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Thursday, April 26, 2012

Peavey MediaMatrix NION Training Seminar Scheduled For May

Peavey Commercial Audio will host a two-day MediaMatrix NION Certification training course in New Orleans during May, where AV designers, consultants, contractors and end users can learn best practices for designing, deploying and implementing MediaMatrix audio distribution, processing and control systems.

The MediaMatrix NION Certification training course instructs AV professionals on the fundamentals of MediaMatrix, the most flexible and scalable audio networking system on the market, as well as how to design and program projects in nWare; how to set up a NION processor; how to create end-user GUIs for nTouch 60 and nTouch 180 touch screens and PC kiosks; and how to integrate the XControl into a MediaMatrix installation.

The MediaMatrix NION Certification training seminar will be held in New Orleans on May 10-11, from 8:30 a.m. to 5 p.m. each day.

Each successful student will receive a completion certificate that can be submitted to InfoComm for receipt of 7.5 renewal credit hours for InfoComm CTS, CTS-I and CTS-D. For full class descriptions and registration, go here.

Peavey has educated thousands of AV system designers, integrators and end users from around the world since MediaMatrix revolutionized the professional audio industry in 1993.

MediaMatrix now offers the content of its renowned seminars as online courses here.

Completion of the online Peavey MediaMatrix Basic or Advanced course earns two hours of credit toward InfoComm’s CTS and CTS-D Certification Renewal.

Learn how MediaMatrix can be used to create virtually any audio system. For more information, go here.

MediaMatrix
Peavey

Wednesday, April 25, 2012

Differences, Cause & Effect And Consequences Of Polarity And Phase

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.

Tuesday, April 24, 2012

Church Sound: The Sound Operator’s True Purpose And Role In Worship

What am I trying to accomplish?

It’s a question we should probably be asking ourselves, as church sound operators, on a daily basis.

In today’s rapid-paced society, where exciting new changes are mixed with fear and turmoil, the result can be a feeling of spinning in circles with no direction or purpose.

This is quite visible at many churches. It’s almost become a marketing battle as to who can do the most to keep people coming in the doors.

As a result, some are willing to compromise beliefs and convictions in favor or appearing “relevant” to mainstream society.

But shouldn’t it be the other way around? Shouldn’t the church be teaching it’s people to make their lifestyle and the “real world” relevant to their faith?

Now let’s narrow this discussion down to those of us who support the technical side of ministry. How can we help a ministry stay on track?

Imagine making a trip by boat from England to New York City. The course is set and you’re on your way, but somehow, the direction of the boat is just a couple of degrees off.

Where do you end up? Most certainly not where you wanted to go.

Working in ministry can be the same way. The problem is that sometimes we don’t realize we’re off-track until we arrive at the wrong place. I’ve come to observe that church sound and technical volunteers (as well as their paid counterparts) face constant danger of going off course.

Who Knows What
Why is this so? Simply, it’s very easy to get caught up in new gear, technologies and theories.

These types of things are always changing, and there are hundreds of different opinions floating around about all of it.

And it may be hard to admit, but it’s true: these are the things that most certainly are not the secret to success in audio ministry.

Don’t get me wrong - quality equipment, properly applied, is essential. Returning to my trip analogy, it sure would be a tough journey to New York if you tried to make it on a surfboard. The same goes for sound systems.

But our job is to be educated about show knows what they’re talking about, and who can provide us with the right tools - not the latest and coolest tools.

Success in all endeavors is greatly determined by whom we associate with. Choose wisely, not just technically or economically. Unfortunately, those who may purport to give advice about sound sometimes doesn’t even know what they don’t know.

We must understand our role within ministry as a whole, and be completely committed to it with excellence. Christianity suffers today because it is horribly misunderstood, mostly due to the fact that it’s poorly communicated.

We all agree that church should always be a venue in which to communicate the truth of Christianity.

But if one attends church and the message is not clearly communicated, we are off track.

Those delivering the message have the obvious responsibility of making sure it is consistently correct.

These folks, however, are dependent on the sound operator to make sure the message is delivered clearly to the congregation.

By the way, when I use the word “message,” I’m not simply referring to the sermon.” Worship and music themselves are also part of the overall message.

Our role is to make sure that all of this is accurately amplified, helping the congregation join in the worship experience. A colleague once compared running sound for a rock concert to running sound in church as the difference between mixing for an audience and mixing with an audience.

