Technician

Monday, January 06, 2014

SynAudCon Announces Spring 2014 In-Person Training Schedule

SynAudCon announces 2014 in-person training schedule.

 
Synergetic Audio Concepts (SynAudCon) has released their in-person seminar schedule for the Spring of 2014.

SynAudCon is renowned for their real-world audio educational offerings through web-based and in-person training offered worldwide.

SynAudCon will offer the three-day “Sound Reinforcement for Technicians” (SRT) in Portland, Oregon on February 24-26, 2014 and again in Cincinnati, Ohio on April 2-4, 2014.

SRT instructor Pat Brown provides hands-on exercises which allow attendees to use a tool kit (that includes meters and other items that are needed) to test and troubleshoot systems. The class also goes into detail on how to use modern dual-channel FFT measurement platforms. On day three, SRT demonstrates the setup of a 3-way triamped loudspeaker, including polarity testing, equalization, crossover selection and signal alignment.

“SynAudCon Digital”, a three-day seminar, will be presented April 28-30, 2014 in North Haven, Connecticut. “SynAudCon Digital” is designed to provide a comprehensive introduction to digital audio, digital signal processing and digital audio networks. The materials presented shorten the learning curve for understanding everything from data formats to networked audio systems with an emphasis on the practical. The seminar is taught by Pat Brown, Steve Macatee and Bradford Benn.

For more specific information about the 2014 schedule, seminar agendas, and online registration, visit the SynAudCon website.

SynAudcon

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Posted by Julie Clark on 01/06 at 03:00 PM
AVLive SoundChurch SoundNewsAVBusinessEducationMeasurementSound ReinforcementTechnicianPermalink

Thursday, January 02, 2014

What Have You Done For Your Ears Lately?

Chances are you make at least part of your living with your ears. Stop and think about it. Could you perform your job as well…would your income level be the same…would your professional reputation be intact if you suffer severe hearing loss?

Both musicians and the live sound technicians who work with them need to be able to hear things. Not just hear them well, but hear them better than the average person. This should make us stop and consider our own hearing health, and the environments that we work in.

What have you done for…(and to) your ears lately?

Work-Related Hazards
Did you have your head deep inside a bass bin, listening for a 60-cycle hum, when somebody pushed “play” on the CD player? Were you walking past the tri-amplified sidefill stack, with your ear at compression driver level, when the lighting crew’s ladder knocked the center stage vocal mic stand over into the floor wedge to induce non-stop feedback? Did the drummer hit his primary crash cymbal, hard, 3 inches from your ear, while you were on the drum riser adjusting the hi-hat microphone?

Each of these typical events can be a daily occurrence on a typical concert stage, but any one of them might be the accident that causes you to have either temporary or permanent hearing loss. This could result in a shortened career and a decreased ability to earn a living with your chosen skill.

Accidents are one thing. Constant and intentional exposure to high sound levels is yet another. Did you just finish a 50-show run in tiny concert clubs with that new speed metal band? Was your powerful cue monitor wedge placed on end only one foot from your right ear as you mixed stage monitors for that entire world tour? Do you check 64 house mic line inputs every day with a ragged set of stereo headphones while listening to a clipping headphone amp?

Chances are good that your ears at least need a rest; but there are also certain techniques that can be employed to offer the maximum amount of protection to your hearing as you continue to do your job.

Hearing Protectors
Earplugs are now in use more and more frequently by ushers, security guards, video crew persons, and others who must work at their job while surrounded by the high-level sound intensity of today’s rock music concert programs.

Throw-away foam-type plugs are often issued on a daily basis at arenas and auditoriums for the working crews; some facilities have a nurse or public health official who will provide these items to any member of the general public audience who complains about loud sound levels.

If you’re a technician who works around powerful sound systems, but is not actually responsible for mixing sound during the show, it is a good idea to have some sort of hearing protection device handy.

The same is true if you are a sound professional who is waiting around for your band to come on while listening to someone else operate a loud system. Here are some basic options:
Disposable Foam Plugs. This type of hearing protection device comes in a small cardboard or plastic pouch, and several can easily be stuffed in a shirt pocket or a briefcase pouch. They are disposable, intended for one-time use. Common brands are E.A.R., and DeciDamp from North Health Care. Such devices offer a noise reduction rating of about 12-20 dB, depending on frequency. These plugs mainly reduce high frequencies.

Re-usable Silicon Insert Plugs. These rubberized insert cushions conceal tiny metal filtering diaphragmatic mechanisms to attenuate sound levels. They are often seen in use by gun buffs, construction workers and heavy equipment operators. The Sonic Valve II comes in its own plastic storage case with a key chain attached, and offers about a 17 dB noise reduction rating. Often available in gun shops or industrial safety supply stores, a pair can run from $15-20.

Personal Custom-Fit Earmolds. The best hearing protection device, and the one most applicable to working around musical sound, is one that attenuates all frequencies evenly. When correctly designed and properly fitted, custom-molded flexible plastic earmolds can offer 15-20 dB of balanced noise level reduction; in other words, full-frequency sound is still heard, but at a reduced level. There are numerous suppliers, who provide custom fitting services as well, such as Sensaphonics.

Industrial Headsets. When maximum attenuation of very loud sounds is desired, particularly at low frequencies, the cushioned headset works well. Offering up to 30 dB of attenuation, hearing protectors from David Clark have cushioned headpads and tight-fitting earseals. This is also an option for person who do not wish to stick standard earplugs inside the ear.  This is the type of protection often seen in use on airport runways and in the cabs of tractors and heavy cranes at construction sites.

Protecting Your Hearing On The Job
Use mini-nearfield monitors as a cue system for live mixing instead of headphones whenever possible.

By placing one or two small, powered monitors at your mixing console position and giving them the output from your stereo cue bus, you are able to solo up a mic input or an output mix and hear the signal without having to put on regular stereo headphones.

Roland, TOA, Yamaha, Tascam and other musical-instrument oriented manufacturers offer a variety of compact products.

This is particularly handy during setup and sound check. Using this method, you’ll have less loud, direct sound putting pressure on your eardrums, yet you will still hear the needed information.

Dummy headphones can be used as a quick way to lower the sound level of what you hear. Simply put on your regular stereo headphones, but don’t plug them into anything. Run the cord into your pocket. This will offer isolation from the louder acoustical environment that surrounds you during a show, while your ears have a chance to rest.

Rests away from the job site should be taken whenever possible. Remove yourself from the noisy environment and take time to have a meal, a nap, read a book, or whatever there is to be done in a quieter space. Focus on finding a ‘quiet zone’...no blaring TV or Walkman headphones. This can mean a walk outdoors, finding a secluded dressing room, or whatever.

The important thing when working around loud sound levels is to give your hearing system and ear mechanism time to recover. If you work in a loud environment, your hearing will be more sensitive and ‘fresh’ if you take regular breaks like this.

Sound Level Meters
If you do not already include a hand-held, battery powered SPL meter in your working toolkit, get one. Don’t rely on assumed level readings from your 1/3-octave real-time analyzer unless you are absolutely sure that the correct microphone is in use, (mic sensitivities can vary greatly, causing erroneous SPL readings), and that the system is properly calibrated. It’s better to have a small portable unit that you can keep in front of you on the mix console, or carry around the venue with you as you check coverage.

These handy devices can range in price from $65 (Radio Shack) to $2,500 (Bruel & Kjaer). I recommend the General Radio 1565-B Sound Level Meter (about $600); this is a hand-held battery powered meter that is approved by US Government agencies for environmental noise measurements. With its OSHA certification sticker, it helps you stand up to noise regulation officials, many of whom may have less sensitive and reliable gear.

Almost any type of SPL meter will do what you need; the accuracy difference between the cheapest and the most expensive can be about 1-2%...this would mean a possible error, plus or minus, of 1-2 dB at around 100 dB SPL. The more sophisticated, expensive units are best for critical situations.

Learn the difference between ‘A’ and ‘C’ weighting filter scales (US Government agency guidelines stipulate the use of C-weighted measurements for noise environments dominated by frequencies below 500 Hz; A-weighted measurements are most useful for making comparative readings in live show environments and discussing levels with others).

Use the sound level meter to get useful information in the front rows, the high balcony, the back of the hall, at the console…wherever you need to know the actual, average sound pressure level of your show.

Find the ideal ‘pocket’ where your show mix is as exciting and powerful as it needs to be, yet where you do not get audience complaints about excessive volume.

Use your meter as a daily reference guide, regardless of the type of acoustical environment.

Paying attention to the level of your system’s operation will be one more step toward protecting your own hearing, as well as that of others.

Long-Term Effects Of Loud Sound
We have probably all experienced TTS (Temporary Threshold Shift) after being exposed to very loud music or other sounds. This is the sensation that someone has stuffed cotton in your ears after you have already walked out of a loud environment; after one or two hours of high-level listening, your shifted hearing threshold may compensate as much as 40 or 50 dB.

In other words, your ears have ‘shut down’ to reject the extra-loud sounds that you have exposed them to. Recovery may take from a few hours to several days.

Prolonged exposure to very loud music can bring on tinnitus, which is a ringing sensation that you hear in your ears, even though no loud sounds are present around you. If you experience this ringing several days after exposure to a powerful sound system, consider that to be your own body’s way of giving you a danger signal. Heed the warning.

Have a regular hearing checkup. Get to know your audiologist or hearing specialist. Once or twice a year, get checked for both air and bone conducted sound sensitivity, speech understanding, and make sure that your inner ear parts are functioning properly.

If your job involves working with live sound, and you want to continue doing it, take time to carefully consider what your own personal approach is going to be as you work to conserve your hearing. You are also preserving your livelihood in the process.

David Scheirman is vice president, tour sound at JBL Professional and is also a long-time contributor of pro audio and sound reinforcement editorial.

