where it enters the chassis) at both ends of the cable.

Figure 1a. The right way to do it.

Figure 1b. Recommmended practice.

A balanced line requires three separate conductors, two of which are signal (+ and -) and one shield (see Figure 1a). The shield serves to guard the sensitive audio lines from interference. Only by using balanced line interconnects can you guarantee (yes, guarantee) hum-free results. Always use twisted pair cable. Chassis tying the shield at each end also guarantees the best possible protection from RFI [radio frequency interference] and other noises [neon signs, lighting dimmers].

Neil Muncy, an electroacoustic consultant and seasoned veteran of years of successful system design, chairs the AES Standards Committee (SC-05-05) working on this subject. He tirelessly tours the world giving seminars and dispensing information on how to successfully hook-up pro audio equipment². He makes the simple point that it is absurd that you cannot go out and buy pro audio equipment from several different manufacturers, buy standard off-the-shelf cable assemblies, come home, hook it all up and have it work hum and noise free. Plug and play. Sadly, almost never is this the case, despite the science and rules of noise-free interconnect known and documented for over 60 years (see References for complete information).

It all boils down to using balanced lines, only balanced lines, and nothing but balanced lines. This is why they were developed. Further, that you tie the shield to the chassis, at the point it enters the chassis, and at both ends of the cable (more on `both ends' later).

Since standard XLR cables come with their shields tied to pin 1 at each end (the shells are not tied, nor need be), this means equipment using 3-pin, XLR-type connectors must tie pin 1 to the chassis (usually called chassis ground) -- not the audio signal ground as is most common.

Not using signal ground is the most radical departure from common pro-audio practice. Not that there is any argument about its validity. There isn't. This is the right way to do it. So why doesn't audio equipment come wired this way? Well, some does, and since 1993, more of it does. That's when Rane started manufacturing some of its products with balanced inputs and outputs tying pin 1 to chassis. So why doesn't everyone do it this way? Because life is messy, some things are hard to change, and there will always be equipment in use that was made before proper grounding practices were in effect.

Unbalanced equipment is another problem: it is everwhere, easily available and inexpensive. All those RCA and 1/4" TS (Tip-Sleeve) connectors found on consumer equipment; effect-loops and insert-points on consoles; signal processing boxes; semi-pro digital and analog tape recorders; computer cards; mixing consoles; et cetera.

The next several pages give tips on how to successfully address hooking up unbalanced equipment. Unbalanced equipment when "blindly" connected with fully balanced units starts a pattern of hum and undesirable operation, requiring extra measures to correct the situation.

The Next Best Right Way To Do It

The quickest, quietest and most foolproof method to connect balanced and unbalanced is to transformer isolate all unbalanced connections. See Figure 2.

Figure 2. Transformer Isolation

Many manufacturers provide several tools for this task, including Rane. Consult your audio dealer to explore the options available.

The goal of these adapters is to allow the use of standard cables. With these transformer isolation boxes, modification of cable assemblies is unnecessary. Virtually any two pieces of audio equipment can be successfully interfaced without risk of unwanted hum and noise.

Another way to create the necessary isolation is to use a direct box. Originally named for its use to convert the high impedance, high level output of an electric guitar to the low impedance, low level input of a recording console, it allowed the player to plug "directly" into the console. Now this term is commonly used to describe any box used to convert unbalanced lines to balanced lines.

The Last Best Right Way To Do It

If transformer isolation is not an option, special cable assemblies are a last resort. The key here is to prevent the shield currents from flowing into a unit whose grounding scheme creates ground loops (hum) in the audio path (i.e., most audio equipment).

It is true that connecting both ends of the shield is theoretically the best way to interconnect equipment -- though this assumes the interconnected equipment is internally grounded properly. Since most equipment is not internally grounded properly, connecting both ends of the shield is not often practiced, since doing so usually creates noisy interconnections.

