This article is provided by Rane as part of the Rane Library and originally appeared in the Syn-Aud-Con Newsletter, vol. 31, nos. 2 & 3, 2003
In his landmark 1994 AES paper, Neil Muncy described a common equipment design error that allows current flowing on the shields of audio wiring to enter equipment and cause audible interference.
He called this design error “the Pin 1 problem,” because it was an improper connection of the shield terminal, pin 1 in XLR connectors.
The Pin 1 Problem
Cable shields are essentially an extension of the shielding enclosure of equipment, and they should be connected directly to that shielding enclosure.
To make equipment cheaper to build, manufacturers started connecting cable shields to the circuit board’s common trace, then took that trace to the chassis. The problem is that any voltage drop across the wiring that is common to both the shield current and the circuit’s path to ground will be injected into the audio circuitry.
Shield currents include the noise currents coupled into ground by power line filters, potential differences between “ground” at opposing ends of long cable runs, and the shield acting as an antenna, picking up RF dimmer noise and other radio signals. It’s quite common to have AM, FM, and TV broadcast signals flowing on the shields of audio wiring.
Neil has observed that most RF interference to audio equipment is caused by a pin problem, and some recent research I’ve done has convinced me that he’s right.
Here’s the inside of a rather expensive condenser mic (Figure 1), the manufacturer of which has long insisted that the shields of mic cables must be connected to the shell in order to prevent RF interference. A quick look at the photo tells us why—this mic has a screaming pin 1 problem at VHF frequencies!
A good connection in the audio band can be a bad connection in the RF band
Figures 1 and 2
The black wire takes pin 1 to the body of the mic, and the orange wire takes pin 1 up to the circuit board. At audio frequencies, that works just fine. But at 56 MHz (the frequency of TV channel 2), the inductive reactance of the black wire is about 4Ω. If you try to use this mic in downtown Chicago with a properly wired mic cable, the shield current induced by Channel 2, Channel 5, and a bunch of FM broadcast stations causes enough voltage drop in the black wire (which the orange wire adds to the audio circuitry) that both the video and FM signals are clearly heard! Rick Chinn’s review of this mic several years ago for a magazine noted RF interference from TV stations, the closest of which were 13 miles away!
This mic has a pin 1 problem too (Figure 2), although not as severe—it takes about 6 dB more RF to cause audible interference. Here, the shield goes to the chassis through the tiny wire connecting pin 1 to the broad tab holding the connector’s retaining screw. The combined length of that path is on the order of 1 cm, which results in about 2Ω of inductive reactance at 60 MHz. Again, circuit common is connected to pin 1 and sees the drop across the inductance. I was first alerted to the problem in this mic when I noticed that tightening the screw reduced the interference by about 3 dB!
The pin 1 problem works in reverse too—any RF noise currents (digital clock noise, for example) flowing in that common impedance create a voltage drop that is coupled out to the shield. The shield, acts as an antenna and noise is radiated to nearby equipment and wiring. It’s not at all uncommon to see shield paths inside equipment 3-5 inches long! At 100 MHz, a 5” long wire is about 60 Ω! At 1 GHz, it’s about 600 Ω.
Line Level Example
Figure 3 shows the rear panel connections to a DSP unit that has a bit of a noise problem. Looking behind the panel, we see the circuit traces go to the chassis via the two black screws at approximately the 4” and 7” points on the ruler that has been laid on top to show dimensions. All the incoming shields are bused together and go to the screws. These long leads (typically 2-4” with all the zigs and zags) are like open doors to radio frequency signals—any current flowing on the shields radiates inside through antenna action, and noise inside the box is coupled out onto the shields, which also act as antennas.
Internal conductors (including PCB traces) that connect to pin 1 can act as antennas.
The Right Way
There are better ways to do this. Some manufacturers (Figure 4 shows a Rane product) provide a chassis screw next to pluggable strip connectors for proper termination of the shield. Inside, the signal leads go directly to RF filter networks.
In a series of tests I recently did on more than 45 condenser mics using standard XLR connectors, nearly all experienced serious interference from cell phones, and about one-third experienced significant interference from TV broadcast stations and/or my ham talkie.
The Neutrik Solution
The European EMC directive, implemented in the late 1990’s, places limits on noise emissions from elec-tronic equipment sold in most European countries.
It caused a lot of manufacturers to get EMC religion, and gave new life to RF engineers working in labs dedicated to verifying compliance.
The engineers, working in these labs and those specializing in designing for good EMC performance, think in terms of RF immunity and computer/digital systems.
Few have much practical experience with analog audio systems, and some of the design solutions they advance can cause us considerable grief—especially the treatment of cable shields.
From an RF point of view, the shell of an XLR connector looks like it should work as the extension of the cable shield.
For broadband RF immunity, it can be helpful to ground shields at every opportunity and carefully bond all grounded objects together at multiple points. Such a philosophy is the basis of the so-called “mesh” ground topology, and it can work well in installations where there’s little difference in potential between grounds at opposite ends of the audio paths. But power system leakage currents cause enough power-related shield current to flow in most real world installations to couple noise into the sound system if both ends of a cable shield are grounded.
Since the 1930’s, engineers have known that audio frequency noise coupling will be minimized with single point (star) grounding, while radio frequency noise coupling is minimized with multi-point (mesh) grounding. The solution is simple—the shield of a balanced audio cable is connected to the shielding enclosure at the driving source, and to the shielding enclosure at the receiving end through a capacitor. It was easy to do this in the 40’s, when RF signals higher than 30 MHz were rarely encountered.
It’s much tougher now, when the interference sources are strong RF signals from UHF cell phones and high power TV transmitters. Again, it’s series inductance that makes life difficult, this time the inductance of the capacitor’s wire leads. See Figures 5 and 6.
Feed-through capacitors have been a solution to this problem since the 1930’s. They mount in a circular hole in an enclosure. One “plate” of the capacitor is a wire going through the enclosure, while the other “plate” is a cylinder surrounding the wire and connected to the enclosure with a dielectric (insulation) between the plates. This circular construction minimizes the inductance through the capacitor to the chassis enclosure, while the wire coming into the chassis enclosure still has inductance. The resulting electrical circuit is an effective RF filter while it also prevents a hole in the shielding.
To provide the most effective shielding, both cable shields and the shells of connectors within equipment should be bonded to the chassis. (Retired Bell Labs engineer and EMC authority Henry Ott observes that if this connection is to the outside of the enclosure, skin effect will keep RF currents outside the enclosure as well.) But what about cable-mount connectors? To make the EMC engineering community happy, mic cables should tie the cable shield to the connector shell, but to make audio folks happy the shield should go only to pin 1.
A few years ago, I proposed the concept of making a cylindrical connection to the shield in cable-mounted connectors much like that in a BNC connector, with that cylinder surrounded by a cylindrical dielectric that was itself surrounded by a cylindrical plate connected to the connector shell. Such construction would form a capacitor having very low series inductance, turning the XLR connector into a two-circuit feed-through capacitor. It would connect the shield to the shell at RF, while isolating it from the shell at audio frequencies and DC. When used on mic cable, the shield is also soldered to pin 1. When used at line level inputs in a rack, an installer could decide not to connect the shield to pin 1, but it would still be connected to the shell at RF through the capacitor.
Engineer Joanne Dow observed that if such a connector were to be used with a connection to pin 1, the capacitor and the series inductance to the chassis through pin 1 would form a parallel resonance, and suggested a ferrite bead surrounding pin 1 in the cable connector to lower the Q of the resonance.
It took a while for these ideas to germinate, but before long, engineers at Neutrik had begun work on a practical implementation (Figure 8), and by early 2002 had developed engineering prototypes. British consultant John Woodgate tested them in his lab, and I tested them both in my lab and in the field. The results met my expectations, and yielded an unexpected bonus—they solved RF pin 1 problems, even when pin 1 was connected at both ends!
A simple circuit study shows why.
Figure 7 shows a cable using the new connectors used with the mic shown in Figure 5.
The concentric capacitor ties the shield to the mic shell with an inductance that is much lower than the wire inside the mic.