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Posted by Keith Clark on 01/02 at 03:04 PM
Live SoundFeatureBlogStudy HallProductionAudioEducationEngineerLoudspeakerTechnicianPermalink

Thursday, December 19, 2013

In The Studio:The Trouble With Cheap Mics

On the whole, inexpensive audio gear sounds better than ever and is a much better value than even a decade ago, and yet...
This article is provided by Bobby Owsinski.

 
In many ways we’re in the golden age of audio gear. On the whole, inexpensive audio gear (under $500) sounds better than ever and is a much better value than even a decade ago and way better than 20 years ago.

The same can be said for mics, as there is a large variety of cheap mics that provide much higher performance for the price than we could have imagined back in the 70s and 80s.

That said, there are some pitfalls to be aware of before you buy. Here’s an excerpt from The Recording Engineer’s Handbook, 3rd Edition, that covers the potential downside of inexpensive mics.

One of the more interesting recent developments in microphones is the availability of some extremely inexpensive condenser and ribbon microphones in the sub-$500 category (in some cases even less than $100).

While you’ll never confuse these with a vintage U 47 or C 12, they do sometimes provide an astonishing level of performance at a price point that we could only dream about a few short years ago. That said, there are some things to be aware of before you make that purchase.

Quality Control’s The Thing
Mics in this category have the same thing in common; they’re either entirely made or all their parts are made in China, and to some degree, mostly in the same factory. Some are made to the specifications of the importer (and therefore cost more) and some are just plain off-the-shelf.

Regardless of how they’re made and to what spec, the biggest issue from that point is how much quality control (or QC, also sometimes known as quality assurance) is involved before the product finds its way into your studio.

Some mics are completely manufactured at the factory and receive a quick QC just to make sure they’re working and these are the least expensive mics available. Others receive another level of QC to get them within a rather wide quality tolerance level, so they cost a little more. Others are QC’d locally by the distributor with only the best ones offered for sale, and these cost still more.

Finally, some mics have only their parts manufactured in China, with final assembly and QC done locally, and of course, these have the highest price in the category.

You Can Never Be Sure Of The Sound
One of the byproducts of the rather loose tolerances due to the different levels of QC is the fact that the sound can vary greatly between mics of the same model and manufacturer.

The more QC (and high the resulting price), the less difference you’ll find, but you still might have to go through a number of them to find one with some magic. This doesn’t happen with the more traditional name brands that cost a lot more, but what you’re buying (besides better components in most cases) is a high assurance that your mic is going to sound as good as any other of the same model from that manufacturer.

In other words, the differences between mics are generally a lot smaller as the price rises.

The Weakness
There are two points that contribute to a mic sounding good or bad, and that’s the capsule and the electronics (this can be said of all mics, really). The tighter the tolerances and better QC on the capsule, the better the mic will sound and the closer each mic will sound to another of the same model.

The electronics is another point entirely in that a bad design can cause distortion at high SPL levels and limit the frequency response, or simply change the sound enough to make it less than desirable. The component tolerances these days are a lot closer than in the past, so that doesn’t enter into the equation as much when it comes to having a bearing on the sound.

In some cases, you can have what could be a inexpensive great mic that’s limited by poorly designed electronics. You can find articles all over the web on how to modify many of these mics, some that make more of a difference to the final sound than others.

If you choose to try doing a mod on a mic yourself, be sure that your soldering chops are really good since there’s generally so little space that a small mistake can render your mic useless.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. Get the 3rd edition of The Recording Engineer’s Handbook here.

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Posted by Keith Clark on 12/19 at 06:01 PM
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Church Sound: Shrinking Buildings & What That Means For Worship Tech

We’re seeing a significant shift from big worship to an emphasis on groups
This article is provided by ChurchTechArts.

 
I’ve been saying this for a while, but there is a change a comin’ in the church. While some disagree with me, I suggest that we are nearing the end of the era for big church buildings with big production going on.

Not that big buildings will disappear all together, but there will be fewer of them and they will not be the sought-after goal of most churches.

I’m hearing similar thoughts from other church leaders, and the latest to chime in is Thom Rainer. Last week he wrote a post listing seven reasons church worship centers will get smaller.

Unlike the last time I referenced a post in an article, I’m pretty much in agreement with him on this one. I really believe this is coming, and it’s going to affect what we do as technical leaders. Let’s consider some of his points.

Multi-Site & Multi-Venue Churches Are On The Rise

We see this everywhere. More and more churches are discovering that they can have a significantly more powerful impact on their community by launching multiple, smaller campuses instead of one big one. Or perhaps they will do multiple venues with different worship styles. Either way, this tend is here to stay (until the next trend, anyway).

What does this mean for us? On the plus side, all of these venues will need at least a basic production technology package. Often, it will need to be portable. So we’ll have a lot of gear to manage.

However, with smaller campuses, come smaller congregations (that’s kind of the point, right?), and smaller budgets. Not many churches will be hiring full-time guys to run campuses. I suspect what we’ll see is churches hiring one or maybe two technology directors who will oversee all the campuses, helping recruit, train and keep volunteers going.

In this scenario, technical leaders won’t be nearly as hands-on; we won’t be able to be in five places at once. But we will need to be really good at putting together packages of gear that can survive being loaded in and out by volunteers each week. If you have holes in your technical systems knowledge, now is the time to fill them in.

We’re Seeing A Significant Shift From Big Worship To An Emphasis On Groups

Thom points out—correctly in my opinion—that churches are starting to move away from the worship service being the central event of the church. It’s not going to go away, but there will be more emphasis on groups. Churches will be needing to raise up more leaders who can lead groups.

What does this mean for us? We’re already seeing it. Churches are becoming less interested in hands-on techs and more interested in technical leaders who can train and develop others to do the work.

Again, this won’t be binary. There will likely always be churches with large tech staffs who do the work. But I suspect we’ll see a shift towards volunteer teams, even in larger churches. If you’re a hard-core tech with no people skills, this is going to be a challenging transition for you.

But if you’re a builder of people and teams, you will do well. Now is the time to start honing those leadership and discipleship skills; you’ll be needing them!

We Will Be Spending Less On Buildings, More On Ministry

Again, more and more churches are foregoing a large, expensive worship center (or sanctuary, auditorium or whatever you want to call it), so they have more funds to invest in community ministry programs. I’ve always been conflicted with how much production technology costs.

On the one hand, I believe if we’re going to commit to doing production, we should do it well, and that takes money. On the other hand, I wonder sometimes if our priorities are misplaced. I’m not settled on this, and I suspect we’ll always live in tension in this regard.

What does this mean for us? Budgets will continue to shrink. We’ll have to find ways to do more with less. We will need to get very creative in how we do production. It may be that we do less production, but do what we do very well.

Hard choices will need to be made, and this will be a problem for some. If you refuse to work on anything but a Grand MA2 or a DiGiCo SD7, this may be hard for you. But if you’re open to scaling back and still doing production with excellence, this is going to be a lot of fun.

Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

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Posted by Keith Clark on 12/19 at 05:35 PM
Church SoundFeatureBlogOpinionTrainingBusinessEngineerSound ReinforcementSystemTechnicianPermalink

Wednesday, December 18, 2013

Phase & Polarity: Causes And Effects, Differences, Consequences

The terms "polarity" and "phase" are often used as if they mean the same thing. They do not.

Polarity and Phase - these terms are often used as if they mean the same thing. They are not.

POLARITY: In electricity this is a simple reversal of the plus and minus voltage. It doesn’t matter whether it is DC or AC voltage. For DC, Turn a battery around in a flashlight and you have inverted or, more commonly stated, reversed the polarity of the voltage going to the light bulb. For AC, interchange the two wires at the input terminals of a loudspeaker and you have reversed the polarity of the signal coming from that loudspeaker.

PHASE: In electricity this refers only to AC signals and there MUST be two signals. The signals MUST be of the same frequency and phase refers to their relationship in time. If both signals arrive at the same point at the same time they are in phase. If they arrive at different times they are out of phase. The only question is how much are they out of phase, or stated another way, what is the phase shift between them?

The important point to note in these definitions is that you can reverse the polarity of one signal and you can measure this change. You need two signals to measure a phase shift.

For convenience, the word “speaker” will be used in place of the more correct term “loudspeaker” in the rest of this article.

A picture is worth 1,000 words… but a few words of explanation can help.

The following figures show the differences and some consequences of polarity and phase. Figures 1 through 12 show graphs of sine wave signals. Actually it is a sine wave from one signal source split two ways. Except for figure 1, one of the splits is “processed” by reversing its polarity and/or by delaying it (phase shifting it) as described. To put this in the real world, imagine two speaker systems side-by-side, each reproducing one of the signal splits. (More precisely, the graphs show what you would see on an oscilloscope looking at the output of a mixing console with each split going to a separate input after one of the splits has been “processed”.)

The vertical scale in the graphs is in arbitrary units of -2 to +2 with lines at each 0.5 interval. If you like, consider this as -2 to +2 volts. Because phase shifts are measured in degrees, the horizontal scale in the graphs is labeled in degrees with a vertical line at each 90-degree point. One full cycle or period of a sine wave is 360 degrees.

Assume that the signals shown are 1 kHz sine waves, in which case each vertical line represents 1/4 millisecond of time. Sound travels in air about 3.4 inches (85 mm) in 1/4 millisecond so each vertical line also represents this distance. Note that in the graphs the signals all start 1/4 millisecond or more from the left so you can clearly see when each signal starts. (The importance of this will be seen in figure 9.) There is no signal along the flat line from -90 to 0 degrees.

Signals In Polarity, In Phase
Figure 1: This shows 3 periods or 3 cycles of two simple sine waves. Both are +/-1 volt high at their peaks = total of 2 volts. One is shown in blue the other in red.

Figure 1: Sine Waves in Fig. 1 Added.