A common solution to these noisy hum and buzz problems involves disconnecting one end of the shield, even though one can not buy off-the-shelf cables with the shield disconnected at one end. The best end to disconnect is the receiving end. If one end of the shield is disconnected, the noisy hum current stops flowing and away goes the hum -- but only at low frequencies. A ground-sending-end-only shield connection minimizes the possibility of high frequency (radio) interference since it prevents the shield from acting as an antenna to the next input. Many reduce this potential RF interference by providing an RF path through a small capacitor (0.1 or 0.01 microfarad ceramic disc) connected from the lifted end of the shield to the chassis. (This is referred to as the "hybrid shield termination" where the sending end is bonded to the chassis and the receiving end is capacitively coupled. See Neutrik's EMC-XLR for example.) The fact that many modern day installers still follow this one-end-only rule with consistent success indicates this and other acceptable solutions to RF issues exist, though the increasing use of digital and wireless technology greatly increases the possibility of future RF problems.

If you've truly isolated your hum problem to a specific unit, chances are, even though the documentation indicates proper chassis grounded shields, the suspect unit is not internally grounded properly. Here is where special test cable assemblies, shown in Figure 3, really come in handy. These assemblies allow you to connect the shield to chassis ground at the point of entry, or to pin 1, or to lift one end of the shield. The task becomes more difficult when the unit you've isolated has multiple inputs and outputs. On a suspect unit with multiple cables, try various configurations on each connection to find out if special cable assemblies are needed at more than one point.

Figure 3. Test cable

See Figure 4 for suggested cable assemblies for your particular interconnection needs. Find the appropriate output configuration (down the left side) and then match this with the correct input configuration (across the top of the page.) Then refer to the following wiring diagrams.

Figure 4. Interconnect chart for locating correct cable assemblies.

Note: (A) This configuration uses a standard "off-the-shelf" cable.

Note: (B) This configuration causes a 6 dB signal loss. Compensate by "turning the system up" 6 dB.

Ground Lifts

Many units come equipped with ground lift switches. In only a few cases can it be shown that a ground lift switch improves ground related noise. (Has a ground lift switch ever really worked for you?) In reality, the presence of a ground lift switch greatly reduces a unit's ability to be "properly" grounded and therefore immune to ground loop hums and buzzes. Ground lifts are simply another Band-Aid to try in case of grounding problems. It is, however, true that an entire system of properly grounded equipment, without ground lift switches, is guaranteed (yes guaranteed) to be hum free. The problem is most equipment is not (both internally and externally, AC system wise) grounded properly.

Most units with ground lifts are shipped so the unit is "grounded" -- meaning the chassis is connected to audio signal ground. (This should be the best and is the "safest" position for a ground lift switch.) If after hooking up your system it exhibits excessive hum or buzzing, there is an incompatibility somewhere in the system's grounding configuration. In addition to these special cable assemblies that may help, here are some more things to try:

Try combinations of lifting grounds on units supplied with lift switches (or links). It is wise to do this with the power off!
If you have an entirely balanced system, verify all chassis are tied to a good earth ground, for safety's sake and hum protection. Completely unbalanced systems never earth ground anything (except cable TV, often a ground loop source). If you have a mixed balanced and unbalanced system, do yourself a favor and use isolation transformers or, if you can't do that, try the special cable assemblies described here and expect it to take many hours to get things quiet. May The Force be with you.
Balanced units with outboard power supplies (wall warts or "bumps" in the line cord) do not ground the chassis through the line cord. Make sure such units are solidly grounded by tying the chassis to an earth ground using a star washer for a reliable contact. (Rane always provides this chassis point as an external screw with a toothed washer.) Any device with a 3-prong AC plug, such as an amplifier, may serve as an earth ground point. Rack rails may or may not serve this purpose depending on screw locations and paint jobs.

Floating, Pseudo, and Quasi-Balancing

During inspection, you may run across a 1/4" output called floating unbalanced, sometimes also called psuedo-balanced or quasi-balanced. In this configuration, the sleeve of the output stage is not connected inside the unit and the ring is connected (usually through a small resistor) to the audio signal ground. This allows the tip and ring to "appear" as an equal impedance, not-quite balanced output stage, even though the output circuitry is unbalanced.

Floating unbalanced often works to drive either a balanced or unbalanced input, depending if a TS or TRS standard cable is plugged into it. When it hums, a special cable is required. See drawings #11 and #12, and do not make the cross-coupled modification of tying the ring and sleeve together.