Not only that, but the ferrite bead around pin 1 inside the new connector adds series impedance to the path through pin 1, effectively disconnecting the shield from the pin 1 problem.
Shield current thus divides, most of it taking the low impedance path through the capacitance to the shell, with very little flowing through pin 1. In effect, the new connector bypasses the problem!
But even this well engineered solution cannot be effective if mating connector shells don’t make good contact, or if the equipment connector shell is not bonded to the enclosure. I own a portable DAT machine that has a massive pin 1 problem, so much so that when it’s used with its own preamp with a dynamic mic in downtown Chicago the detected FM and TV stations are nearly as loud as an interviewer a foot from the mic.
The new EMC connector completely eliminates the detected RF—if I carefully push against the side of the connector to force the connector shells to make contact. But without that pressure, the shells can lose contact and the RF interference returns! And the new connector doesn’t help a popular mixer with a serious RF pin 1 problem, because the equipment connector shell is not connected to the enclosure!
Now that I’ve been alerted to them, these connector shell mating problems are turning out to be more prevalent than any of us would have suspected. It appears not to be limited to the use of off-brand or pirated connectors—it shows up on connectors that have been verified to have come from the same manufacturer! I’ve also seen it in connectors built into very good quality preamps and mics from a variety of manufacturers. So far, the list includes Audio-Technica, Mackie, Neumann, Sound Devices, and Tascam.
1. Neil A. Muncy, “Noise Susceptibility in Analog and Digital Signal Processing Systems,” presented at the 97th AES Convention of the Audio Engineering Society in San Francisco, CA, Nov. 1994.
Jim Brown is the (retired) principal consultant of Audio Systems Group, a small consulting firm specializing in the design of sound systems for public spaces - theaters, churches, stadiums, jazz clubs.
Update: Webinar Component Added To Live Sound Advice Instructional Seminar Coming Up This Wednesday
Virtually attend (free) instructional event this week focusing on how to sound check, properly set gain structures, EQ techniques, basic mixing, and more
The upcoming Live Sound Advice free instructional seminar on the art of the sound check—and much more—that’s slated for Wednesday, January 20 in Williamsport, MD will also be available as a live webinar. (Find out more about the webinar here.)
The live webinar component was added due to high demand for the seminar, which will have representatives from more than 50 churches in attendance.
The seminar/webinar will be led by Mike Sokol, a veteran audio educator who has worked as a live sound, recording, and design engineer for more than 45 years. Over that time, he’s run sound for thousands of worship, music, and political events. In addition, he’s also an adjunct professor at Shenandoah Conservatory in Winchester, VA.
The pre-webinar portion of the seminar will begin by 6:15 pm on Wednesday, with the actual presentation beginning at 6:30 pm. A live band will be on hand to enhance the hands-on training experience.
Topics to be covered include:
Mike Sokol with the DiGiCo SD21 console he’ll be utilizing in the seminar/webinar.
—How to sound check pastors and praise teams —Properly setting system gain structures —Equalization techniques —Basic mixing techniques —And more, followed by a Q&A session.
This free seminar/webinar is open to anyone working with live sound with a desire to learn the essentials, experience the latest gear, and expand their knowledge of sound system best practices.
Registration is still open for the seminar, which is being held at the Williamsport Banquet Hall, 2 Brandy Drive, Williamsport, MD 21795. Register here. And again, find out more about the webinar here.
Live Sound Advice is the free instructional blog and seminar series provided by Live Sound Co., a professional audio reinforcement, installation, education and event production company.
SynAudCon Presenting Training In Houston (Sound System Design) & Newark (SR For Technicians)
Registration is open for both three-day seminars providing a range of crucial information; approved for InfoComm RUs and BICSI CECs
SynAudCon has announced that registration is now open for two upcoming seminars, including Sound System Design in Houston (February 22-24) as well as Sound Reinforcement for Technicians (April 26-28) in Newark, NJ.
Lead by instructor Pat Brown, the three-day Sound System Design seminar in Houston will present a logical, intuitive, comprehensive approach to system design. Key topics covered include improving speech intelligibility, determining the best loudspeaker type for a given application, and matching amplifiers and loudspeakers.
Another goal is helping attendees determine, from the drawing board stage, that their system designs will work. Also covered is the use of computer room modeling and how it can speed up the design process.
Sound System Design will be held at the Fairfield Inn, located 2.5 miles from George Bush International Airport (IAH). The seminar is approved for 24 InfoComm RUs and 21 BICSI CECs. Registration and more information is available online here.
Also led by Brown, the three-day Sound Reinforcement for Technicians (SRT) in Newark will utilize a multi-media presentation that will instruct attendees on how to understand the audio signal chain, establish the proper gain structure, maximize the signal-to-noise ratio, and equalize the system.
During SRT, the measurement of voltage, impedance, polarity, SPL and STIPA will be demonstrated. More importantly, attendees will learn what these measurements mean and how to use them to ensure a system performs optimally. On the third day, Brown will demonstrate an optimized equalization process that brings the system to its fullest potential in the shortest possible time.
SRT will take place at the Kenilworth Inn, located 8 miles from Newark International Liberty Airport and about a half-hour from Manhattan. The seminar is approved for 24 InfoComm RUs, 21 BICSI CECs, and 10 CEDIA CEUs. Registration and more information is available online here.
ProSoundWeb presents at least two feature articles every day of the working week, meaning that there are 40-plus long-form articles highlighted each and every month.
That’s a lot. In fact, so much so that we got to thinking that it would be handy to present a round-up of the most-read articles for those who might have missed at least some of them the first time around.
What follows is the top 5 most-read articles on PSW for the month of December 2015. Note that since the articles aren’t all posted at the same time, we apply the same timeframe (length of time) for each when measuring total readership.
Also note that immediately following the top 5, PSW editor Keith Clark offers some additional suggestions of recently published articles worth checking out. These articles also scored quite well in terms of readership but were just outside the head of the list.
Without further adieu, here are the top 5 articles on PSW in December.
A primer on mixing console form and function; whether it’s digital or analog, many things remain the same. By Craig Leerman
2.Is Sound Subjective?
A wide-ranging discussion in the context of worship audio, including PA tuning, the goal of live music mixing, the role of politics and more. By Mike Sessler
Proper Loudspeaker Placement: How To Avoid Lobes and Nulls
Let’s say the sound system in the house of worship you’re working on goes into feedback whenever microphones pass under the loudspeaker array.
Worse yet, there are “soft spots” in some sections of the audience area.
Choir mics “squeal” before they are loud enough and the podium mic “rings” annoyingly for some presenters. You know that the system should be equalized to eliminate these problems.
So you install an equalizer and the feedback is reduced, but the soft spots persist and the system just doesn’t sound good.
But that’s why you, the consummate audio professional, are there.
After some careful listening tests, a “problem area” within the room is chosen for the measurement mic placement.
This is a place in the seating where people complain that they can’t hear, or a place where the mic consistently goes into feedback, such as directly under the loudspeaker array. The measurement looks something like that shown in Figure 1.
Figure 1: Comb filter caused by a time offset between two loudspeakers. The audibility of comb filters has always been the subject of heated debate. While humans may not be very sensitive to narrow notches in the spectrum, the spacial lobing implied by the comb filter can excessively excite rooms and dramatically reduce gain-before-feedback.
The response clearly shows an acoustic “comb filter” that results from a time offset between two sound arrivals at the measurement position.
The measurer first makes certain that the secondary arrival isn’t simply the result of a bad mic placement (floor bounce, etc.) or loudspeaker placement (ceiling or wall bounce, etc.).
After ruling out these two possibilities, it becomes apparent that the multiple arrivals are due to the overlapping patterns of two loudspeakers being used to provide audience coverage.
Standing at the mic position and simply looking at the array, noting that you are clearly within the coverage pattern of two loudspeakers suspended over the stage, confirms the suspicion. Sound travels at a single constant speed.
Yet, in this case, there are two loudspeakers.
Therefore every location in the room that is receiving direct sound at equal level from both loudspeakers (except for the center line where the distance to each loudspeaker is exactly equal), will receive two signals arriving at different times.
This time offset causes the comb filtering.