Figure 2: This is what happens when the two are combined (= added together). This is exactly what would happen on a line exactly between the two side-by-side speakers. The two signal beings being in phase and in polarity add up so the peaks are now at the +/- 2 volt lines = 4 volts or twice the original signals. Acoustically this is an increase of 6 dB = 20 x log(1+1).

Figure 2: Two Sine Waves - Same Polarity & Phase.

Signals Out of Polarity
Figure 3: This is like figure 1 but the second sine wave, shown in red, has been reversed in polarity. As you can see the + and - voltage points are exactly opposite from the first sine wave, shown in blue. This would be accomplished by reversing the +/- input connection on the speaker reproducing the red sine wave.

Figure 3: Two Sine Waves - Red = Polarity Reversed.

Figure 4: This is what happens when the two are combined. Each point of the two signals being in phase, but opposite polarity, adds up to zero. Acoustically this is an infinite decrease of output. Because you can’t take the log of 0 assume the difference is actually 0.0.01 volts (the dots = 58 more zeros). 20 x log of this number is -1200 dB. That should be pretty quiet. You can’t easily hear this with two speakers because of having two ears. But using a very carefully positioned microphone to measure this in a place with no sound reflections, you would find almost no signal.

Figure 4: Sine Waves in Fig. 3 Added.

Signals Ot of Phase

Figure 5: The second sine wave, shown in red, starts 1/4 millisecond later (90 degrees later) than the first one, shown in blue. Put another way, the second signal has been delayed by 1/4 millisecond.

Figure 5: Two Sine Waves - Red = Phase Shifted 90 Degrees.

Figure 6: This is what happens when the two are combined and it’s pretty interesting. First notice that the peaks are almost at the +/-1.5 volt lines. The value is actually +/-1.414 volts. This is a 3 dB increase. This would be like listening to two speakers but the one reproducing the red sine wave is 3.4 inches (85 mm) further away from you than the other. The first thing you hear is only from the speaker reproducing the blue sine wave. The black line starts when the sound from the second speaker is heard and this line is the combined signal of both speakers.

Figure 6: Sine Waves in Fig 5 Added

Suppose the speaker reproducing the red signal were only 2.25 inches (57 mm) further away. The signals would be shifted by only 60 degrees. The increase for the combined signal would be about 4.5 dB. So the amount of phase shift is important.

The second thing to notice is what happens at 1/4 millisecond or 90 degrees after the blue signal starts when the second signal “kicks” into the picture represented by the line turning black. There is a distinct change in the waveform.

The third thing to notice is that the entire waveform after the “glitch” is shifted in time compared to figure 7 about 45 degrees = average of 0 and 90 degrees.

Signals Out Of Phase And Polarity

Figure 7: The second sine wave, shown in red, is a combination of the sine wave in figures 3 and 5. The signal not only has its polarity reversed but it is shifted in phase by 90 degrees compared to the first signal, shown in blue. In this case the speaker reproducing the red sine wave has its +/- input connection reversed in polarity and is 3.4 inches (85 mm) further away from you than the one reproducing the blue sine wave.

Figure 7: Two Sine Waves - Red = Phase Shifted 90 Degrees & Polarity Reversed.

Figure 8: This is what happens when the two signals are combined. The picture is similar to figure 6 with two important differences. First the “glitch” at the point where the second signal starts is different. This is the point where the line turns black. Second is that the entire waveform is shifted by 45 degrees again but this time to the left of the original signal.

Figure 8: Sine Waves in Fig. 7 Added.

The “Glitches”
The glitches in figures 6 and 8 give an indication of what happens during the onset of a signal. While the so-called steady state portion of the combined signal (shown by the black portion of the lines) looks the same except for the amplitude change, these glitches will affect the transient attack of sounds. This is not to say that either will sound horrible, but a phase shift between otherwise identical replicas of a sound WILL make a difference in the sound of the initial transient attacks, depending on the frequency and amount of phase shift.

This is exactly the kind of phenomena that can occur in the crossover region of a speaker. This is because the distance from each driver to the listener is usually different and the crossover itself shifts the phase of the signal between the drivers. Speaker designers are often faced with a choice between something like what you see in figures 6 and 8. Neither is “correct” so a designer can only choose the one that “listens” better. Just looking at these two, I would bet the waveform in figure 8 might sound better and the choice would be to reverse the polarity of one of the drivers. These crossover “glitches” occur only over a small range of frequencies where both drivers reproduce the sound. It is well accepted by designers that this kind of “improvement” is sonically more significant than the fact that frequencies above and below the crossover point may be out of polarity.

Signal Phase Shifted 180 Degrees
This is where many get into trouble in thinking that phase and polarity are the same thing, meaning that it is often assumed that a 180 degree phase shift and reversing the polarity are the same.

Figure 9: In this figure each sine wave lasts for only 2-1/2 cycles. The second sine wave, shown in red, is shifted in phase 180 degrees from the first shown in blue. This is what would happen if the speaker reproducing the red sine wave were about 6.8 inches (170 mm) further away from you than the one reproducing the blue sine wave. You can see that between the 180 and 900 degrees the signals LOOK like they are simply out of polarity but they are NOT. It is VERY important to note that if you could not see the beginning or the end of these signals you could not tell whether they were out of polarity or 180 degrees out of phase. Too often this is what causes confusion between a polarity reverse and a 180 degree phase shift.

Figure 9: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.

Figure 10: This is the result of combing the two signals. Unlike figure 4 where the signals are simply out of polarity, and completely cancel, there are clearly two positive halves of a sine wave visible before and after the two signals cancel along the black line between 180 and 900 degrees. The first is from the blue sine wave in figure 9 that occurs before the start of the red sine wave. The second is from the red sine wave in figure 9 that continues after the blue sine wave stopped.

Figure 10: Sine Waves in Fig. 9 Added.

Signal Phase Shifted 180 Degrees And Reversed In Polarity

Figure 11: This is the same as figure 9 but the polarity of the red signal is reversed from figure 9.

Figure 11: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.

Figure 12: This is the two signals in figure 11 combined. Between the 180 and 900 degrees, the signals add much like in figure 2. However there are significant differences in the overall 90 to 1080 degree signal. The first 1/2 sine wave of this signal is only from the blue sine wave from figure 11. The last 1/2 sine wave is only from the red sine wave in figure 11. You can clearly see that both of these 1/2 sine waves are only 1 volt at the peaks. This is a clear difference from figure 2 where all the peaks reach 2 volts.

Figure 12: Sine Waves in Fig. 11 Added.

The reason is that the two signals in figure 11, even though identical, are offset by 180 degrees. They add together only between 180 and 900 degrees when both are being heard. More importantly, during this time period DIFFERENT parts of the same signal have added together. For example you can see that between 180 and 360 degrees it is the second 1/2 of the blue signal’s first complete sine wave that adds to the first 1/2 of the red signal’s first complete sine wave.

Real Audio Signals
Sine waves are easy to look at to dramatically show the difference between polarity and phase. Armed with this knowledge you can look at figures 13 through 18 that show something like a real audio signal where the effects of polarity and phase are more difficult to see.

The signal shown in these figures was a generated by a mathematical algorithm that produces something close to a pink noise signal. Pink noise contains all frequencies with an equal amount of energy in each octave band. Real audio signals don’t look much different than pink noise (but one would hope they sound better!). The scales on these graphs are arbitrary. You can look at the vertical scales as +/-3 volts if you like. However, because of the way the signal was generated, there was no way to define absolute time or degrees along the horizontal scales. Suffice it to say that the phase-shifted signal used in these figures was shifted by one data point out of the 240 data points that make up the signal lines.

There is one important thing to understand about phase shift. The amount of time one signal is delayed from another will have different effects at different frequencies. Assume there is a 1 millisecond time difference between two identical signals. At 500 Hz the result will be as shown in figure 10 because at 500 Hz the 1 millisecond time difference is a phase shift of 180 degrees. The signals are offset by 1/2 a cycle.

At 1 kHz the signals will be offset by 1 complete cycle. In other words you would hear one cycle from the first signal then both combine then you’d hear the one cycle from the second signal after the first stopped. This is similar to what is shown in figure 12 (which shows only 1/2 cycle) but is not the result of the same conditions that were used to make figure 12.

At 250 Hz the effect would be as shown in figure 6 because a 1 millisecond time difference corresponds to a 90 degree phase shift at 250 Hz or an offset of 1/4 cycle. At lower frequencies the phase shift would be even less and the signals would tend to add as in figure 2, approaching but never quite reaching the 6 dB increase shown in that figure.

Contrary to phase, polarity affects all frequencies the same way. It makes the positive portions negative and the negative portions positive. Put another way, it simply flips the signal over the same way at all frequencies. With these things in mind, examine figures 12 through 18

Effects of Polarity and Phase On “Real” Audio Signals
Figure 13: This shows a pink noise signal generated as noted above.

Figure 13.

Figure 14: This shows both the original signal in blue and what happens when an identical but phase shifted signal is added to it, as shown in red. The red signal is similar to the combined signal shown in figure 6. Note the increases in signal level and the changes in the waveform (many glitches). However you can also see the combined signal follows the original fairly closely.

Figure 14.

Figure 15: This shows both the original signal in blue and what happens when the phase shifted signal is also reversed in polarity and combined with it, as shown in red. In this case there are huge differences between the original and combined signal.

Figure 15.

Figure 16: To better understand what is going on, this figure shows an averaged or integrated version of the pink noise signal in figure 13. This is basically what would you would see if you graphed the readings from a typical SPL meter for the signal in figure 13.

Figure 16.

Figure 17: This shows the averaged signal from figure 16, in blue, and the averaged combined signal from figure 14, in red. Note that there are primarily level differences (mostly increases). Otherwise the two lines look very similar.

Figure 17.