Summary

If you are unable to do things correctly (i.e. use fully balanced wiring with shields tied to the chassis at the point of entry, or transformer isolate all unbalanced signals from balanced signals) then there is no guarantee that a hum free interconnect can be achieved, nor is there a definite scheme that will assure noise free operation in all configurations.

Winning the Wiring Wars

Use balanced connections whenever possible, with the shield bonded to the metal chassis at both ends.
Transformer isolate all unbalanced connections from balanced connections.
Use special cable assemblies when unbalanced lines cannot be transformer isolated.
Any unbalanced cable must be kept under ten feet (three meters) in length. Lengths longer than this will amplify all the nasty side effects of unbalanced circuitry's ground loops.
When all else fails, digitize everything, use fiber optic cable and enter a whole new realm of problems.

References

Neil A. Muncy, "Noise Susceptibility in Analog and Digital Signal Processing Systems," presented at the 97th AES Convention of Audio Engineering Society in San Francisco, CA, Nov. 1994.
Grounding, Shielding, and Interconnections in Analog & Digital Signal Processing Systems: Understanding the Basics; Workshops designed and presented by Neil Muncy and Cal Perkins, at the 97th AES Convention of Audio Engineering Society in San Francisco, CA, Nov. 1994.
The entire June 1995 AES Journal, Vol. 43, No. 6, available $6 members, $11 nonmembers from the Audio Engineering Society, 60 E. 42nd St., New York, NY, 10165-2520.
Phillip Giddings, Audio System Design and Installation (SAMS, Indiana, 1990).
Ralph Morrison, Noise and Other Interfering Signals (Wiley, New York, 1992).
Henry W. Ott, Noise Reduction Techniques in Electronic Systems, 2nd Edition (Wiley, New York, 1988).
Cal Perkins, "Measurement Techniques for Debugging Electronic Systems and Their Instrumentation," The Proceedings of the 11th International AES Conference: Audio Test & Measurement, Portland, OR, May 1992, pp. 82-92 (Audio Engineering Society, New York, 1992).
Macatee, RaneNote "Grounding and Shielding Audio Devices," Rane Corporation, 1994.
Philip Giddings, "Grounding and Shielding for Sound and Video," S&VC, Sept. 20th, 1995.
AES48-2005: AES standard on interconnections "Grounding and EMC practices -- Shields of connectors in audio equipment containing active circuitry" (Audio Engineering Society, New York, 2005).

Band-Aid is a registered trademark of Johnson & Johnson

Note: this is not basic Sound Guy instruction: these are advanced tools. In other words, don't modify cables, or try your own version of grounding unless you really understand exactly what type of output and input you have to connect.

This document is also available as a pdf document.

Note: A version of this RaneNote was published in the Journal of the Audio Engineering Society, Vol. 43, No. 6, June, 1995.

Install Your Own Church Sound System? Here Are Some Cautionary Tales

2010-10-15T10:03:00.000-07:00

While installing a sound system isn't exactly rocket science, it is more complex than painting one's house. That's one reason why you need to do your homework

September 10, 2010, by Curt Taipale

Audio consultants often find themselves working with people in churches who seem eternally bent on saving money at any cost. This is the kind of church that will call with the seemingly innocent request to have the consultant design a new sound system for them. At some point in the conversation they’ll add that they want to do the installation themselves.

That approach can be a mixed blessing both for the consultant and for the church. On the one hand, at least they’re using a consultant’s seasoned advice to make the best choices of gear for them to use. The problem starts when they begin to think that the process of actually installing the gear isn’t all that difficult.

Momentary Lapses of Intelligence
Here are some textbook cases. In order to protect the innocent, I’ll use their real names.

So one day my friend Warren calls me and announces that his church is ready to renovate their existing sound system, and they want to do it right this time. He invites me to meet with their sound committee, and within a few days I’ve got the project to design the system.

In order to save money, the church plans to use volunteers to run all the wire, hang the loudspeakers, and wire up the sound booth gear. I insisted on wiring up the amplifier rack myself, to be a friend, save them the work, and me the headache of possibly having to fix it later.

A couple of months later the equipment is all sitting at the church, and the troops are ready to proceed with the install. So I arrive with TEF and solder station in hand ready to talk them through the install.