Therefore, every time we are at work at the console, or plugging in a microphone on the platform, we must ask ourselves what is our role, and what are we really trying to accomplish?

Role Playing
A high level of value must be placed on attitude. All too often, I’ve witnessed a frightening level of alienation between worship leaders, musicians, pastors and the “tech folks.”

Many view the role of sound operator as one that should be invisible. However, tho those of us actually working with sound, that role is far from invisible.

The best way to change this problem is to change our attitude, always look at at things from everyone’s perspective, rather than just our own.

At the same time, sound operators should never view their role as secondary or insignificant, regardless of how anyone else sees it. But if the worship leader calls it a support role, that’s OK, because is a support role to and for that individual.

It doesn’t make any difference who gets the credit, or who gets talked about at Sunday dinner. What the sound operator does matters, even if others don’t recognize it.

Our most important mission remains the same. Every time we take our position of supporting worship at our church, the only question we need to ask is simple: “What am I trying to accomplish?”

The answer will always show us the way.

Rob Stam has served as an AV system designer and installer for more than 20 years, and has been active as both a musician and sound operator as a church member.

Monday, April 23, 2012

DiGiCo’s Training Seminars Reach Russia

As important to digital console manufacturer DiGiCo as the quality of its products is the support and education it provides its customers.

The company runs a series of well-attended training seminars both at its UK headquarters and around the globe.

The latest location added to its roster is Russia, with its first seminar for the territory held in February at the recording and performance studios of the famous artist Igor Sandler by its Russian distributor, Aris Pro.

DiGiCo’s Vice President of Sales Ian Staddon was joined by highly respected engineer Dave Bracey, known for his work as Front of House engineer for artists such as Robbie Williams, Massive Attack and Bjork, with whom he is currently touring.

The evening prior to the seminar saw Ian and Dave interviewed live on Igor Sandler’s famous late night rock radio show, which, according to the network, had some 1.4 million Muscovite’s watching the live web feed.

On the day, the pair presented a mixture of technical theory and hands on practice, with the seminars covering DiGiCo’s entire range of products, from the flagship SD7 to UB MADI, the latest ground-breaking offering from the company’s Solutions division. The sessions were rounded off with a Mixing Master Class from Bracey, where he demonstrated why the SD7 is his console of choice and emphasizing the use and value of features such as snapshots when using digital consoles.

As a finale to the event, all the attendees were treated to an enjoyable evening of food, drink and a live performance by an AC/DC cover band mixed on the SD7.

“The seminar was well organized, and I had a lot of time to discuss my questions with the DiGiCo and WAVES specialists,” recalls Alexander Odelevsky,Head of the Sound Reinforcement department
at The Moscow International House of Music. “We were able to fully explore the consoles. For new users it was a great experience to be able to study on DiGiCo consoles, and a great opportunity to address a few challenges and discuss some deep hardware and software nuances for pro sound engineers like me, who are experienced in DiGiCo consoles. Thanks to the DiGiCo/WAVES/Aris Pro teams for a great event.”

“Everybody really liked the seminar,” says Yaroslav Udovik, managing director of Aris Pro. “In fact, we plan to do more, as we have identified a need to provide more information to the end users in the Russian market.”

“It’s fantastic to be able to support exciting markets in this way,” adds Staddon. “The feedback we’ve had from the attendees was extremely encouraging and we’re looking forward to a return visit in the near future.”

DiGiCo

Guitar Center Opens Two New GC Studios Facilities

In the first quarter of 2012, Guitar Center expanded operations at two of its stores – Highland Park, Illinois, and Tucson, Arizona – by adding Guitar Center Studios, an in-house, state-of-the-art lesson facility, which will create unrivaled opportunities for area musicians of all ages and skill levels.

Guitar Center Studios provides music lessons from beginner to advanced featuring certified instructors teaching world-class curriculum as well as one-on-one courses on Pro Tools, Logic Pro and GarageBand. Guitar Center Studios is now the most modern and affordable lessons facility in the area.

“The opening of our new GC Studios at our Highland Park and Tucson stores is an important moment for musicians in these areas,” commented Gene Joly, Guitar Center Executive VP of Stores. “As arts programs are consistently being downsized at schools across the country, we feel it’s important to create these opportunities for the next generation of musicians.”