Figure 18: This really shows what is going on in figure 15. The blue line is the averaged signal from figure 16. The red line is the averaged signal from figure 15. The red line shows that the out of polarity and phase-shifted signal approaches a straight line. Because you are looking at a broad frequency range, you are seeing a severe cancellation of the lower frequencies due to the polarity reversal. However, unlike the low frequencies, the upper frequencies do not completely cancel due to the phase shift. The red line contains primarily high frequency energy. In the blue signal the higher frequencies are the small “bumps”. These can be clearly seen in the red signal and most of them correspond to those in the blue signal.

Figure 18.

Figure 18 is a prime example of what you would hear if you stand exactly between two speakers playing the same signal (i.e. mono) with one speaker out of polarity. The bass will disappear. But, there will always be a difference in distance between you and the speakers due to the spacing of your two ears and probably a slight overall difference in distance between you and each speaker. A difference in distance means a difference in the time arrival and thus there will be phase shifts between the sound from the two speakers. The amount of shift will vary with frequency. Because of the shorter wavelengths at high frequencies, the phase shifts allow most of the highs to be heard. They may be out of polarity but the effect is like what is shown in figure 8. Also, in a room you would also hear sound reflections from the floor, walls, and ceiling. You would only hear something like the red line in figure 18 outdoors away from any reflective surfaces or in an anechoic chamber.

Figure 19.

The small distance between your ears and any small difference in distance from you to each speaker do not cause appreciable phase shifts at low frequencies. This is because of the considerably larger wavelengths. The difference in your distance from each speaker might be only 1 inch (25 mm). However, the wavelength of even a 1 kHz sound is roughly 1 foot (300 mm) and at 100 Hz roughly 10 feet (3 m). At the lower frequencies the polarity difference predominates because the phase shifts due to the difference in your distance from the speakers is very small compared to the wavelengths of the low frequencies. Thus the lower frequency signals, being nearly in phase but out of polarity, will cancel like in figure 4. The lower the frequency the less the phase shift between the two speakers and the greater the cancellation.

A Polarity / Phase Field Trip!!
(As with all physical exercise, check with your doctor first, who might not recommend you do this for some reason.)

Find two railroad tracks, lie across them, and wait.

Two trains, one on each track, come along. Both are right side up and both hit you at exactly the same time. The trains are in polarity and in phase.

The same thing happens again and both trains hit you at exactly the same time. However, this time one train is upside down. That is a polarity reversal.

The third time both trains are right side up but one hits you first and the other hits you shortly after the first. That is a phase shift.

The last time the second train is upside down and hits you later than the first. That is both a polarity reversal and a phase shift.

Summary
So there you have it. Although this has only touched on a few areas concerning phase and polarity issues, it is hoped you better understand the difference between the two and a few of the effects of each. Remember that the audio frequency range covers wavelengths of over 30 feet (10 meters) at the lowest frequencies to less than an inch (under 25 mm) at the highest frequencies.

While a reversal of polarity will affect all frequencies identically, a difference in time arrival between two otherwise identical signals will have very different effects on the phase between them. The amount of phase shift will be different at different frequencies and this will depend on how much time difference there is between the arrival of the two signals.

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Posted by Keith Clark on 12/18 at 03:35 PM
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IK Multimedia releases UltraTuner For iPhone, iPad And iPod Touch

UltraTuner utilizes a new patented detection engine making it the most accurate instrument tuner available for iOS - 10 times more accurate than a mechanical strobe tuner

IK Multimedia is pleased to announce UltraTuner, the most precise digital chromatic instrument tuner app for iPhone, iPod touch and iPad.

UltraTuner is available as standalone app for iPhone and iPod touch or as an in-app purchase within AmpliTube for iPhone and iPad (in all versions).

UltraTuner features one of the fastest, smoothest and most responsive tuning engines of all tuner apps, and is precise to well below a hundredth of a cent - far beyond human perception of pitch, making it 10 times more accurate than a mechanical strobe tuner (considered the gold standard of tuners).

UltraTuner’s accurate pitch detection is achieved through a proprietary dual-analysis processing engine that allows the tuner to get a much more accurate reading of the frequency from the incoming note. The fast response of the tuner makes tuning a mechanical instrument a breeze. UltraTuner’s precision also makes it the perfect app tuner for intonating string instruments like guitar or bass, and by offering note precision down to less than .01 (1/100th) of a cent, making it ideal for calibrating electronic instruments like vintage synths.

UltraTuner’s precision and speed provides players and engineers with a pocket-sized professional instrument tuner always at hand and offers two basic modes of operation: Stage and Studio.

When in Stage mode, UltraTuner is a straightforward, ultra-accurate instrument tuner featuring a unique, simple interface that’s very easy to see in low light conditions. It features a graduated flat and sharp display to indicate degrees from pitch, and when the note is in tune, the display turns green. UltraTuner utilizes the entire device screen for tuning so it is easily visible from a distance on dimly lit stages.

In studio mode, UltraTuner provides pitch tracking - allowing players to monitor pitch over time. This is especially useful in the studio for monitoring pitch of non-chromatic instruments like vocals, violin and brass etc., on recordings over time, and very useful for vocal and instrument pitch training. Studio mode also offers two different visualizations: pitch over time and a proprietary oscilloscope wave form display.

The pitch over time display shows the accuracy of a note as it is played and decays. The oscilloscope wave form display indicates flat or sharp notes by animating the waveform to the left if the note is flat, completely still if the note is right on, and animating the waveform to the right if the note is sharp. These visualizations make it very quick and easy to reference notes while playing a non-chromatic instrument or singing in the studio.

In the settings window, players can quickly calibrate the “A” note, set the temperament type, set the root key for tracking and adjust the audio sensitivity. UltraTuner can use the device’s built-in microphone for audio input, and works with all analog or digital audio interfaces for iPhone, iPad and iPod touch, like IK’s iRig HD, iRig PRO and the iRig MIC series of products.

Key Features:

  Innovative patented tuning engine that provides the highest level of responsiveness and accuracy with 0.01 cent resolution
  Dual Mode: Stage and Studio
  Large Tuning indication visible from a distance (stage mode)
  Tuning History to see the tuning variations over time (studio mode)
  Temperament selection includes: Equal, Pythagorean, Just Major, 1/4 Comma Mean Tone, Kimberger III, Werckmeister III, Young and Kellner
  Useful not only for tuning, but also for non-chromatic instrument training

UltraTuner is available now as a standalone app from the App StoreSM for iPhone and iPod touch for $4.99/€4.49, or as an in-app purchase within AmpliTube for iPhone (Free/LE/Full versions), and AmpliTube for iPad (Free/Full versions).

IK Multimedia

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Posted by Julie Clark on 12/18 at 10:30 AM
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Tuesday, December 17, 2013

Church Sound: Mixing Like A Pro, Part 4—Making EQ Work For You

Looking at the actual sonic makeup of the sounds that come from the stage
This article is provided by CCI Solutions.

 
Editor’s Note: Go here to read previous installments in this series.

In the previous article, we took a look at various equalization (EQ) tools, identified their functions and what they can do for us. After all, we can’t use our tools effectively if we don’t know what they are or how to use them.

We’ve covered what EQ does to shape frequencies in our sound system audio, but to know how to make EQ work for us we have to look at the actual sonic makeup of the sounds that come from the stage.

Putting EQ Into Words
Before we focus on what we’re EQ’ing, we need to learn how to interpret common language into tangible EQ adjustments. You know what I’m talking about, comments like “it’s too edgy” or “it sounds muddy.” What does that actually mean?

The chart below gives us some hints using words like rumble, muddy and edge. With the help of this chart, we can take an educated guess that when someone says an input sounds “honky,” they’re referring to something in the 440-1,000 Hz range.

It’s not an exact science, I know, but getting a good feel for what responses are elicited by certain frequencies can help us in making EQ adjustments quickly. So the next time someone tells you the electric guitar sounds “edgy” or “crunchy,” you know you need to quickly look at the 2,000-4,000 Hz range to attack your problem.

Click to enlarge. Go here for an interactive version of this chart.

Focusing On Frequency Ranges
Everything we hear is made up of a range of frequencies. Each sound that hits our eardrums is made up of a collage of frequencies at a blend of sound pressure levels that our brain interprets as “the sound.”

Just as each person’s voice has a unique makeup and signature, every instrument or vocals has a makeup of frequencies that is unique to it. In order to talk about how to EQ an input, we need to learn what frequencies are involved in the sound sources we’re working with.

The chart is a great place to start to begin to understand what frequencies make up the sounds we experience on a Sunday morning. It shows us the range of any given source and it shows us the frequencies we need to focus on—and not focus on.

For example, the range of a guitar will typically start around 80 Hz and will top off around 5 kHz. Knowing there is nothing being produced below 80 Hz, the first thing we can do is turn on the low cut/high pass filter to eliminate any low frequency junk that our guitar isn’t actually producing.

We also know that the guitar isn’t producing frequencies over 5 kHz, so turning up the highs above that just adds unhelpful noise. Based on this chart we know our focus needs to be between 80 and 5 kHz.

Critical Sound — The Voice
Our most critical source in the church, the human voice also has clear-cut frequency ranges, regardless of whether your voices are singing or speaking.

While everyone’s voice has minor variations, the male voice produces frequencies between 100 and 16 kHz. While the female voice also shares the top end of 16 kHz, it doesn’t typically produce any frequencies below 240 Hz.

The first thing this should tell you is your low cut or high pass filter should almost always be engaged on these inputs. As you can see on our chart (previous page), the warmth or boominess of the voice is between 100-250 Hz, so most of the time there is nothing worth having below 100 Hz.

The most important frequency range in the voice in my opinion, and the one I see most commonly mis-adjusted, is the intelligibility range in the high-mids (2 kHz to 4 kHz).

When listening to vocals that are “honky” or “tinny,” I often see sound guys reach for the high-mids and adjust those down to try and improve the sound.

As we can see on the chart, we’re actually attacking the intelligibility when lowering the high-mids, and missing the “honky/tinny” sound that’s in the 400-2 kHz range. It seems like such a small miss on paper, but this mistake will often cost the vocals their clarity in the mix.