Now right off the bat, I’m scared by what I see. To free myself from any liability in the future, I do what every good consultant does - I don’t give them any advice at all about how to hang the loudspeakers in a safe manner. That’s really the job of the sound contractor.

They assure me that they’ve researched their hanging method carefully, and at my insistence have even had a structural engineer sign off on their solution, but I make a mental note to not find myself standing under the cluster for any length of time.

After a lot of scraped knuckles, sweat, grunts and groans, the loudspeaker wire, microphone snakes and return lines are finally pulled into place. At around 1 am on the third day of the installation, we finally light up the system and start to voice it.

By this time, everyone is toast. I’m so tired I can hardly see straight, let alone hear really well. The volunteer crew is absolutely wiped out, but we’re so close now that they’re not about to leave without hearing the system lit up for the first time, so they’re napping on the pews while I continue to work.

To their credit, there were no polarity reversals anywhere in the system. Bless God, somebody was paying attention.

Don’t get me wrong. The church loves their new sound system. And I’m sure the crew has good memories of the time they invested on that project.

But by the end of the project everyone was wiped out, stressed out, on the verge of being mad at everyone, and just plain in a bad mood.

More Angst
I recently finished another project like this. My friend Duane had his best “ain’t no way on earth that’ll happen” look on his face when he considered the idea of using a sound contractor to do their installation.

So I designed the system, gave them a shopping list, and answered a myriad of questions as the project went from a few pieces of paper to loudspeakers hanging somewhat precariously from the steel.

Here again, the weakness seems to come in not knowing precisely how to safely hang really heavy loudspeakers over people’s heads. Hanging heavy loudspeakers isn’t easy in the first place. Getting them aimed precisely where they need to be aimed is an additional challenge.

But when I saw the loudspeakers hanging from S-hooks and swing-set chain, I knew they had ignored my urging to buy their hardware from a professional rigging supplier. They didn’t have a smile on their face when I insisted that they replace the chain and hardware with the real stuff. And don’t even get me started on the points they wanted to hang the boxes from.

Part of the angst of Duane’s project came through the scheduling. All involved wanted the system to be in place in time for their Easter pageant. Flying the loudspeakers meant having to move scaffolding into the room in order to pull wire and hang loudspeakers.

During the same time frame, the drama and music team needed access to the stage for their pageant rehearsals several evenings each week, so having scaffolding on the stage was a problem. Just try sharing a stage with those two groups.

The installers had to remove the scaffolding and all of their stuff each evening so that the pageant rehearsals could continue as scheduled. That process added undue pressure on the volunteer sound installers.

You wouldn’t make the same mistake?! I’m sure that’s what Duane felt. Happened anyway. Nobody lost their salvation over it, but it’s certainly something they wouldn’t do on purpose again.

Before I continue, please understand. The guys that find themselves in these predicaments aren’t dolts. They’re bright, sharp, astute, focused, detail-type personalities.

But by the time they realize that they’re in over their heads, it’s too late to drop back and do anything else about it. They’ve got to see it through and get on with life.

So what makes such a rational, educated person think that they can install a sound system just as well as a seasoned sound contractor? Contractors have years of hard-won experience they can draw upon every time they hang a loudspeaker or wire up a rack.

As well meaning as they are, churches who set out to do this work on their own are in no better position than that sound contractor on his very first install.

I talked recently with my new friend, Rod, who was just then receiving the equipment that I specified and he ordered. His conversation with me then was filled with the usual confidence that both Duane and Warren shared in their first dialogs with me.

Rod was certain that he could have the cluster in the air and ready to get sound out of it within the next couple of weeks. I tried my best to cool his optimism while still being encouraging. I knew that if his experience was anything like most of the others, he’d be in for a real surprise!

Well, I just spent this past weekend commissioning the sound system that I designed and that Rod and friends installed. I called him last Thursday night before I left to make sure that he was really, truly ready for me to be there, and he assured me that all would be fine.

When I arrived in his town, I called again and his response was, “Well, we’ll be ready, but don’t hurry over here.” As I walked into the church, he had just finished making the final connections in the amp rack. To their credit, our system voicing process wasn’t delayed.

Rod finally realized - just like Warren, and Duane, and others have - that installing a sound system properly isn’t as simple or as easy as many want to think early on in the process.