Figure 2: Represents the lobing (a form of destructive interference) between two spaced loudspeakers at a single frequency.
An acoustic comb filter can produce undesirable coloration of the sound and loss of definition. It can even change where the sound seems to be coming from, ruining the “imaging” of the system.
The possible “options” are: 1. Set the analyzer resolution to smooth the comb filtering, and then adjust the equalizer for the desired response. This is not a solution. It just masks the problem.
2. Ignore the comb filtering and simply “notch” the frequencies that are prone to feedback. Even though this is a common approach, it is treating the symptom and not the problem.
Excessive frequency notching can ruin the sound of the system. Why filter out sound that needs to be there?
3. Conclude that humans aren’t all that sensitive to narrow notches in the spectrum, so the comb filters are just something that we can live with.
This is rationalizing the problem and is simply not true. It’s usually the explanation provided by someone who is responsible for the problem in the first place!
4. Get out the old one-third octave real-time analyzer. You can’t see the comb filters on it.
For many years, audio professionals did not have high-resolution analyzers that could identify arrival time problems. The system response looked fine on a one-third octave analyzer, but it still sounded bad.
Today’s analyzers are vastly more powerful and can reveal much more about the nature of a sound problem.
5. Inform the owner that the current loudspeaker placement has created some problems that cannot be “corrected” electronically. The only real solution is to relocate the existing loudspeakers or redesign the array.
Figure 4: Placing a greater physical distance between the loudspeakers.
Unfortunately, the sad reality is that only the last option is likely to fix the system.
An acoustic comb filter is a symptom of a more significant problem. When two loudspeakers are placed in close proximity, the resultant distance offset will cause “lobing” in the speaker’s radiation pattern.
Lobes can be described as “fingers” of sound pressure “maximums” in the three-dimensional space surrounding the array.
The fingers are separated by nulls or axis of minimal sound pressure level. The fingers typically cause problems with microphones, since a mic is likely to feedback when it is placed within a lobe.
Figure 5: Use of aggressive pattern control to reduce the overlap.
The nulls cause problems for the audience, since parts of the audio spectrum that are critical for speech intelligibility (understanding the words) are cancelled at some listener’s seats.
When a series of these lobes and nulls exist, the visual representation of the frequency response at one listener position will resemble the teeth of a comb, with a sequence of peaks and valleys.
This is a far cry from the “perfect” system response that would look more like a flat line. As such, a comb filter is the symptom of a spatial problem that has resulted from a loudspeaker selection and placement choice.
To illustrate, look at the simulations shown below (Figure 6), which show such a condition performed with the EASE sound system design software package.
Two loudspeakers with low directivity control have been separated by two feet.
The resultant does not represent accurate sound reproduction and can cause the afore-mentioned problems with acoustic gain and speech intelligibility.
Please note that it is certainly possible to build quality “arrayable” loudspeakers, and there are a number of good examples in the marketplace.
However, all of them have several parameters in common:
1. Large physical size
2. Horn-loaded components
3. Aggressive pattern control to minimize interaction with adjacent loudspeakers
If these loudspeaker requirements present problems for a particular venue due to the required large physical size, then smaller loudspeakers can be used (usually in greater number) if they are placed sufficiently close to the listeners (i.e. exploded arrays or distributed systems).
Figure 6: Left - The balloon plot displays the 3-dimensional sound radiation from the two-device array described in the text. Right - The traditional horizontal polar plot views the equator of the balloon as viewed from above for one frequency.
Radio broadcast engineers have understood for years the importance of proper antenna array design to control lobing in RF radiation to steer their signal to certain areas within the listening range and away from others.
For instance, if a station is licensed to radiate 50 killowatts of power, they can use an antenna array to steer the radiated signal up and down an interstate highway rather than out across a sparsely inhabited area. In fact, if they do it wrong, they can be in violation of federal law and therefore subject to prosecution.
Loudspeaker array designers must work with the same physical laws and principles as antenna designers. The only difference is that they can’t be prosecuted for bad sound.
Balloon plots are useful because they show the three-dimensional radiation pattern from a loudspeaker or group of loudspeakers located at the center of the balloon.
The plot describes what is happening at a single frequency. The plots can be generated for multiple frequencies to more fully describe the performance of an array. The balloon plot of a “perfect” loudspeaker would be the same, regardless of frequency.
Comb filtering in the magnitude response (a measurement at a single point in space) is evidence of lobing in the spatial radiation of the array.
Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops online and around the world. For more information go to www.prosoundtraining.com.
The Young Soundman: “The Hardest Part Is Getting Paid!”
Dear Young Soundman,
I finally got my first gig to tech a real system, using giant line arrays in a multi-level performance hall for a one-off performance of some ’70s band I’ve never heard of. I flew out and met the rest of the crew, and was told that the main person to make happy was the front-of-house operator.
While this seems simple enough, he asked some basic questions like “How are you going to tune the room”, and so forth. And before I could even answer, he said, “I hope you’re going to use more than just an RTA, and not do that stupid Haas thing. My ears know the difference and I want my delays set with precision.” (Or something like that.)
I was taught to measure and then add delay to create localization. Is this not the way to go? I didn’t do what he asked and ended up in a huge debate.
After things calmed down, we got through the gig. Did I do the right thing? What would you have done? I just finished school and want to be the best! But already I’m confused as to what to do. No one warned me of the dangers of old soundmen.
The Newbie —————————————————————————————————————————————————
Congrats on getting a paying gig. Did you have to bribe someone to get the job? Meanwhile there are experienced techs starving and you just took their gig. Good job!
You have several problems. First, whatever “sound god” you begged to teach you didn’t tell you (or didn’t know how) to play politics.
Your job was simple: Make the house engineer happy. Immediately you failed by starting an argument. Good job!
It’s a good thing you don’t want another gig… ever!
No one likes it when fresh virgin blood shows up and ticks off the old dog. (Even if Old Yeller should have been put out of his mixing misery years ago.)
Big deal: he’s wrong and you’re right. I hope you’re happy walking along the beach sipping Corona knowing that you were right even if it cost your next job. Good job!
Next time keep your mouth shut, watch some CNN, and learn about politics. When facing battles like this one, have someone else fight them for you. Even when you lose (and you will), no one knows about it, and you live to mix another day.
At the end of the day, it’s not your system, it’s his. He should have it tuned any damn way he wants. If he wants subs on an aux, give him the best aux fed subs ever. If he wants to “smiley face” the EQ, fine, let him. It’s your responsibility to set the system up so that it is capable of doing this effectively, in order to give him the best/worst sounding smiley face ever.
You lost the argument before it even started. Regardless of who “won” or who was “right”, it’s a stupid argument. Good job!
And while you were arguing, you missed an opportunity to sneak your friends in through the back door.
What would I do, you ask? Great question, except I never give away my secrets lest you take my gig as well. I will say that I don’t prescribe either practice you noted, and my delays sound just fine.
I’m glad you got a paying gig. Now you get to find out what the hardest part about live sound really is - getting paid!
The Young Soundman
If you like The Young Soundman, be sure to check out the views and rants of the The Old Soundman.
RCF Audio Academy Offers Sound Engineering University Course
Learning center for electro-acoustics has offered continuous education opportunities for students and professionals since 2009.
For the fifth year in a row, RCF Audio Academy introduces the Sound Engineering university course.
RCF Audio Academy is a high-level learning center for electro-acoustics, offering continuous education opportunities for students and professionals since 2009.
Download the brochure about the Sound Engineering university course – 2016 edition and listen to the previous participants’ opinions in the video below.
The course is organized by the Department of Science and Methods for Engineering of the University of Modena and Reggio Emilia, in cooperation with RCF SpA, an international manufacturer of professional sound products and systems.
The primary goal is to provide engineers with a specific knowledge of the state-of-the-art technologies related to sound, transducers and speaker systems and their applications in the professional market.
The course is organized in modules taking the participants through a learning process from the basis of electro-acoustics components like transducers, amplifiers and digital processors till the design of a complete audio project based on a real case study. An international reference project carried out by RCF engineers will be used as a studying guideline to be followed during the course.