What To Do With Q
If you’re fortunate enough to have a full parametric EQ with a Q knob, you have a tool that allows us to get very specific with our EQ adjustments or make general, sweeping changes. Most of the time we want to make fairly general adjustments and a single octave change is great, which is a Q value of 1.

If you’re needing to subtract a bit of “honkiness” from your guitar though, a 2 dB cut centered at 700 Hz with a lower Q (maybe .7) will give you a broader, wider cut to effect everything in that 400-1 kHz range.

Or if you’re fighting a particular frequency for feedback, you can make your Q very high (maybe 5 or 6) so that you are narrowly notching out the frequency that’s causing you issues, leaving the rest of the frequencies alone and keeping some semblance of natural sound.

The Q, if you have it, really gives you a great deal more flexibility in the adjustments you make.

Feedback
Speaking of feedback, one last thing our chart can help us with is learning what individual frequencies sound like. We’ve all dealt with feedback at some point. A frequency that’s sensitive enough that when amplified the mic picks it up from a monitor or speaker again and again creating a feedback loop.

When feedback occurs, one common way to attack it is adjusting the EQ to decrease the gain of that frequency. To be able to do that quickly and effectively, we need to know what individual frequencies sound like.

Frequency Killing
At the bottom of the chart is a standard 88-key piano that shows what frequency each note produces. When it comes to training your ears to be able to quickly respond to feedback, sitting at a piano or keyboard with this chart can help you learn exactly what frequencies sound like.

Try it sometime: sit at a keyboard and focus on a typical problem range of 200-500 Hz and press one key over and over, training your brain to recognize the tone of each frequency.

The middle C is a great place to start, with a frequency of 256 Hz. I find this to be a common problem for many churches. Then jump up to middle A, with a frequency of 440 Hz. Do this occasionally, spanning the entire frequency spectrum and you’ll be a frequency killer in no time!

Wrap Up
Hopefully at this point you’re feeling more confident in what EQ is and what it can do for you. The difficult part is that there is no clear cut formula for what will and won’t work.

I can’t tell you that you should always cut certain frequencies to get a great sounding input. Our vocals and instruments are living, breathing, unique things and they all have their own flavor.

On top of that, every sound system and facility have their own nuances that come into play. When it comes to EQ, you have to trust what you’re hearing. Use the chart. Print one out and keep it at your sound console for reference.

 

Duke DeJong has more than 12 years of experience as a technical artist, trainer and collaborator for ministries. CCI Solutions is a leading source for AV and lighting equipment, also providing system design and contracting as well as acoustic consulting. Find out more here.

 

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Posted by Keith Clark on 12/17 at 04:34 PM
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In The Studio: Tips For Better Take Management

One of the major differences between an aspiring producer and an established one...
This article is provided by the Pro Audio Files.

 
One of the major differences I’ve seen between an aspiring producer and an established producer is simple playlist (take) management. Great producers will usually have a very clean session in regard to organization and take management.

Gizmos
Technology is great. It allows us to do things that have never been done before, all in the comfort of our own homes. But when is it a hinderance? When do we become a prisoner of all the possibilities? When do we start to drown in endless options?

Established producers often have a lot of clarity within their sessions. They’re not concerned with countless possibilities, rather the best option.

This means when it comes to comping tracks and saving takes, decisions are made quickly.

Saving 20 takes per part may seem like a reasonable idea to many. What if you want a different variation on the part? Not sure the timing is locked? Not sure which take has the best tuning? What if? What if? What if?

Too many “what if’s” lead to a muddy production. It’s important to make decisions. Clarity throughout the process is important. Firstly, because it affects the performances.

A guitar chord that’s off is going to trigger the bass note to be off and then the percussionist has a hard time locking in. Before long, you have a session where the whole band is a little shaky. Not making decisive decisions can create a spiral effect on the stability of the production.

Momentary Lapse Of Reason
There is also the memory lapse effect. You record a bunch of takes and while you’re working, everything seems clear in your mind: Take 12 had a good bit, take 15 was mostly good, but you want to grab the beginning from take 4.

If you put the song down for a few days and come back to the session it’s going to be hard to remember the nuances between takes.

Commit. If it’s still not good enough, re-track it. At this point, you’re better off getting a single take then a patched edit for the sake of feel. I’m always in favor of replaying the part rather than extensive edits. It will take the same amount of time and the full take will still sound better.

Worm Hole
Aspiring producers/musicians get caught in the trap of playing too much and not listening. I like to set a rule of stopping after 4 takes and giving a really good listen. Don’t set record to do an endless loop. Loop recording means you’re not listening and most likely spacing out at times.

Perspective
It’s hard to hear the music the way it really sounds while you’re playing. This is another reason why you need to stop and listen as often as you can. If you’re the producer and player your perspective is biased.

When you stop, put your instrument down and trust your ears. Listen, make notes, and re-take. Don’t be noodling on your instrument while listening. This is the only way to make really fine adjustments. It may seem like it’s the long approach, but in reality, it will save you time.

Hit It
Here is how I like to track a vocal session.

First, I’ve taken time to choose the correct mic, preamp, compressor, incense, tea, lighting and dialed in a headphone mix. (Note: It’s very important to have a great headphone mix. It will result in less fatigue and frustration from the performer.)

Next, I like record a couple of full passes before we even think about punches. Let the performer get into the vibe of the song.

After 3-4 takes, stop. Take a few second break for water and then listen. Before we listen, I make sure we both have a pencil and paper. As we review each take, we write notes of what we liked or didn’t.

Listening to 8 takes in a row is overwhelming! It’s too much to digest. Plus, I’ve heard that if you listen to 9 takes in a row it could cause bowel irritation. Ok, I made that up. But, if I have to listen to 9 takes in a row of the 3rd part background vocal I’m going to be calling my friend Johnny Walker Red… And we’re gonna have a loooong chat, if ya know what I mean.

When the last take has completed playing, we compare notes and see if we have a comp. In the event the overall performance is not there, we repeat the 3-4 take run, break, then listen, take notes, comp.

If we just need a few bits, we comp the the take and punch in where needed. Notice I mention we comp BEFORE we punch!

Performance Drift
There is something I like to call “Performance Drift.” This is when the artists’ performance changes dramatically from the first take to the last. Volume, expression, and enthusiasm may have shifted during flight. Limiting tracking to 3-4 takes at a clip prevents performance drift as there will be breaks and reviewing that keeps it fresh.

Hash It Out
Don’t use recording as your practice. Need to review something because it’s not right? Stop playback and run it. Work it out. Be prepared and ready when the red light is on.

Don’t have the mindset of “I’ll fix it later.” The performance will always suffer. Even though we know comping and punching is an option, it’s good to pretend that it is not. A coherent take will always sound better.

Binary Composting
Don’t be a take-hoarder. Go ahead and delete! Don’t be afraid. Why live in the past, when you can be in the present? Last take only so-so? DELETE.

It’s also good idea to delete all unused audio from your sessions. It’s no use carrying around that baggage. No reason to have 20 gigs of audio that you’re not using. A bloated session is harder to backup or track down a file if need be. Plus, it takes longer to load.

If you’re not using it, send it off to greener pastures (aka your trash bin). Think of it as composting for 1’s and 0’s. Dare I say binary composting?!?!

Adios
Before I tell the musician a session is over, I make sure I have a comp I can live with. It should include all crossfades and edits cleaned. I want to know I have the part and what it sounds like. Leave nothing to the imagination…except which island your summer home will be on after your single blows up.

 
Mark Marshall is a producer, songwriter, session musician and instructor based in NYC.

Be sure to visit The Pro Audio Files for more great recording content. To comment or ask questions about this article, go here.

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Posted by Keith Clark on 12/17 at 04:05 PM
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Friday, December 13, 2013

Tech Tip Of The Day: Pre/Post Confusion

What is the difference between a post-fader and a pre-fader aux send and in what situations would I use either one?
Provided by Sweetwater.

 
Q: I was recruited to be on our church tech team, and I’m really glad I said yes.

I really enjoy this whole audio thing, however, there’s so much I don’t understand, but I know it’s a learning process. Anyway, everybody tells me I should just ask questions, so I guess one thing that’s tripping me up is all this pre/post stuff.

What’s the difference between a post-fader and a pre-fader aux send and in what situations would I use either one?

A: On a console, pre and post indicate possibilities to override the normal signal routing for various purposes.

For example, a post-fader aux send taps the incoming signal from the channel at a point after the channel fader. This means that when the channel fader is down, no signal is sent out the aux send(s) on that channel.

Post-fader aux sends are generally used as “effects sends,” to send a signal out from a particular channel to an effects processor.

Since the channel fader controls the level of signal being sent to the main mix as well as the level of signal being sent out the aux send, when the channel fades down, the level of the “wet” signal follows the level of the “dry” signal. If the level of the wet signal did not follow the level of the dry signal, the effect would still be heard after the channel fades out.

A pre-fader aux send taps the incoming signal from the channel at a point that is before the channel fader. So, when the channel fader is down, the signal is still being sent to the aux bus.

Pre-fader aux sends are helpful for live sound reinforcement situations where the FOH console is doubling as the stage monitor mix console. When setting up stage monitor mixes, it is ideal to be able to control the level of these mixes independently from the front-of-house mix.

If the position of the channel fader affected the level of that channel in each monitor mix, it would be necessary to constantly adjust the monitor mix (Aux) Sends after changing the level of the channel fader. Or more simply put, when a screaming guitar solo is boosted in the front-of-house mix, everybody on stage would get an earful from their monitor mix.

There is also another distinction known as pre or post EQ, which at this point should be fairly self-explanatory. Any pre fader send could still be pre or post EQ. In a live situation, pre fader and pre EQ sends are usually best where the mixer may be feeding stage monitors.