Yeah, But ...
Look, I know your church is different. I know y’all won’t make the same mistakes that most other churches make during this process. And I know your church will end up with an award-winning sound system that will make every sound contractor green with envy.

But just humor me. Tell me you’re at least going to consider hiring a first rate sound contractor for your next system installation. It’ll make me feel better.

For what it’s worth, I also know that there are some contractors out there who shouldn’t be in business. Frankly, you probably could do the work better than some sound contractors out there.

Even though we’re not professional painters, I think that my wife and I do a more careful job of painting our house simply because it’s our house - we live there every day and care about it more than your typical painting contractor would.

But as much as I know about electricity and electronics, I’m not going to volunteer to wire our next house. I might do some extra stuff - like putting lights in the closets, adding phone outlets in all of the rooms, and so on. But I’m not interested in doing the entire job myself.

While installing a sound system isn’t exactly rocket science, it is more complex than painting one’s house. That’s one reason why you need to do your homework on the contractors you’re considering.

Please at least seriously think through all of the realities before you let your church go off the deep end in their eagerness to simply save some money. They may save a few dollars during the installation, but the toll that the process exacts from the church’s volunteers may not be worth it in the long run.

People are more important than money. And it may be that later on y’all will find yourselves doing the work all over again.

A Quiet Voice of Reason
Now, don’t go around telling everyone that Curt said that no church should install their own sound system. I didn’t say that. All I hope to offer here is a voice of reason in your eager pursuit to save a couple of bucks.

If you’re thinking about installing your own sound system, please determine now that you will sort through every possible issue. Develop a contingency plan for all of the things that are going to go differently than you plan, because they will. Step back and think it through before your eagerness gets the best of you.

For example, what are you going to do when the input panels for the floor pockets don’t come in with the connectors laid out the way you told them to? What are you going to do when you discover - after the scaffolding and scissor lift are long gone - that you hung the cluster two feet higher than it should have been?

What are you going to do when the mic snake you ordered arrives with totally the wrong connectors? What are you going to do when your consultant discovers through his acoustical testing that two of your four main loudspeakers have their woofers wired out of polarity - that they came that way from the loudspeaker manufacturer!?!

No, really. Tell me what you’re going to do. Because if you’re installing the sound system yourself, you ARE the sound contractor. It’s your job to make sure the installation and every device in the project is working correctly and installed properly.

And if you’re like most churches, you’re not only installing the sound system, you’re also installing the video system, and the stage lighting system, and… The size of the task can mushroom beyond your wildest expectations in no time.

I assume you’ll be trying to accomplish this task while gainfully employed in another job, so your installation efforts will be done in the evenings and on weekends. You’ll probably need to take vacation time during the last few days of the project when everything comes together.

And I assume that, if you can find volunteers as eager to help you as you are to take on this project, that they too will be there whenever they can. Be prepared to discover that their available times might not be the same times as you plan to work, or nearly as often.

Do your best to step away from the project long enough to see the big picture and what the process is going to do to you, to your life, to your family, to your friends who are going to help you get this job done, and to your church.

The church as a whole has enough people who have been burned out or hurt emotionally through their service to their local church. We don’t need to add any new people to that list.

Uncle, Uncle
Okay, I’ve beat you up enough. If after all of this you’re still convinced that you need to do the install yourself, get prayed up and go for it. As long as you know up front that it’s not as easy as you think it will be.

The reality is that there can be tremendous value to having church staff and/or volunteers install their own sound system. More important than the money you’ll save is the fact that they’ll emotionally take ownership of the system more quickly.

Also, if anything ever goes wrong with the system - and we both know that will be discovered on Sunday morning before the service - your volunteers will know where every piece of equipment and scrap of wire is in the entire facility, and how it’s hooked up.

They may even be able to track down the problem and fix it before the service instead of sometime later next week when the sound contractor’s audio technician can schedule an appointment. That works of course until the folks that did the install get relocated by their job, or move to another church for some reason.

Remember that, whatever happens, God is still on the throne. So have fun. Or else.

Curt Taipale heads up Church Soundcheck, a thriving community dedicated to helping technical worship personnel. Courtesy ProSoundWeb; used by permission.