The learning topics are grouped into four modules as follows:
1) Environmental Acoustics in open air and inner spaces including design and simulation methods.
2) Transducers and speaker systems including clusters, arrays and point source configurations.
3) Electronics including circuit analysis, filters, amplifiers and DSP.
4) Sound systems design including practical experiences in measurements, listening tests and tuning.
The technical contents provided in this University course will be completed by the direct experience of RCF laboratories, since 1949 involved in the design, manufacturing and market applications of professional products and systems for theatres, concerts, auditoriums, airports, cruise ships, shopping malls, cinemas, stadia.
Church Sound: Getting A Great Worship Sound Experience
All of us love to have a consistently great audio experience during praise and worship as well as the sermon.
But how do we get there?
To begin, let us understand the basic block diagram of a typical sound reinforcement system as shown in Figure 1 below.
For a typical sound reinforcement system, the singers and musical instruments (in orange box) are the sound sources that are usually picked up by microphones. For some musical instruments, DI (direct injection) boxes are employed.
These signals are then channeled to a FOH (front of house) console/mixer. The mix engineer then mixes these inputs and mix is then fed to the loudspeaker processors, which in turn feed the various power amplifiers which drives the loudspeaker systems.
For a properly set up sound system, the congregation is thus able to hear the amplified sound well even at the furthest seats.
Figure 1: A simplified block diagram of a typical church sound reinforcement system
In more elaborate set up, to provide customized mix for each singers and musicians, a parallel feed is channeled to a separate monitor mixer where this monitor mix is performed by a dedicated monitor mix engineer.
For a multi-site church setup, a simplified signal flow diagram incorporating simplified sound reinforcement block diagram is as shown in Figure 2.
Figure 2: A simplified block diagram of the sound reinforcement system for the main venue (highlighted in light yellow section) with simplified signal flow diagram to multi-site venues (highlighted in green section)
In some cases, the audio feed to the various multi-site venues may come from a separate audio mixer with a dedicated mix engineer located at a separate studio control room where the sound mixing is done based on the studio monitor loudspeakers. This allows the FOH mix engineer to concentrate solely on the mix for the main venue and not worry about the mix for multi-site venues.
The audio mix for the multi-site venue together with the video mix is then fed to fibre encoders which can be linked via commercial fibre network to the various multi-site venues for eventual decoding and feeding into the respective sound systems on-site.
Based on the above sound system block diagram and signal flow, the next step is to map out what are the successful ingredients for a great audio experience. Figure 3 below broadly paints the critical factors to consider for a great sound experience for the congregation:
Figure 3: Critical factors for obtaining a great sound experience
Before jumping into sound mixing and other sound equipment, it is important to have excellent sound sources to begin with. Vocals from worship leader(s) and musical instruments create these original sources of sound.
The various key components contributing to great sound sources are as follows:
1. Anointed and Inspired Songs
Gifted Gospel songwriters penned down lyrics which are scripture based and these words can minister to the hearts and souls of the congregation. When these lyrics are combined with appropriate and inspired tunes, they become anointed spiritual songs that can have lasting profound effects on the worshippers.
2. Voice Quality
It may not take great effort to recognize singers who are blessed with marvellous voices. This is dependent on the anatomy of the laryngeal (voice box) and this is the part which is attributed to natural gifting.
3. Vocal Skills
Great natural voice quality is a good starting point. There is also the learned portion that contributes to the voice quality by applying appropriate control of the laryngeal muscles. Good auditory skill is also required as feedback to the brain for the proper control of the various voice box muscles to stay in tune and other body muscles to provide the necessary expression required by the songs. Skillful singers can sing in parts e.g. soprano, alto, tenor and bass without getting confused and go out of tune from hearing the voices of other singers.
4. Musical Instrument Sound Quality
Great sounding musical instruments are a good accompaniment to excellent voices. Otherwise, it would be an acapella (singing without musical instrument accompaniment) all the time. Musicians can be quite particular in choosing their musical instrument as they seek after a particular sound and feel. It is no surprise then that great sounding and highly sought after musical instruments command a premium price.
5. Musicians’ Skills
It may not be difficult to note that it takes great skills to play a musical instrument proficiently. Divine gifting and years of dedicated training on a particular musical instrument plays a critical role in establishing a good foundation. Other than mastering their own musical instruments, musicians must also be able to play harmoniously as a band or an ensemble under the guidance of a music director before they can provide successful accompaniment to the singers.
6. Music Arrangement
Music arrangement dictates how each musical instrument should be played and how the vocals should take their parts. Contemporary music arrangement typically consists of the following key components:
Melody: This carries the tune of the music. For vocals, it is normally taken by the worship leaders. As for the musical instruments, at times certain instrument, e.g., an electric guitarist, a keyboardist or a violinist may take the lead role in playing a few bars.
Harmony: This harmonizes with the melody. Suitable chords at appropriate times harmonize the melody very nicely. For the vocals, the backup singers may choose to sing in parts. For the musical instruments, playing notes within suitable chords can harmonize pleasantly.
Fill: Melody only occupies certain time slots within the time frame. If all the vocals and instruments are only occupying the same time melody time slots, it can sound quite empty since there are considerable gaps (i.e. no sound) in between the melody. This is where the “fill” comes in. Vocals can fill the gaps by singing in between melody. Likewise, musical instruments playing the fill part including sustained notes can make the entire music sounds much more wholesome.
Rhythm: This provides the beat and thus the tempo of the overall music. This part is normally taken by a suitable percussion instrument such as the drums and is usually accompanied by the bass guitar.
The importance of recognizing these key components of music arrangement would become more apparent in the later portion of this article where the role of the mix engineer is being expounded. In summary, in a praise and worship setting, the above six different elements directly influencing the quality of the sound source are typically the responsibilities of the worship team.
The sound engineer needs to mike or otherwise connects these sound sources, optimally operate the sound mixer with careful technical considerations and then artistically blend all these inputs to a wholesome nice sounding production and channel to sound reinforcement systems at various venues. Mixing on the sound mixer board is just one of the many skills required for a good worship experience.
The other skills involved are described in the following paragraphs:
7. Miking Techniques
To convert the sound sources to electrical signals that can be utilized by the sound mixer, one common way is to place suitable microphones near the sound sources.
Each musical instrument presents a unique challenge and thus requires different types of microphones and placements to obtain the desired sound.
Microphones are typically placed at the “sweet spots” of the musical instruments and angled away from other musical instruments to obtain a cleaner sound pick up and also angled away from loudspeakers to provide a better gain before feedback.
As for vocalists, handheld microphones are usually employed. It also takes some skills for the vocalists to handle the microphones appropriately without causing a nightmare to the sound mix engineer.
Since the intent of this article is to just present an overview, it does not dwell into the details of miking techniques for the various musical instruments and for vocalists. More details can be obtained from reputable microphone manufacturers and from the experiences of seasoned sound engineers.
In some cases, the electrical outputs of sound sources are readily available such as synthesizer/keyboard instruments and electric and bass guitars. These may be connected to the sound mixer board often via DI (direct injection) boxes which has the ground lift feature to help eliminate humming caused by ground loops.
For electric and bass guitars, in addition to miking the respective guitar amplifiers, getting the sound via DI boxes provides alternative paths for the mix engineer to combine both sources to obtain the desired sound for that particular musical instrument.
In the pursuit of a great sound, the quality of the sound sources and miking techniques is so critical that whatever sound quality is being compromised up front, it cannot be compensated downstream in the process.
8. FOH (Front of House)/Multi-site Sound Mixing
With proper miking and electrical connections made, all the sound sources are now readily available in suitable electrical format. The next challenge is to mix all these signals appropriately. The sound mix engineer has a daunting task to take all these individual sound sources and recombine them in a fashion that is in line with the music director’s vision.
Just how challenging this task can be? In a more sophisticated set up, there can be easily more than 50 input channels to mix. In addition, there are numerous parameters per channel to get the desired sound. Even an entry level digital mixer would typically consist of the following parameters:
• Signal level related: gain, fader, pan i.e. sub-total of 3 parameters
• Tone controls: 4 parametric equalizer bands each consisting of 3 parameters (gain, frequency and Q) i.e. sub-total of 12 parameters
• Dynamics: threshold, compression ratio, attack time, release time, knee, gain i.e. a sub-total of 6 parameters
• Effects: Depending on the types of effects, typically there are few parameters involved.