Additionally, there are options such as PFL (or pre-fade-listen), which generally sends a signal to monitor outputs regardless of the setting of that channel’s fader, and simultaneously mutes the other channels. In other words, PFL allows you to solo a channel even if the fader is pulled all the way down.

Note that on most consoles, this affects monitors only, and does not interfere with main, tape, or aux outs. In broadcast situations, PFL is often referred to as “cueing.”

For more tech tips go to Sweetwater.com

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Posted by Keith Clark on 12/13 at 10:23 AM
Church SoundFeatureBlogStudy HallConsolesEducationMixerSignalSystemTechnicianPermalink

Thursday, December 12, 2013

Perspective: Meeting The Challenges Of The Gig

Identifying your primary audience is key.

As a sound engineer working in the concert and corporate event markets, I’ve found it useful to identify my primary (most important) audience for every gig.

Is it the band that hired me? The band manager? The promoter? The people buying tickets to the show? The people that own the venue? The sound company I’m working for?

It can be tough to figure. The easy answer is you work for who pays you, but it can be a bit more complicated than that. Maybe the following experiences can help answer the question.

I toured with a particular artist for several years. We played medium-sized venues in larger cities (Roseland Ballroom in New York, The Warfield in San Francisco, etc.) in addition to being the support act on several arena tours.

At some point, the band started asking me to mix them “as loud as possible” (after completing a record with a producer who monitored at extremely high SPL).

They were already frequently too loud before I put them through the PA.

Sometimes I had a hard time getting the vocals above the ambient level of the guitars, even in larger venues.

I knew their fans - after all, they would talk to me at every show - and they didn’t like it.

They took the lyrics seriously, and would sing along the entire set. The clearer the vocals, the louder the fans sang (and the higher the energy levels in the room).

Although the band hired me, the fans really paid the bills, so I identified them as my primary audience.

If they weren’t happy, the band would eventually hear about it, so I worked to convince the band that the fans really wanted to hear the vocals and understand the words above all else.

Once they understood, the band stopped insisting on a punishing loud mix, and even began turning down their guitars if I couldn’t get the vocals audible over the stage volume.

Figuring out the primary audience at corporate events is even trickier because there are additional audience layers in play, such as event planners and clients.

A few years ago I traveled to Tampa to mix a band at an official NFL party tied into the Super Bowl - a large event (3,000 people) in the main hall of the Tampa convention center.

A local sound company provided a multi-zoned system, with main and delay line arrays, subs, and front fills.

All About Business
Although the band hired me, I’ve done enough events like this to know that the event planner calls the shots, often at the direction of the client paying the bills.

If the planner tells me to turn it down, I do - even if the band wants me to turn it up.

Luckily in this particular case, the band has done these types of events for years, so they understand that it’s all about business.

(As far as I can tell, the pecking order seems to be, in order of importance: attendees, food, interior design, floral, lighting, sound, entertainment.)

At sound check, the event planner asked me to turn up the volume.

I happily complied, knowing that once the party started I would almost certainly be asked to turn it down. (It’s usually best to keep this knowledge to yourself and let the situation play out rather than offer any resistance.)

I arranged my mix accordingly, putting the band on a VCA and inserting a compressor on the main stereo bus.

I also decided that, when asked to turn down the volume, I could decrease the level of the delays and main arrays while still maintaining the volume (and energy) on the dance floor by turning up the level in the front fills.

The Point
Sure enough, as soon as the band hit the stage, a woman I’d never seen before asked me to turn down the volume.

I said O.K., and politely asked her name, and then asked the system tech to radio the event planner and find out if the woman had the authority to make the request. The event planner said yes - the woman was the assistant to the NFL commissioner.

Here, finally, was my real main audience f o r the event. The person paying the bills for a corporate event wants less volume, then no problem.

I turned down the arrays a couple of notches and also took some 2-3 K bite out of them, then brightened up the overall mix (8 kHz-plus) a bit for clarity, and pushed up the volume in the front fills.

For the next three hours, the band played, the party-goers drank, ate, schmoozed (and finally started to dance), and I was left to actually mix the show instead of responding to requests to turn it up or down.

The event planner will most likely book the band again, I will most likely mix the band again, and we can all continue to make a living.

And that’s the point. To make a living working in sound, we often find ourselves having to do things that those paying the bills find enjoyable.

Do it, and politely, and you just might be asked back.

Nick Pellicciotto has worked in the live sound industry for over 15 years, touring as a mix engineer for acts like Fugazi, Lucinda Williams, and Modest Mouse.

{extended}
Posted by Keith Clark on 12/12 at 12:00 PM
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Wednesday, December 11, 2013

Tech Tip Of The Day: Getting Your Wire Gauge Right

How do I choose the right gauge of wire for loudspeakers?
Provided by Sweetwater.

 
Q: Despite my experience within the A/V industry (I wont say how long it’s been), I’ve never quite gotten the hang of everything. Admittedly, this is because I’m a bit more of a video expert, but I’d like to remedy that.

So I wanted to get clarification on a few things. The largest (and perhaps most important) being, how do I choose the correct gauge of wire for loudspeakers?

A: Selection of the appropriate wire gauge is important to system operation.

A cable that’s too “light” will result in amplifier power being wasted due to the series resistance of the cable. It will also result in the loss of low-frequency performance due to a degraded damping factor.

On the other hand, a cable that is too “heavy” is unnecessarily awkward and costly. In general you want to keep your line losses (“insertion” losses) below 0.5 dB (though some engineers would argue this is still too much loss).

The impedance of the load (speaker), the length of cable, the cable gauge, and to less extent the output impedance of the amplifier all play a role in how well the signal gets from the amp to the speaker. Essentially, distance and the impedance of the loudspeaker are the two factors to consider when determining wire gauge.

The following table shows the approximate signal losses in speaker cable for a 100-foot amplifier-to-speaker distance at various impedances:

10 AWG: 4 Ohm = .44 dB, 8 Ohm = .22 dB, 16 Ohm = .11 dB
12 AWG: 4 Ohm = .69 dB, 8 Ohm = .35 dB, 16 Ohm = .18 dB
14 AWG: 4 Ohm = 1.07 dB, 8 Ohm = .55 dB, 16 Ohm = .28 dB
16 AWG: 4 Ohm = 1.65 dB, 8 Ohm = .86 dB, 16 Ohm = .44 dB
18 AWG: 4 Ohm = 2.49 dB, 8 Ohm = 1.33 dB, 16 Ohm = .69 dB

As you can see, an 18-gauge cable with a 4-Ohm speaker at 100 feet results in 2.5 dB of loss. A loss of 3 dB would mean that half the amplifier’s power is being dissipated by the wire, not the speaker!

The following information comes from JBL. It shows some suggested wire gauges for different distances and different impedances.

• 10 feet, 4, 8 & 16 Ohm load = 20 AWG
• 25 feet, 4 Ohm load = 15 - 20 AWG
• 25 feet, 8 & 16 Ohm load = 20 AWG
• 50 feet, 4 Ohm load = 10 - 15 AWG
• 50 feet, 8 Ohm load = 15 AWG
• 50 feet, 16 Ohm load = 15 - 20 AWG
• 100 feet, 4 Ohm load = 10 AWG
• 100 feet, 8 Ohm load = 10 - 15 AWG
• 100 feet, 16 Ohm load = 15 - 18 AWG
• 150 feet, 4 Ohm load = 8 AWG
• 150 feet, 8 Ohm load = 12 AWG
• 150 feet, 16 Ohm load = 15 AWG
• 200 feet, 4 Ohm load = 5 - 8 AWG
• 200 feet, 8 Ohm load = 10 AWG
• 200 feet, 16 Ohm load = 10 - 15 AWG

Some engineers would argue these figures are too conservative, and in “real-world” applications a heavier gauge is needed for the best sound. Whether everyone agrees with these figures or not it should at least be understood that distance and impedance play a major role in how the wire reacts.

Further, in high power applications it may make sense to get much more “stingy” when it comes to power loss. For example, a “small” 0.5 dB loss at 1000 watts is still a loss of more than 100 watts of power! In the end, it’s probably best to shoot for higher grade, lower gauged wire in almost any circumstance for best results.

For more tech tips go to Sweetwater.com

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Posted by Keith Clark on 12/11 at 09:38 AM
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Thursday, December 05, 2013

Exceeding Standards: Stewart Independent Productions Puts It All Together

"We just decided to pull together everything we'd learned, the best people we'd worked with and the best gear we know of." -- Shannon Stewart

Working sound and lighting gigs together in high school, Shannon Stewart and Dan DeVisser didn’t really know where they were headed. But one thing was certain: they were already hooked on the production business.

Fast-forward 20 years to find the long-time friends heading up Stewart Independent Productions, a full-service national production company located in Southwest Michigan, that encompasses everything they’ve learned in their collective 40-plus years in the business.

“We just decided to pull together everything we’d learned, the best people we’d worked with and the best gear we know of,” explains Stewart. “And we succeeded. Twenty years later, we’re exactly what we wanted to be – a smaller service-oriented company with standards and ethics that emulate the big players in the industry.”

Stewart Independent has worked with dozens of national acts as a top regional supplier, including 10,000 Maniacs, Barenaked Ladies, Black Eyed Peas, Blues Traveler and many others. One area of particular focus is providing full-service production services to several well-known institutes of higher learning in the area, the most significant being the University of Notre Dame just across the state line in Indiana.

Long-time working relationships that both company principals had with the university led to it becoming one of their first clients, and it remains a staple of their business to this day. It’s also an experience that has help shape what their company represents.

“We really cut our teeth as production managers and as a full-service production management company at the University of Notre Dame,” Stewart explains. “Their pursuit of excellence and our drive for the same made us a perfect match from the beginning. During the last 17 years we have grown up together – it’s been one of the honors of my lifetime to be the production manager for student activities at the university.”

Stewart Independent staffer helping guest engineer with the Avid SC48 deployed at front of house at this year’s Notre Dame Block Party concert.