Shure's Summary of the 2010 White Spaces Order

2010-10-08T09:59:00.000-07:00

On September 23, 2010, the FCC issued a Second Memorandum Opinion and Order finalizing rules that make the “white spaces” in the TV bands – unoccupied channels between over-the-air TV stations – available for use by unlicensed broadband wireless devices such as next-generation smartphones, computers, and other consumer and commercial products. The 2010 Order mandates the operating rules and technical specifications for both Fixed and Portable TV Band devices, and finalizes some legal and technical issues that remained unresolved since the previous November 2008 Order.

Wireless Microphones Remain Legal Throughout TV Bands

The White Spaces Order does not alter the ability of wireless microphones, in-ear monitors, intercom systems, and related equipment to operate in the TV bands. Wireless microphone users may continue to operate, with or without a license, on any VHF or UHF TV channel (2-51, except 37) that is not assigned to a local TV station or Public Safety agency.

Avoiding Interference In The Future

The 2010 Order establishes a set of operating protocols that create safe havens for wireless microphone users who may encounter interference in the future. The majority of wireless microphone users (who typically use fewer than 20 wireless systems) will be able to operate in designated TV channels that are off-limits to TV Band Devices, thus eliminating the potential for interference. Expanded protection for large events will be provided through a geo-location database. Together these protection schemes will enable both small and large users of wireless microphones to operate without interference from new TV Band Devices.

Reserved Channels Around 37

The 2010 Order stipulates that two TV channels in each market will be reserved for wireless microphone use. These will be the first channel above and the first channel below TV channel 37 that are not occupied by a local TV station. If unoccupied channels are not available both above and below channel 37, the first two channels nearest to channel 37 will be reserved. Because occupied TV channels vary by market, the reserved channels will vary by market also. The channel map in a particular city might look like this:

Because one TV channel can accommodate up to 8 wireless microphones (depending on model), the two reserved channels will allow users to operate up to 16 wireless microphones with no threat from interference from TV Band Devices.

Clear Channels Adjacent To TV Stations (14-20)

Additional TV channels between 14 and 20 will also be off-limits to TV Band Devices due to the technical restrictions that govern their operation. Portable TV Band Devices are not permitted to transmit on TV channels 14-20, and Fixed TV Band Devices are not permitted to transmit on the channels immediately adjacent to the TV channels used by local TV stations or Public Safety agencies. These adjacent channels in the 14-20 range are also effectively reserved for wireless microphone use without risk of interference from TV Band Devices. A sample channel map for a typical city where one TV channel is assigned for Public Safety communications is shown below.

Geo-Location Database Protects Large Events

Large-scale wireless mic users will be able to achieve expanded protection for specific events through the geo-location database prescribed by the FCC in 2008. The 2010 Order requires that every TV Band Device must receive (either directly or through another device) a list from the database of available TV channels at its location before transmitting. Both licensed and unlicensed wireless microphone users will be able to register the date, time, location, and duration of an event and the TV channels used by their wireless systems in the database. Portable TV Band Devices will be prevented from transmitting on those TV channels when they are within a 400-meter “exclusion zone” around that location; Fixed TV Band Devices are subject to a larger 1-kilometer exclusion zone.

Venues must request database registration at least 30 days in advance, and must certify that at least 6 wireless microphones are operating in each of the reserved TV channels available at that location. The FCC will make requests for database registration public and will provide an opportunity for public comment or objections. The Order does not specify who will administer the geo-location database or when it will be operational.

Spectrum Sensing No Longer Required

The 2008 Order stipulated that all TV Band Devices must include “spectrum sensing” technology that would enable them to detect and avoid TV stations as well as wireless microphones that are not registered in the database. The 2010 Order removes the spectrum sensing requirement, concluding that this technology would add cost and complexity to TV Band Devices and would be redundant to the protection provided by the geo-location database. Portable devices that rely on sensing only (without database access) will still be permitted, but will be limited to 50 milliwatts of transmit power and will be required to pass stringent laboratory and field tests to prove that they will not interfere with incumbent users.