• Switches: Phantom power, mute, EQ on/off, dynamic on/off, effects on/off
Hence, there could be around 25 parameters and a few switches per channel and multiply that by the number of channels e.g. 50, there are over 1,000 parameters to get fairly close to the optimum values in order to obtain a decent mix.
In addition, the mix engineer needs to recognize the key components (melody, harmony, fill and rhythm) of the musical arrangement as the musical instruments and singers may change role during the song. This knowledge would allow the appropriate prominence placement of each sound source in the mix. Also the different fader levels and equalization applied to each channel to obtain the desired level and tonal property may need to be tweaked along the way. Mixing is so dynamic that constant adjustments are often required to keep in line with the music emotions.
Optimizing the compression settings is also not an easy task. Other than obtaining the right sound, there is a fine balance between having acceptable level variation for that channel and yet able to provide the required intended dynamics for that channel to sound musical.
Mix engineers need to have a unique blend of good technical knowledge and artistic skill. Examples of technical skills involved include understanding signal to noise ratio for optimizing the gain structure and having a good foundation of time and frequency domain concept for proper application of the parametric equalizer and compression parameters in the sound mixer board.
In addition to technical skills, the mix engineer also needs to possess a good level of artistic skills to know how each sound source should sound like and how to blend these individual sound sources together in a wholesome manner appropriate to the music genre.
9. Monitor Mixing
Each musician does not just play their instruments on stage alone. They need to hear the other musical instruments and vocals in order to harmoniously blend in their instruments with the whole band. The same applies to the singers. They would need to hear themselves, other singers and musical instruments.
Each musician and singer may require their own unique mix to perform optimally. Therefore, in a more elaborate set up, there is a separate sound mixer board along with a dedicated monitor mix engineer to provide customized mix for each musician and singer. This would ensure that the monitoring needs of each musician and singer are well taken care of. To facilitate communication and other stage supports, the location of this monitor mixer is usually near the stage.
10. Other Stage Supports
Stage monitor loudspeakers can be used to feed the required mix to the musicians and singers.
In a modern band with drums and electric guitars, the stage can get pretty loud.
For the stage monitor loudspeakers to be heard, the volume has to be cranked up. But this can cause undesirable sound spill over into microphones, non-targeted musicians and singers and to the audience.
Hence, it is common for musicians and singers to adopt the use of IEMs (in-ear monitors) to resolve this sound spill over issue. For more freedom to roam on stage, wireless IEMs are often utilized.
In the same manner, wireless microphones allow singers and some musicians freedom to roam around on stage.
All these wireless devices require close support in ensuring that the correct device is given to the right musician and singer, the devices are appropriately positioned, and batteries are new or fully recharged before the start of the worship.
In addition, due technical consideration must be given to operation involving more than one wireless transmitting device to avoid radio frequency interference caused by intermodulation distortion especially when transmitting devices are operating in close proximity of each other.
Hence, the transmitting frequencies from wireless devices are usually carefully chosen to avoid such interference from intermodulation as well as from known radio frequency sources such as television stations or other wireless devices. Prior complex frequency allocation computation and dedicated radio frequency spectrum analyzer may be employed to monitor for interference free channels for trouble-free transmission of wireless devices.
Also, there is maintenance work such as sanitization of the microphones and IEMs. Visual inspection, continuity and contact checks for cables and connectors are also required to keep all the sound equipment in tip top condition.
All these considerations and arrangements involved beyond just good technical skills in dealing with sound equipment but also good inter-personal skills especially with the musicians and singers since the main function is to support these people who are on stage.
The sound mixer typically output in left-right or for some cases in left-center-right configuration. There may be auxiliary outputs to feed to other locations such as rest rooms and waiting areas. For auxiliary-fed subwoofer configuration, sound sources which have low frequency contents such as the kick, toms, bass guitar or CD sources are fed to a dedicated mixer auxiliary channel where this output is then used to feed the subwoofers.
All these left, center, right outputs as well as the auxiliary outputs are then fed to loudspeaker processors which then in turn feed the power amplifiers which drive the various loudspeakers. The next section describes the importance of the sound reinforcement system design, optimization and operation. Also due considerations must be given to room acoustics and the quality of the sound equipment especially for the loudspeakers.
11. Sound System Design
Before the loudspeakers and other sound equipment are being installed, the sound system must first be designed. The design process usually begins with the understanding of the overall sound system requirements. The next step involves the choice of appropriate loudspeaker models as well as the positioning and aiming of these loudspeakers.
Reputable pro audio loudspeaker manufacturers allow sound system designers access to computer simulation software that can aid in this design process. Refer to Figure 4 for an example of a computer simulation of loudspeaker line arrays.
Figure 4: Example of a computer simulation of loudspeaker line array performance parameters.
The user can input the venue layout in 3 dimensions, choose the appropriate loudspeaker models and then position and aim the loudspeakers to see the overall sound pressure level distribution and frequency response at various locations. Since this is a computer simulation, it allows the user to try as many loudspeaker positions and aiming angles and gain settings as required to determine if the intended design goals can be first obtained before any physical installation takes place.
Before even attempting on the simulation software, the sound system designer must be able to lay down the desired sound system performance characteristics such as targeted maximum SPL (sound pressure level), maximum allowable SPL spread between the front and rear seats and frequency response target.
A good technical knowledge of the applications of loudspeakers including the characteristics and limitations of point source loudspeakers and line arrays is also required.
Since no sound system is perfect, trade-offs between various performance parameters are inevitable and hence experience and strong technical background are required to perform these trade-offs.
It is advantageous for the seating arrangement to be planned together with the sound system design since there will always be non-optimal sound quality areas. This may include areas which have “phasing” challenges for example locations directly in between similar loudspeaker clusters.
For this case, it would be best to have aisles instead of seats for such locations.
In like manner, when video projection screens are being utilized, both seating arrangement and the loudspeaker positions have to be considered together. There should be acceptable view angles from the seats to the projection screens. The positions of the loudspeakers are carefully selected to avoid excessive localization errors between the visual source and sound source.
For best performance, it is essential to consider the sound system, video projection system and seat arrangements altogether as a system during the design stage to enable appropriate trade-offs to be made up front.
12. Sound Equipment Quality
During the design stage, the quality of the sound equipment must also be taken in account. Since the failure of a single piece of equipment may very well fail the entire sound system, the reliability of each piece of equipment and the quality of the service support must be considered.
As for the sound quality aspect, reputable pro-audio sound equipment manufacturers are usually able to design pretty decent sounding electronic equipment such as sound mixers, loudspeaker processors and power amplifiers. These days, the quality of the electronics are usually quite decent especially with the use of high sampling rate and high resolution analog-digital converters and well established digital processing for the loudspeaker processors.
As for power amplifiers, the electronic designs to achieve good sound have been well established over the years. The main challenge for accurate sound reproduction lies in the conversion of electrical energy to sound energy. As such, the main distinction of the overall sound quality among various manufacturers is usually highly dependent on the sound quality of the loudspeaker system.
Therefore, there is an obvious desire to keep up to date in the pro audio world to constantly look out for reputable manufacturers who are able to design nice sounding loudspeakers and the associated electronics such loudspeaker processors and power amplifiers which are sometimes also built into the loudspeakers.
13. Sound System Optimization
Having a good sound system design with good quality sound equipment is not the end point. The next stage involves optimizing the sound system. This typically involves adjusting the parameters of the loudspeaker processors for optimum performance. These parameters are usually related to equalization settings, loudspeaker cross-over parameters and delay settings.
Appropriate objective measurement technique such as transfer function measurement based on dual-channel FFT (Fast Fourier Transform) or TDS (Time Delay Spectrometry) is often used as a starting point to aid in sound system optimization with subjective auditory confirmation that the settings chosen is acceptable.
The measurement also requires supporting acoustical measurement equipment including measurement microphone, sound card and computer software. A screen capture of such a measurement tool is shown in Figure 5.