Learning Is Key
At the outset of each school year in late August, and before the craziness of student kick-off week, Stewart provides a full training session for Notre Dame student sound techs who will be working with the varied and many sound reinforcement systems installed throughout campus. These same techs also join the Stewart Independent crew when they’re onsite for larger productions.

“We provide an introduction to pro audio that we call Practical Application of Live Sound Reinforcement,” Stewart says. “It’s one of my jobs to insure that the university has qualified student audio techs available to handle smaller events. We also offer Practical Applications of Live Concert Production for those interested in lighting, staging and video. And, students continue to receive hands-on training throughout the year.”

In the training courses, Stewart details the basics of audio and ultimately how to use and troubleshoot a system. Another key aspect focuses on how to interact with visiting production crew as well as working in a professional manner and maintaining a positive attitude.

“It’s great to get the opportunity to train people the right way, long before they’ve had the chance to develop what we consider to be bad habits,” he notes. “I’m pleased to say that many we’ve interfaced with have been asked to work for us on the road, and some of them are still with us.”

Kicking Off
In addition to the educational efforts, that first week marks the beginning of nine very busy months for Stewart Independent at the university.

“It’s always a little crazy but we keep it well organized,” Stewart notes with a laugh. “There are tons of activities designed to welcome the new and returning students and many of them require sound systems, stages, lighting and even video. It’s exciting, but a lot of work.”

He and his team set up temporary office space near the university’s world-famous football stadium, making sure they have “boots on the ground” to meet any special requests while also mapping out and refining the approach for a range of highly trafficked events.

The company plays it smart, deploying variations of the same sound reinforcement elements for the majority of live events held during kick-off week. Go-to components include RCF line arrays, subwoofers and monitors, as well as Avid VENUE SC48 digital consoles.

“We started using RCF a few years ago when we were looking for a self-powered and processed single 18-inch subwoofer for monitor applications,” Stewart says. “Our goal was to be able to create multiple subwoofer configurations. We ended up astounded with the sheer output and punch of the TTS18-A subs, so we added a few for monitor applications and a few more for PA.”

Sound check for the Block Party, with RCF monitors, a Stewart Independent staple, deployed on stage.

That led to TT25-SMA floor monitors, providing a tight 40-degree by 40-degree coverage pattern that’s desirable in several applications, followed by TT45-SMA monitors loaded with double 12-inch woofers that handle wider coverage needs. Staying on stage, next up were HDL 20-A line array modules for side fill applications.

“They sounded so great out of the box, horn loaded, almost 100 percent weatherproof, and so easy to fly that we ended up getting enough so that we would have a great powered PA line array in house as well as killer side fills,” he explains. “It sounds terrific and provides exceptionally long throw, with the reviews from those who’ve used being stellar. It’s now our ‘go-to’ system for just about everything.”

Members of the Stewart Independent team onsite at a project, including (left to right)  Christian Chambers, Sam Skalbeck, Shannon Stewart, Dan DeVisser, Austin Lanning, Joe Watrac, Ross Labardee. Long-time team members Mati Johnson and Scott Frost were also working the event but were not available for the photo opp.


The SC48 digital consoles see constant use, cited for ease of use, familiarity among a wide range of engineers, and the assortment of available plug-ins. “It is amazing that we can have all of the effects from our analog days in our digital consoles,” Stewart says. “I just love these boards.”

Blowing The Roof Off
Let’s take a look at how the Stewart Independent team deploys that gear, starting with Domer Fest, an event for first year students that features a mixer and dance in a field house with free standing tents and activities just outside. The main stage is outfitted with four individual DJ packages, accompanied by a lighting rig that could be found at higher end night clubs.

The four DJ rigs are mixed down to an SC48 at stage left, and from there, signal goes to HDL 20-A main arrays. Add TT25-SMA monitors for in fill and eight TTS18-A subwoofers to deliver the serious low end that the applications requires, and as Stewart notes, “we’re ready to blow the roof off the place.”

Next up, the team and many of these components move along to the opening of the Academic Year Mass Picnic, an evening event for 7,000 that takes place on a large campus quad, featuring live music.

This is followed by a large outdoor comedy show as well as starting load-in for the culmination of kick-off week, the B1 Block Party. “The student techs receive a true taste of live event production during that first week. It’s almost a baptism by fire,” Stewart chuckles.

The comedy show requires setting up a hydraulic Stageline SL100 stage, as well as a Barco B10 video wall and a significant complement of RCF loudspeakers joined by an SC48 console. At the same time, another Stageline stage, this time a SL320, and two Barco video walls for the B1 Block Party need to be put together outside the football stadium.

Meeting Expectations
This year’s Block Party reinforcement system, serving up a national artist as well as a variety of top regional artists for several thousand in attendance, was significant in scale. As a result, Edge ShowTek of Chicago was contracted to fulfill Stewart Independent’s design calling for 12-deep NEXO GEO D line arrays for mains, flown left and right, and joined by NEXO RS18 subwoofers in mono blocks on the deck.

The Block Party by day during setup, and later at full throttle.

Several HDL 20-A array modules were posted on stage to provide stage fill, with performers served by several TT25 and TT45 wedges positioned as needed. The national act had Yamaha consoles at its disposal, including PM5DRH at front of house and an M7CL for monitors. SC48s did the same for the regional acts.

“This system worked really well. Particularly for national acts, the audience expects it to be loud,” Stewart notes. “But because it was held outside and there are residential areas nearby, the sound also needed to be contained. What we were able to attain were the expected concert levels volume and punch that dropped off where we needed it to.” 

The Block Party marks the culmination of a very hectic, concentrated period of activity for professional and student tech teams alike, but the time for breathing room is brief, with the university launching into a steady stream of events, programs and more for the next several months.

“It’s vital that our systems, as well as the way we implement them, be done with the right combination of quality and efficiency,” Stewart concludes. “We must allow enough time to insure a high level of service- and face-time with the client and it’s constituents. They have very high standards, and exceeding those standards is something we’re very proud of.”

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Posted by Keith Clark on 12/05 at 10:54 AM
Live SoundFeatureBlogStudy HallBusinessConcertConsolesEngineerLine ArrayLoudspeakerMixerMonitoringSound ReinforcementSubwooferTechnicianPermalink

Now Available On Video: Audio-Technica “Ask Me Anything” Sessions At 135 AES Convention

Audio-Technica’s “Ask Me Anything” (AMA) sessions at the recent 135th AES Convention in New York are now available for viewing on the company website here. (Note that the session with Grammy Award winning producer/engineer Frank Filipetti is presented below.)

“Ask Me Anything” was held at the A-T booth on the AES show floor. Questions were submitted by AES attendees, as well as online at www.livestream.com and via Twitter at #ATliveAES – and were then relayed to the presenters through a moderator during the 30-minute Q&A sessions. 

The sessions include discussions at the A-T booth at AES with:

—Joel Singer, Grammy Award-winning engineer/mixer, co-founder and chief engineer of Music Mix Mobile

—Jackie Green, VP R&D/Engineering at Audio-Technica

—Frank Filipetti, Grammy Award-winning music producer, engineer and mixer

—Richie Castellano, musician and YouTube sensation

—Richard Chycki, mixer, engineer and producer

—Carl Tatz, TEC Award-nominated recording studio designer

—Frank Wells, then-current president of AES

—And Jimmy Douglass, Grammy Award-winning recording engineer/record producer

“We were extremely happy with the AMA sessions in the A-T booth at the AES show this year,” states Gary Boss, Audio-Technica marketing director. “Not only was this a unique opportunity for both attendees at the show and those at home to ask insightful questions, but the content will be archived for a whole new audience to benefit from all of our presenters’ wisdom and insights.

“There were laughs, thought provoking answers and a few uncomfortable moments.  Exactly the kind of scenario where we get to see the true genius behind our guests and understand why they are at the tops of their respective fields.” 

The Frank Filipetti session follows, and go here to check out the full series.

 

 

Audio-Technica

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Posted by Keith Clark on 12/05 at 08:26 AM
Live SoundRecordingChurch SoundNewsVideoTrainingBusinessEducationEngineerSound ReinforcementStudioTechnicianPermalink

Church Sound: Twelve Steps To Christmas Program Survival

Critical tips to keep in mind as we wade into the most hectic season of the year...

‘Tis the season ... In honor of the classic holiday song “12 Days of Christmas,” if you’re in the midst of Christmas programs right now, please take advantage of at least some of the tips offered in the following “12 steps” survival guide.

12: Start organized with a plan

You know the saying, fail to plan and you can plan on failing. Guess what, it’s true. Talent will only carry you so far. 

The really great musician, technicians and artists know how to make things happen. Whether it’s a written plan (which I recommend) or just a mental plan you’ve thought through in advance, the process of planning can make all the difference is both the success of the event and your own sanity.

11: Check your gear to make sure it’s all in working order

This goes with planning. There’s nothing more frustrating than pulling a bad mic cable, or using a broken mic that hasn’t worked for six months, or suddenly needing an extra console channel only to find that the only one left doesn’t work.

10: Work ahead, work the plan

Think ahead and do the tasks ahead of time that you can. For instance, checking gear two weeks in advance leaves ample time to get items repaired or replaced.

9: Identify what’s really important and focus on those things

Don’t get caught up focusing on that “cool effect” you want for one song and miss the more important stuff, like doing a line check before the band shows up! 

A couple of years ago, this one almost got me, but fortunately sanity prevailed and I gave up on the “really cool” edge-blended video screen backdrop that captured way too much of my attention for a few days. 

I was ignoring the truly important things, like making sure the main P.A. was in top shape and getting the lighting cues recorded, aspects that are much more important and ultimately led to a successful program.

In the end, nobody but me (and my tireless volunteer Wayne) even knew we weren’t deploying the super-cool video thing.