“Many of our recent store openings over the last year-plus have featured GC Studios, and those facilities have been a huge success across the board. We look forward to serving the music communities of these areas in this increased capacity.”

Guitar Center Highland Park is open seven days a week. Store hours are 11 a.m. to 9 p.m. Monday through Thursday, 10 a.m. to 9 p.m. Friday, 10 a.m. to 8 p.m. Saturday and 11 a.m. to 6 p.m. Sunday.

Guitar Center Tucson is open seven days a week. Store hours are 10 a.m. to 8 p.m. Monday through Friday, 10 a.m. to 7 p.m. Saturday and 11 a.m. to 6 p.m. Sunday. Guitar Center Studios hours at both locations follow the same schedule as their respective stores.

Guitar Center

Thursday, April 19, 2012

Enter To Win Free Admission To An Upcoming Dave Rat Audio Seminar

Dave Rat, owner of Rat Sound Systems, long-time mix engineer, product designer, and noted pro audio author, is presenting audio educational seminars that are being held regionally as he travels as front of house engineer for the ongoing Red Hot Chili Peppers tour.

Find tour more about the seminars here.

Dave’s offering the PSW/LAB community the chance to attend an upcoming seminar -- free of charge -- at any of the following tour stops in the month of May:

For your chance to win, simply enter our drawing below.

The winner, who gets to choose the site/session to attend, will be selected in advance of the New York/Newark session on May 4.

Monday, April 16, 2012

Harman Offering Regional HiQnet System Architect & JBL HiQnet Performance Manager Training Courses

In association with rep firm Robert Louis Associates, Harman’s System Development and Integration Group (SDIG) has announced in-person training sessions for HiQnet System Architect and JBL HiQnet Performance Manager software in Pennsylvania and Ohio.

The training will be lead by Harman product application specialist Emilian Wojtowycz, an expert in the implementation of System Architect and Performance Manger for installed and live sound applications.

The training sessions will take place at the following dates and locations:

April 30 – May 1: HiQnet System Architect, Pittsburgh, PA

May 2: JBL HiQnet Performance Manager, Columbus, OH

May 3 – May 4: HiQnet System Architect, Columbus, OH

HiQnet is a communications protocol that enables all the compatible devices in the audio signal path, from mixing consoles to loudspeakers, to seamlessly communicate with each other. 

System Architect 3 is the software used to set up and configure a HiQnet system, by means of a series of control panels that are displayed on a PC and can be customized by the user.

System Architect enables designers to configure and control an installed sound system, using the HiQnet communications protocol that enables all the compatible devices in the audio signal path, from mixing consoles to loudspeakers, to communicate with each other seamlessly. 

System Architect 3 features a system design philosophy centered on workflow and the use of a diagrammatic representation of the installed or live sound venue. Devices are arranged by both their physical and logical placement allowing the designer to ‘educate’ System Architect about how they are to be used. In return the software is able to provide automation of many of the laborious system design tasks for free.

Attendees of the System Architect courses will learn:

·    Design workflow modes

·    Overview of Ethernet AVB technology and AVB routing

·    Comprehensive design workflow and system tools

·    Custom and master panel creation

·    Going online and networking

·    Day-to-day operation

Both System Architect courses award attendees with 5.5 InfoComm CTS RU credits.

JBL HiQnet Performance Manager is a software application derived from the System Architect core code but tailored for live sound operation. It is designed to configure networked audio systems within performance venues such as theatre, house of worship, and corporate and other performance sound events.

Attendees of the Performance Manager course will learn the way in which Performance Manager guides the configuration of system design workflow:

·    Working with array templates and the JBL Line Array Calculator II tool

·    Adding passive or powered speakers automatically

·    Adding and associating amplifier racks automatically

·    Simplified drag-and-drop networking

·    Using the built-in test, tuning and calibration control interfaces

·    Running and monitoring the system with the dedicated show mode

“System Architect and Performance Manager are two powerful tools that make the design, setup, and tuning of audio systems faster and more efficient than ever before. These training courses will provide attendees with the knowledge to harness these resources for their everyday use,” stated Adam Holladay, market manager, Harman SDIG.

To apply for a course, go here on the Robert Louis Associates website. 