Figure 5: A screen capture of a dual channel FFT acoustic software (SmaartLive 7) that performs the frequency response measurement of a loudspeaker.
The optimization process often requires careful trade-offs between one system performance parameters with other performance parameters.
Typical areas of consideration includes loudspeaker coverage area, difference in SPL (sound pressure level) between the loudest and quietest seats, absence of hot spots and elimination of dominant resonances and loudspeaker overall frequency and phase responses.
This calls for good detailed technical understanding on how loudspeakers interact with each other, room acoustics, measurement techniques, engineering judgement if measurement results are valid and also final auditory confirmation on whether the optimization process has reached its goals.
Most modern loudspeakers from reputable manufacturers are quite decent sounding if the application is correct and when properly optimized.
However, probably due to limited in-depth technical expertise in this very specialized area, not all sound systems are well optimized to reap the maximum benefits.
14. Room Acoustics
The room acoustics play a large part in affecting how the loudspeaker system would sound like in a room. The best sound equipment can sound mediocre at best when housed in an acoustic environment that is not appropriate for the worship style. For contemporary worship and for clean clear sermon, the room should have appropriate acoustical treatment.
An acoustician is required to design an appropriate acoustic treatment of the room during the design stage of the building. Room modes are highly dependent on the interior dimensions of the room. For these reasons, it is highly desirable for architects to incorporate the recommendations from acoustician into the building plan and not as an after-thought when the building project is completed.
The location and the types of acoustic treatment and construction details for the walls, ceilings and floors, whether the intent of the acoustic treatment is to be absorptive or diffusive or a combination of both are specified by the acoustician after the overall functional requirements of the room is understood.
15. Sound System Operation
With the sound system properly designed and set up, the next challenge is to maintain the sound quality consistency over time. This calls for identification of potential problems that may occur and the lessons learned from historical issues.
Based on these inputs, a comprehensive sound check list can be formulated to minimize the occurrence of repeated problems. The sound check procedure will then call for the execution of these sound checks. Sound checks are especially important for multi-site non-dedicated worship rental venues where the sound system is set up by a third party and subjected to frequency tear down and re-installation.
Another consideration for these multi-site venues is that the sound mix engineer is not located in the same venue as the congregation. Therefore, it is desirable for the sound system to have some basic capability for adjustment of volume and tone by sound duty persons at these multi-site venues. This is to ensure that the loudness and tonal quality is appropriate throughout the worship service since the mix engineer is unable to hear and feel the result of the mix at these remote locations.
The volume and tonal adjustments are usually minor if all the other factors (Points 1 to 14) are already well established. Nevertheless, it is important for sound techs to hear what the congregation is experiencing and to provide real time minor level and tone adjustment to avoid long response time associated with voice or text communications back to the mix engineer. The sound techs at these overflow sites can also provide useful and constructive feedback to the mix engineer on the mix itself.
In some venues, the on-site sound equipment is located away from the congregation. In a cinema situation, the audio equipment is usually housed in the projection room which is located away from the congregation. In such situation, wireless mixing capability can be considered. Please refer to Figure 6 for an example.
To fulfill the above roles effectively, the sound team must be well trained to acquire the required skill set. This calls for a comprehensive training program especially when sound volunteers are involved.
Fig 6: In situation whereby the sound equipment is located away from congregation seating area, digital mixer which supports iPad App via Wi-Fi router can be utilized for remote adjustment of sound mixer parameters. This arrangement allows sound duty persons to be located with the congregation to hear what the congregation is hearing.
Achieving a great sound experience is a highly complex process that requires each and every one of the above input factors to be in top form at all times since just one weak link can break the chain.
These functions though usually support by a small number of persons, plays a critical role in ensuring that the worship team and the preacher is able to reach to every congregation members.
It is also hopeful that this article is beneficial in the following areas:
1. The information given above provides an overview of the entire sound system to the sound duty persons located at various locations.
2. It serves to inform the sound techs that although each person may only seem to be playing a small role, that each role is absolutely essential in order that the entire system is functioning well as intended.
3. Each role is specialized in its own field, has its own unique challenges and there is so much more to learn and grow in this sound ministry.
4. It identifies the dominant factors required to achieve a great sound experience.
5. This can be helpful as the starting point to help identify the areas for assessment and monitoring and where necessary, improvements can be made to achieve a great sound experience for the congregation.
Other than being skilful in the required roles, it is important that each member understands the Biblical principles and serve with the right dedication and attitude. Each member of the sound team should thus render others better than himself/herself (Phil 2:3) while working together harmoniously as different members of a body (Roms 12:4) for the common goal to serve the congregation in the best possible manner for God’s glory.
The mission of the sound team is to support the worship team and the pastor in the preaching of the Gospel to the congregation. Missing this critical function could very well mean this Good News is unable to reach effectively to each and every congregation member.
Although the sound ministry is usually so few in numbers, it runs so silently in the background that often goes unnoticed unless something goes very wrong. It may also not be readily apparent that the sound ministry has such a profound impact on the lives of the people attending the church worship services that in reality, it is a life-changing ministry.
KL Gan had been with Seagate Singapore Design Centre for over 28 years and has recently retired from his role as an executive director in research and development department. He continues to serve as a sound volunteer with New Creation Church in Singapore.
I often hear stories about church leaders who won’t let the sound team “turn any knobs” on the mixing console.
The leaders expect a great music mix to happen every week simply because it previously sounded great one time. Or perhaps there was one lucky service where none of the mics had any feedback.
The assumption is that if there were no problems last week, there shouldn’t be any problems this week as long as the sound team doesn’t turn any knobs or push any buttons.
I’ve also heard stories about sound-system installers who “set the board once” and tell the sound team they’re only allowed to move the faders but not to touch any of the “special knobs” on the top half of the mixer.
You know the knobs I’m talking about: those mysterious EQ and AUX controls that if turned incorrectly can cause all sorts of feedback and assorted sonic mischief.
But I’m here to tell you that to make a great mix and to turn yourself into a great mixer you need a few failures.
Failures are good for you and as I often tell my teenage boys, you learn way more from failure than you do from success.
How so? Let me illustrate with a few personal examples.
When I started my first real job as a robotics designer I held the dubious position of being the youngest design engineer in the history of the company.
I was fresh out of school and had already been designing my own gadgets from mini-bikes to rockets from the time I was ten years old.
And while I entered the job with a great deal of confidence, I was soon in over my head. Because instead of just building a wacky gadget for myself in my basement, now I was doing it with someone else’s money.
And it was not only the money; I had a team of mechanics and electricians who would build anything I drew up on a blueprint. This was both a dream come true and the biggest terror of my design life.
What if I designed something that was a failure? Then everyone would know that I had made a mistake. Would I lose my job if I failed at a design? Talk about second guessing myself….
Now these weren’t misplaced fears. I was designing packaging robotics that would go on an assembly line manned by dozens of shift workers.
And each assembly line would produce upwards of 10,000 items per shift with two shifts a day.
So if one of my designs failed after being installed it could cost the company literally thousands of dollars per hour while the production line was shut down to fix whatever I had goofed up.
Talk about more than a little pressure for the new “wunderkind” in engineering.
So as I was designing a new way to automatically stack boxes prior to feeding them into a shrink wrap tunnel, my boss asked me why I seemed so worried about this particular project.
I replied that it would be the first project I had designed that was totally mine from start to finish, and I was worried about failure.
The boss then asked me to take a walk back to the “North Wall” to look for parts to build a prototype of my new machine.
Now the North Wall was so named because it was a parts graveyard behind the north railroad dock where all the old machines were dumped.
As we picked through the various dead machines looking for a particular relay or air cylinder, he told me the stories behind each part.
This one had just worn out, he explained, or this had been used for a product we no longer made, and this one was from some failed experiment.
And as he showed me more and more “failures,” many of which were his own, he kindly explained to me that I too would have some failures on the North Wall.
Failures, I thought… here I was terrified of a design mistake that would cause my design career to come crashing down and here he was already expecting me to fail. But then he told me the real secret to doing great things.