8: Invest in those around you

Tech folks and musicians can get so focused on the tasks at hand that we tend to forget about those around us. Instead, use this time to let some of the less experienced folks shadow you. Teach them by showing what you’re doing and explaining why.

And it never hurts to bring chocolate…

7: Have fun and smile

Everybody wants to be on a winning—not whining—team. If you win overall, people will come back and continue to work and volunteer, pouring their hearts into the event.

6: Be flexible

Stuff happens. It just does. At a Michael Card concert I worked, the piano player became very ill the day prior to the concert. Rather than panic, the promoter of the event recruited a very gifted pianist to sit in. 

Was it ideal? Was it what Michael wanted? Did it turn out great? (The answers in order are NO, NO, YES)

The pianist hit the ball out of the park, he sight-read the music during rehearsal, practiced between rehearsal and show time, and absolutely nailed it!

5: Know when to say “no”

OK, back to that “cool” edge-blended video backdrop… Sometimes you have to just reel it in and say no and move on. I always say that it’s better to do 75 percent of the program (cut out the last 25 percent) at 100 percent quality rather that 100 percent of the show at 75 percent quality.

4: Pace yourself

The older I get, the more important this becomes. The adrenaline rush is great, but the crash after it is terrible.

Know your limits—take breaks, eat healthy (and regularly scheduled) meals, go for walks, take some “chill time” when things hit a fever pitch…

3: Don’t overdo the caffeine

My overall intake of caffeine tends to spike around Christmas production time. The short term gain in energy is not worth it in the long haul. (Althought I must admit that sometimes I forget this one…)

2: Ask for help, call an expert

Why do we hate to do this? Almost every time I break down and call tech support or ask an informed friend to help out, the problem gets fixed rather quickly, and then I’m invariably left asking myself why I wasted five hours before making the call.

1:Maintain the right spirit

We can’t give what we don’t have, so if we don’t have the right spirit, we will fail. This comes in the form of being lousy to work with while not being of help and inspiration to others. We must show up ready to serve.

I hope your Christmas productions go off without a hitch, although realistically, that’s hardly ever the case. Keeping this survival guide in mind can help make things better from a technical standpoint, but more importantly, it can help us enjoy and appreciate the spirit of the season, and this transmits to those around us. And that’s the real point.

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound and production at his church for more than 30 years.

{extended}
Posted by Keith Clark on 12/05 at 08:21 AM
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Wednesday, December 04, 2013

Church Sound: Transitioning From Analog To Digital Mixing

A whole new way of doing the same old things
This article is provided by Gowing Associates.

 
I’m in the process of helping one of my churches transition from an analog mixer to a digital mixer.

They were in need of more channels than their Allen & Heath 16-channel MixWiz with some outboard gear (front of house EQ, couple of compressors, effects unit) could provide.

Based on the maximum number of channels that they anticipated needing over the next five years, I recommended the PreSonus StudioLive 24.4, one of the least expensive 24-channel digital mixers on the market.

The church has two audio volunteers that are pretty much average in their knowledge of sound and sound systems so this would be a typical transition for a lot of churches in the 100-400 person attendance range. Volunteers selected more for their willingness to serve than their knowledge of audio. I know that nothing has been touched with the front of house EQ, compressors and FX since I helped them set it up about a year ago.

Some things that you need to consider in this transition is how uncomfortable the volunteers are going to be until they make the paradigm switch from the analog WYSIWYG (what you see is what you get) to the digital layers.

Depending on the digital board, layers control everything from different grouping of faders (1-8, 9-16, etc) to control over the aux sends, FX, etc. Outboard gear usually goes away and everything is now handled with the digital mixer. It’s a big transition and you shouldn’t minimize it, but treat it with care and planning and the transition will go smoothly.

Getting Started
What I recommend is that the digital mixer not be put into service immediately but be brought into a two-to-four-week training duty cycle. It requires some mics and cables as well as a couple of speakers for monitors and front of house stand-ins. If you have instruments that you can plug in that helps as well. Keep the existing analog system going as the production system until everyone has been trained and is comfortable with the digital board.

Before you start with the digital mixer, make sure everyone has reviewed the user manual. A digital board is a computer with knobs and faders and is significantly more complex than an analog mixer. While they are pretty robust, you can still mess them up and repairs can be costly.

An Investment of Protection
One thing to invest in if you haven’t is a top-line power conditioner like those from Furman. I also recommend a computer UPS (battery backup) from a company like APC or Tripp Lite. Get a decent capacity one. The reason is that because a digital mixer is a computer, when power is interrupted you can’t just switch it back on like an analog mixer. You need to boot it up and, depending on the mixer, that could take anywhere from a minute to several minutes.

Having a UPS unit, the mixer will stay powered on, so even if the rest of the system is knocked offline by the power interruption, when the power comes back on, the mixer will still be up.

Unboxing The Mixer
Once you get the mixer unboxed, check for any damage. If everything looks good bring all faders down to minimum and turn on the mixer. I like to let the mixer “burn in” for about four hours with nothing going on or plugged in just to let all the electronics warm up to full operating temperature. This will check to ensure that nothing is shorting out. Be aware of any burning electrical smell or smoke. If you detect either one shut the mixer down immediately and unplug it. Contact the vendor.

Preparing For Training
The StudioLive is close to an analog board in that all the channel faders are on one surface as opposed to layers. This makes the transition somewhat easier. All effects, aux send levels are controlled through the center “Fat Channel.” That will be where most of the confusion is going to come in so be prepared to spend a lot of time going through this area.

The StudioLive is set up pretty easy so I was able to figure 85% of the board out without looking at the manual. There are also a ton of video tutorials on the PreSonus site and YouTube that can help with anything to do with the board. But for volunteer sound techs it will be a bit of a challenge.

Building A Mini-System
Hook up a mic to channel 1 on the mixer and hook up a speaker to aux send 1 and to front of house. This will be the basic training setup.

Once you get it hooked up, bring up the gain to an appropriate level. A digital board is less forgiving about exceeding the 0 level than an analog board before going into clipping so run the level less than needed for training until you get comfortable with the way the board handles signals.

Don’t worry about EQ settings or FX yet. All you want to do is to learn the signal flow from the channel to the aux send and FOH.

Once you’ve figured out how to adjust the aux send levels for the channel and you can adjust FOH level you’ve gotten over the initial hump.

Using EQ
The next thing you’ll want to learn is how to adjust EQ’s for each channel. Depending on the digital mixer you’ll either have a screen that will have a parametric equalizer, or in the case of the StudioLive, you’ll have the knob adjustments for high, high mid, low mid and low bands. As with all digital mixers you are able to set the frequency points for all these bands as well as the Q, which is the width of the frequency adjustment. This is a lot more adjustability than what an analog mixer has and is worth spending some time practicing.

After the channel EQs get figured out you’ll want to adjust the front of house EQ. On the StudioLive it’s set the same way that the individual channel EQs are set. One nice advantage about digital mixers is that most of them have a library of preset EQs that you can start with. The StudioLive has built in a nice set of professional quality EQ presets that are good enough to leave alone and assign to each channel.

The other nice feature of digital boards is the ability to save all your settings to a scene. So you are able to set up multiple scenes for different worship teams or different instruments and recall them just by dialing up the scene and pressing the load button. So no more needing to reserve channels based on who’s playing that day.

Enter Effects
The power of digital mixers means that you can assign FX to each and every channel, both to auxes and to front of house, so you’ve got a lot of flexibility. Just remember that just because you can doesn’t mean you should. Less is more, at least in the beginning. Some boards give you more FX capabilities than others. The StudioLive offers two channels of FX, others more.

Multi-track Recording
Another advantage that digital mixers have is that they usually provide some form of multi-track recording capability. In the case of the StudioLive, it’s provided by a FireWire port into the provided Studio One software. This means you can record each channel separately into your computer, as long as it has a Firewire port.

One very cool reason for doing this for the worship team is the ability to do what’s called a Virtual Sound Check. What that means is that you don’t need the worship team there to set up the board. You can play back the individual tracks back into their respective board channels and use those tracks as the sound check.

Then, once the band gets in, sound check is very minimal. It’s also a great way for the sound team to train on the board and allows them to massage settings without needing the musicians.

Saving Scenes
Once you get everything set the way you want it remember to save your settings to a scene. I usually recommend naming the scene with the church name and 1. That way you can always recover your baseline settings.

Sound techs should create their own “sandbox” scene, which allows them to manipulate settings and save it to their own scene without affecting the master scene. Make sure that no one other than the lead sound tech saves to the master scene.

Once you’ve got the master scene saved it won’t matter what changes people make to the board during the week. Bringing back the master scene will only require a quick push of a button, and in the case of the StudioLive, resetting the gain and adjusting the faders. In other digital boards, gain settings and fader positions are saved within the scene.

Making The Switch
Once the sound techs are comfortable with the digital board then it’s time to switch out the old analog board with the new digital one. Check all your settings. Be sure any settings you change are saved to the master scene once you’re happy with how everything sounds.

Finally, when you shut things down, do NOT shut things down by just turning off the power conditioner. This WILL damage the digital mixer. Follow the shutdown procedure in the manual. It can be anything from just powering off the mixer with the mixer’s power switch to a shut-down procedure on the screen.

Summary
A digital mixer is a whole new way of doing the same old things. It’s exciting as well as terrifying for volunteers, so go slow. Take it one step at a time and ensure they are comfortable with the new system before putting it into production. You’ll achieve a seamless transition and have fun doing it!

Brian Gowing has helped over 30 churches meet their technology requirements. Brian works towards shepherding the church, analyzing their technical requirements, sourcing the equipment, installing the equipment and training the volunteer personnel.  As he likes to say, “equipping the saints with technology to help spread the Good News.” Contact Brian here.

{extended}
Posted by Keith Clark on 12/04 at 10:53 AM
Church SoundFeaturePollAnalogConsolesDigitalEngineerSound ReinforcementTechnicianPermalink
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