No Changes To Wireless Microphone Licensing

Wireless microphone licenses are currently available to broadcasters, television and motion picture production companies, and cable TV networks. In January 2010, the FCC asked for comments regarding to what extent license eligibility should be expanded. The 2010 White Spaces Order does not address this issue, and given the extensive protections for unlicensed wireless microphone users mandated in this Order, it is unclear what importance the Commission will attach to the matter of license eligibility going forward.

What Happens Next

Before TV Band Devices enter the market, the database administrators must be selected; database access and registration procedures must be developed, and new devices must be submitted to the FCC for approval. The Commission is still considering potential changes to the spectrum (including the TV bands) in connection with the National Broadband Plan which raise complex legislative, regulatory, and technical issues that may take years to resolve.

Copyright 2010 Shure Incorporated
Original document found here.

Comment: This is a very valuable document. Shure has clarified an obtuse federal document, and translated into intelligible terms. Practical application: Every wireless mic that was legal before the decision is still legal after the decision, though we may have to be more intentional about our frequency choices in the future.

Opinion: This will definitely affect us, but not for some time. The required federal database won't be available until 2011, and the infrastructure required for such products won't likely be in place until at least 2012.

Shure Joins Legislators in Applauding FCC to Protect Wireless Microphone Users

2010-09-29T08:08:00.000-07:00

NILES, IL, September 23, 1010—Shure Incorporated and a number of Congressional legislators applauded a decision by the Federal Communications Commission to protect wireless microphone users from interference from “white space devices.” The Memorandum Opinion and Order issued today by the FCC reserves two TV channels nationwide for wireless microphone use. The reserved channels are off-limits to UHF Band Devices that operate in the white spaces between assigned TV stations, thus preventing them from interfering with wireless microphone signals on those channels. Large-scale users would be able to achieve extended protection for specific events through the geo-location database prescribed by the FCC in 2008.
“It’s clear that the FCC carefully considered the needs of wireless microphone users while crafting this Order,” said Sandy LaMantia, President and CEO of Shure Incorporated. “The reserved channels will provide a safe harbor in which musicians, small theaters, houses of worship, and businesses can operate their wireless microphone systems without interference from new TV Band Devices.”

Legislators have been actively following the FCC’s plan to allow unlicensed devices to share the “white spaces” and the potential impact on wireless microphone users. Representative Bobby Rush (D-IL) submitted a bill that would require wireless microphone users to be protected. “The legislation I introduced called for interference protection for professional wireless microphones in the wide variety of venues in which they are used today,” said Representative Rush. “This Order effectively grants that protection and will ensure continuity of service for houses of worship, theater, music tours and venues, sporting events, and the various civic and corporate environments that rely on quality audio in America today.”

Other legislators noted the contributions that wireless microphone users make to the nation’s economy. "Music is a lifeline to Nashville’s economy, and the live music industry depends more than ever on wireless microphones to connect our artists to the audience,” said Congressman Jim Cooper (TN-05). “The FCC's order helps guarantee that the flawless sound music fans expect will continue without interference from new consumer wireless devices."

“Las Vegas has always been -- and still remains -- the Entertainment Capital of the World and live performances are always big draw for tourists. Those who visit southern Nevada are drawn not only by the excitement of Las Vegas, but also by the wide array of musical talent and thrilling theatrical performances available. The FCC Order will enable headliners and performers up and down the Las Vegas Strip to keep delivering their innovative material live for the enjoyment of our tens of millions of guests," said Representative Shelley Berkley (D-NV).

Lawmakers also praised the thorough and detailed research that preceded the FCC’s ruling. "After many years of comprehensive discussions and scientific analysis, the FCC’s final White Spaces Rules recognize that on Broadway 'the show must go on' and that a modern theater performance requires the use of dozens of interference-free wireless microphone systems every single night," said Representative Carolyn Maloney (D-NY).

“The Commission received input over many years from a broad range of wireless microphone users during this proceeding,” said Mark Brunner, Senior Director of Global Brand Management at Shure. “Their activities – spanning live music, theater, worship, broadcasting, and many others – represent an important contribution to our society and to our economy. The Order demonstrates the FCC’s commitment to supporting America’s position as a leader in the creation of news, sports, cultural, and entertainment content at venues of all sizes.”

From Shure.

The FCC's full announcement is here.

A very helpful whitepaper concerning the implications of the decision is available here. 