He said that if I didn’t have any failed experiments on the dock, then I wasn’t trying hard enough. That was an interesting perspective.
If I took the easy way out on a project, then while I might not fail, there certainly wouldn’t be any new ideas or great accomplishments. Hmm…. that made sense.
I did finally go on to design a unique box stacker that the company later patented, but that’s another story.
Go Ahead, Make a Mistake
More recently, I had a young audio apprentice who was hugely talented, but just hadn’t made enough mistakes yet to be great.
So when he asked why I was so quick at finding sound system failures and what he could do to do get better, I told him he just needed to fail more.
Yes, it still sounded counterintuitive, but what makes each of us an effective troubleshooter or audio mixer is the memory of past failures.
Sort of like the first time you stuck your finger in a light socket you learned that getting shocked was to be avoided, or stepping on the gas while driving your car on black ice could put you on a Nantucket Sleigh Ride through a field.
I’m sure you remember that lesson, don’t you?
Just like when I was pestering my 17 year old son to double check the traffic while pulling out of our driveway, but it wasn’t until he pulled out in front of a big truck that screeched to a stop six inches from his driver’s door that he understood what I was talking about.
White as a sheet hours later, he told me that he now understood what I was bugging him about. But it never made an impression until he made that mistake himself. I’m just glad he didn’t get T-boned.
Take It To The Limit
This is all to say that the next time you’re running the mixing board at your worship service and you’re afraid to try something a little over the top for fear of failing, go ahead and push the envelope a bit.
And if you fail, then learn from that mistake and move on.
But if you succeed, then you’ve made one more step towards being an effective sound technician, and moved one step closer to the best of all mixing situations — a church service where absolutely nobody complains about the mix.
Yes, you’re working towards a null-zero situation, which seems counterintuitive, but is, in fact, the primary goal.
Getting everything to work without feedback and so everyone can hear is the first step. Only after that can you reach for the next level of inspirational mixing and that’s only achievable after you’ve made a lot of mistakes and failed a few times.
Go Big or Go Home
Therefore, go ahead and plan for failure, but try for greatness. Show up at practice with the praise team and experiment with the equalization or monitor sends.
If it all works, then that’s great. However, if it doesn’t work, that’s even better. Now you’ll know the limitations of the sound system and what to avoid.
You’ve felt failure and know that you can survive. That’s the only way we grow both as individuals and technicians.
Mike Sokol is lead trainer for Live Sound Co. in Maryland, and lead writer of the Live Sound Advice blog. He’s a veteran audio educator and is also an adjunct professor at Shenandoah Conservatory in Winchester, VA.
HOW-TO Sound Workshops Demonstrate “How To” With DiGiCo SD9
DiGiCo partners with Mike Sokol and Hector La Torre of Fits & Starts Productions to help train churches and schools
For more than 15 years, Fits & Starts Productions’ HOW-TO Sound Workshops have been traveling around the United States providing pro audio training sessions for houses of worship and schools/colleges.
Visiting approximately 36 cities each year, chief instructor Mike Sokol and the F&S team provide a comprehensive, full-day, hands-on opportunity to learn the essentials of setting up and running a sound system, and one of the primary tools they rely on is DiGiCo’s SD9 mixing console.
“We first took the SD9 out on the HOW-TO tours more than three years ago, and it’s been a prominent part of our setup since then,” says Fits & Starts managing partner Hector La Torre, who notes that the console serves as the workshop’s featured instructional digital mixer.
Aside from carrying the main PA for the instructor, the SD9 is also is used to run music and signal so that attendees can see and hear how the console works and just how a digital console provides flexibility and overall cost-saving.
When asked why the DiGiCo desk was selected again this year, La Torre responded, “The SD9 showed us its excellent capabilities on a couple of our previous tours, so it was a simple choice this time around. The board has operated flawlessly despite being banged around in our road truck for months, so why take a chance on another console that might fail? I don’t want to see 40-50 frowning faces at any of my workshops. DiGiCo’s equipment is robust and cost-effective for the primary markets we serve, plus the manufacturer has always been sensitive to the market’s needs and delivered solid support. I have nothing but praise for them.
“We also used this console again to let attendees know a smaller frame size and lower price range mixer, the S21, was going to be hitting the market soon. Folks could see that DiGiCo offered options for different budgets while maintaining high quality.”
He quickly adds that attendees have loved getting their hands on the SD9, and that several universities and churches have even chosen to purchase their own based on their positive experience with the product at these sessions.
“While we work mostly as hands-on sound trainers we also do system design and organizational consulting,” La Torre adds. “Many, many churches and schools often don’t know what they need in terms of audio gear, and consequently make terrible, costly choices. We help them clear that hurdle.”
“Traveling with the SD9 has been a real pleasure,” shares HOW-TO’s chief instructor Mike Sokol.
“It’s powerful enough to mix the largest church venue I was in, but small enough to fit anywhere. And every sound technician, from the newest entry level to the seasoned veteran, was impressed by the SD9 workflow as well as the sound. It’s a real class act. This is going to be hard to top with their new S21, but I’m confident that DiGiCo has that one dialed in as well.”
Fits & Starts has served as the nation’s leading provider of professional audio training workshops for the worship and education markets for over 15 years. For details on the company and its schedule, visit the link below.
When we discuss the ways that AV can become more IT, it is important to emphasize that the transition to becoming more IT is not one that should take AV away from its core competencies.
I have had a lot of discussions lately with some of the best minds in AV and we seem to all agree that we (AV) need to continue to promote our value. This is to say that we all agreed that AV can do many things that no other industry can do.
AV rules the physical space. Nothing happens in the physical space (sight or sound) without AV working some form of magic (that magic being the mixture of art and science). When AV adds IT for the purpose of monitoring, managing, sending content, collaborating, conferencing, controlling and much more, the business opportunity is huge and the problem-solving we can do for the customer is immense.
That is where the focus should be — adding IT to AV. But how?
1. Partnering And Connecting
During my training sessions that last two or more days, I also teach juggling. I do this especially when I am teaching an AV/IT class. I tie in the analogy that in an AV/IT class there are three topics being covered; audio, video and IT. In the analogy those topics represent the three juggling balls.
I then have these students partner up in twos. When one partner throws a ball, the other catches. The pairs can easily jump to two juggling balls. This does two things: it allows the thrower the ability to focus on their throw without worrying about catching and this allows the catching partner to catch without worrying about making an accurate throw.
The teamwork being used in this exercise helps me to illustrate the value in partnering in AV/IT. The exercise also illustrates that when you are partnering you can focus on your core competency and trust that your partner can focus on theirs (one is throwing and the other is catching).
We also discuss that partnering mitigates risks. While there is shared risk of course there is shared return so the major goal should be to seek out more customers together than either partner had alone to make sure the partnership yields new business.
The biggest lesson I use the juggling analogy to teach is this: when you are an expert in AV (two of the three) and you can juggle those two really well, but you pick up that third juggling ball and you drop them all, you have failed at juggling (in our case AV/IT integration). Even if the AV part of the job is perfect, if you picked up the third ball and dropped it, you failed at all three in the customer’s eyes.
Partnering allows you to juggle in front of the customer with low risk. A good partnership will have a way for both parties to continue to grow throughout the relationship and not feel threatened by the other’s ability to grow.
Connecting is also important—joining industry associations and social groups in IT will help an AV company stay informed of technologies and trends. This will also help with networking (people networking) and possible sources for candidates, contractors and technical resources.
2. Physician, Heal Thyself
It is nearly impossible to sell, support and promote something you don’t believe in. If you believe in something you need to implement it. As AV/IT integrators our networks need to be solid and they need to be able to support unified communications and collaboration (UC&C).
There is a saying that states the Cobbler’s kids go barefoot. Our industry is not an exception to this saying. I often find that our demo facilities are lacking in capabilities or esthetics. In the case of AV/IT the demo facilities need to be solid. IT people expect an integrator to be able to prove that they can do for themselves what they propose to do for the customer. Make sure your IT system supports what you expect to sell, support and promote.
3. Set A Plan And Take It Step-By-Step
AV adding IT is something that needs to have methodical and planned approach. I have seen several companies add IT to their business model and the ones that do it well do it with a business plan.
It sounds so simple, but you would be amazed at how often I have seen the opposite approach. I’ve seen AV companies that add IT as a second thought. They simply add a few products to their mix and throw some additional responsibilities on their internal IT guy to support a few customers and wonder why they are not growing their IT business.
Adding IT requires adding appropriate resources and deploying those resources. Those resources may come from a partnership, but there is need for additional resources nonetheless.
The other point I’d like to make is that it is just as important to communicate the plan to everyone involved as it is to have a plan in the first place. Everyone involved should have a good understanding of what the plan is and what every step of the plan looks like. Over-communication is infinitely better than none.
4. Hire The Right Employees ... And Customers
One of the biggest challenges in adding IT to AV is ensuring you have a team that is capable and supportive of this new endeavor. At times, getting this team assembled requires hiring new employees.
In AV we tend to “fill vacant positions” rather than looking at where our business will be in a year or three years down the road and determining what our needs will be then. We tend to try to hire replacements with the same skill sets as previous employees. When adding IT to AV we need new skill sets and with that we need to look in new directions.
Start your job descriptions from scratch. Write the job description as if you were doing a needs analysis on your company. You will often find if you start from scratch rather than using an existing job description you will come up with something completely different. Sometimes change is good.
What about your customers? Are your customers asking for IT in their AV? If not, you may need to hire some new customers. Your customers may be the ones who are stagnant and if that’s the case you may be missing out on a lot of new business. Sometimes you need to look at business trends and wonder why your customers are not following them. You may need to hire some new customers, and possibly even fire some others.
5. Training And Certifications
You are not always going to hire new employees to embark on new endeavors (or new customers for that matter). So you will also need to grow the ones you have. Training is the best way to do this.
I completely believe in this. I have trained people most of my career. It is the best way to get loyal team members to grow and stay happy. If you have people with the aptitude and attitude, you will find that they are often worth their weight in gold. Making a training investment in them is well worth it. These types of employees are willing to grow with you and they will take your organization to new heights. The risk of not training your people is far greater than training them and having them leave.
Certifications in IT will gain your organization a foot in the door. A lot of the certifications do not hold much weight in the IT industry, but they do meet a litmus test and allow your people to get in the door to start a discussion. From there, your people need to sell the value of AV and IT combined or they will fall into the trap of competing with every other IT integrator at their game. So, change the game.
The bottom line is that AV is the true value-add in AV/IT. IT will bring the business, but AV will prove that your company can do things that no other player can. So again I say, change the game—and win.
Maxwell Kopsho (CTS-D/I, PMP, CQT, CCNA R&S and Security, CompTIA Network+ and CTT+) has worked in the AV industry for over 18 years in various management and technical roles. Over the last 28 years Max has acquired an extensive background in supporting AV and IT systems, computer networks, telecom, and VTC systems.
Convention Committee Announced for AES 140th International Convention In Paris
Will offer four full days of in-depth programs and presentations, facility tours, and a three-day manufacturer exhibition
The Audio Engineering Society (AES) has announced the official convention committee for the 140th International AES Convention, set to take place June 4-7, 2016, at the Palais des Congrès de Paris in Paris, France.
Co-chaired by Michael Williams and Umberto Zanghieri, the 140th convention will offer four full days of in-depth programs and presentations, facility tours, and a three-day manufacturer exhibition.
Charged with leading the paper sessions at the convention are paper co-chairs Thomas Gorne (Germany), Wolfgang Klippel (Germany), Bergane Periaux (France), Robin Reumers (Belgium), and Dejan Todovoric (Serbia). Co-chairs for the convention’s workshops presentations will be Natanya Ford (UK) and Rob Toulson (UK).
The technical tours will be chaired by Phillippe Labroue (France), while additional support will be provided by facilities co-chairs Layan Thornton (France) and Nadjia Wallaszkovits (Austria).
“Our convention chairs Umberto Zanghieri and Michael Williams have put together an impressive team to build the program for the 140th AES Convention,” says Bob Moses, AES executive director. “The 140th is going to be a great event in a great city. If you are serious about audio, you seriously need to join us in Paris.”
SoundGirls.Org Presenting Slate Of Tour Managing & FOH/ME Seminars In California And New York
Learning the keys to wearing two crucial hats in the touring production world to enhance employment opportunities
SoundGirls.Org is presenting a series of seminars on handling the dual roles of tour management with front of house or monitor engineering, with two upcoming dates slated for California followed by another in New York City.
“When starting out as a front-of-house or monitor engineer, many tours require you to wear two hats,” explains Karrie Keyes, a veteran mix engineer and co-founder of SoundGirls.Org. “The tour manager and FOH/ME, or production manager and FOH/ME, are the most common dual roles you will find. Being able to handle both roles effectively makes you more valuable, increases your skill set, and allows you to gain the experience needed to tour solely as a sound engineer or tour manager.”
First on the itinerary is a seminar at the Rock & Roll Warehouse at 501 Bitritto Way in Modesto, CA, on Saturday, December 19 from noon to 3 pm. Presenters include Rachel Ryan, FOH and tour manager for PHOX and monitor engineer for The Strokes, as well as Chez Stock, FOH and tour manager for several independent artists, including Yuna, Dorothy, and Empress Of.
Next up is a seminar at Planetwood Productions, 5163 Shearin Ave in Los Angeles on Saturday, January 2, also scheduled for noon to 3 pm. Presenters are Chez Stock and Rachel Ryan. Owned by Catharine Wood, Planetwood Productions is a production facility that handles everything from commissioned pieces for ESPN to producing singles and albums for singer-songwriters.
The educational opportunity moves to the East Coast with a seminar the following week at The Unicorn, 105 Henry Street at Pike Street in New York City, scheduled for Saturday, January 9 from noon to 2:30 pm. Presenting is Claudia Englehart, a tour manager and FOH engineer for Bill Frisell since 1989 who has worked with many renowned artists such as Ryuichi Sakamoto, Wayne Shorter, Herbie Hancock, Michael Brecker, David Byrne, Laurie Anderson, and many others.
“What do you need to know to tour manage, how to juggle sound check, and get your artists to sound check?” concludes Michelle Sabolchick Pettinato, a veteran mix engineer and co-founder of SoundGirls.Org, “Come find out The Good, The Bad, and The Ugly of wearing two hats.”
Biamp Systems Announces New Online Certification Training For Audia
'Audia for Technicians' self-paced course allows participants to improve audio know-how while earning two renewal units toward InfoComm CTS credential
Biamp Systems unveils an all-new online training course for its Audia digital audio platform.
Designed to be flexible and on-demand, the self-paced course allows integrators and end-users to gain insights into maintaining the company’s Audia solutions.
“Audia installations continue to power audio for facilities around the globe, creating the need for us to broaden the training available to meet the continued learning needs of the industry,” said Kiley Henner, director of customer experience at Biamp Systems.
“Targeted at professionals tasked with operating and maintaining an Audia system, the ‘Audia for Technicians’ online training course provides valuable knowledge on how to successfully leverage and maintain the benefits of our networked audio solution.”
“Audia for Technicians” is the latest self-paced course from Biamp aimed at providing the knowledge and tools needed for providing maintenance and making minor modifications to existing Biamp AudiaFLEX and Nexia platform audio systems.
Providing two renewal units toward InfoComm International’s Certified Technology Specialist (CTS) credential, the online learning module focuses on existing systems by looking at topics such as AudiaFLEX and Nexia hardware, navigating configuration software, connectivity layers for communicating with hardware, and how peripheral devices such as audio expanders and controllers impact the overall AV system.
In addition, participants will learn how multiple Audia and Nexia systems transmit sound between each other.
To complement the online training experience, Biamp also provides participants with an extensive collection of on-demand videos via the Biamp Training channel on YouTube. To further assist users in harnessing the full potential of Biamp products, the company has created Cornerstone — an online technical support knowledgebase containing detailed technical information on all Biamp products. As a result, partners gain access to an informative, efficient, and well-rounded learning environment.
More information on Biamp’s new Audia online certification training and how to register for the course is available here.
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