Harman HiQnet
Harman

First Yamaha CL Console Field Demo Attracts Crowd At ACIR Professional

With more than 60 audio professionals in attendance, sponsor and Yamaha dealer ACIR Professional in New Jersey deemed the first field demo of the new Yamaha CL Console series a success.

Yamaha systems application engineer Kevin Kimmel presented the new CL Series Digital Console along with the Portico RND 5045 Primary Source Enhancer and Dugan-MY16 card.

Audio professionals present were from the touring, house of worship and performance venue markets and hailed from the Atlantic City area, and areas such as central New Jersey, New York City, and Maryland.

“We are quite pleased with the success of this event,” states Bobby Harper, vice president of sales at ACIR Professional. “The CL console will fill a price point void, provide a new audio transmission platform, and also solve the problems of sharing digital head amps. We know that both Yamaha and ACIR Professional will do quite well in the sales and marketing efforts of the new console.”

Yamaha

Thursday, April 12, 2012

Registration Opens For AES Conference On Music-Induced Hearing Disorders

The Audio Engineering Society (AES) has confirmed that registration is now open for its 47th International Conference on Music-Induced Hearing Disorders, which will take place at Columbia College in downtown Chicago, June 20-22, 2012.

The conference presents expert knowledge from audio engineers, academic researchers, medical experts and cutting-edge manufacturers, with a total of 18 papers being presented over two days.

“The conference is a great opportunity for people to learn a wide array of perspectives on hearing health in the music industry,” says Michael Santucci, conference chair and president of Sensaphonics Hearing Conservation, one of the conference sponsors. “We have several presenters coming in from Europe, along with experts from several U.S. universities and manufacturers.

“This is a great opportunity for AES members to gain critical knowledge on the issue of hearing health in the music industry, and to network with the leading experts in the field.”

The papers being presented span a wide range of topics relevant to the music industry, including measurement techniques for in-ear monitors and portable music device, new research in measurement and diagnosis of hearing problems, and new hearing health products. In addition to the papers being presented, the conference will also have trade show booths from its platinum sponsors.

Full program details and secure online registration are now available here.

Costs for the AES 47th International Conference on Music Induced Hearing Disorders are $600 for AES members, $700 for non-members, and $300 for students, and includes conference attendance, premium on-site catering (two light breakfasts, two lunches, one full dinner) and related social events.

Hotel room blocks at attractive rates have been reserved for conference attendees at two nearby downtown Chicago hotels. Attendees are encouraged to reserve their spot early to ensure availability.

AES
Sensaphonics

Wednesday, April 11, 2012

Sound Wave Propagation: The Bigger Picture Of What We Hear

Previously (here and here), we’ve been looking at sound on a “microscopic” level, examining particle motion as sound propagates through air.

This time, let’s look at a larger picture of sound wave propagation.

A vibrating object will disturb the surrounding air medium causing localized changes in pressure and particle displacement with the transference of acoustical energy in the form of a sound wave.

Waves can be broadly classified as being either transverse or longitudinal. The distinction for each wave type corresponds to the relationship to which direction the particles in the medium move relative to the direction the wave moves.

For a transverse wave, the particles of the medium move at right angles to the direction of the wave propagation.

Examples of transverse waves include ocean waves, vibrating strings on a musical instrument, and light and other electro-magnetic radiation.

For a longitudinal wave the direction of the wave propagation is parallel (same direction) as the motion of the medium particles.

Sound waves in air are unique in that they propagate as longitudinal waves. Figure 1 shows concepts for transverse and longitudinal wave motion.

Figure 1. Transverse (top) and longitudinal wave motion (bottom) showing transmission medium motion and wave propagation direction. Image credit: Sennheiser. (click to enlarge)

A sound wave in air will propagate away from a source as a wavefront with the speed of 343 m/s. The sound particle vibrations travel outward from the source with the same phase. Sources can take the form of simple geometry, called point or monopole (e.g., single loudspeaker or small hole in a wall), line (e.g., moving vehicular traffic or many closely spaced loudspeakers), plane (e.g., top plate of double bass, large building surfaces, or large two dimensional loudspeaker array), or complex comprising two or more simple sources (e.g., musical instrument or machine).

The wavefront takes on a spherical geometry at a distance from the source that is larger than the source dimensions and are called spherical waves. At farther distances the wavefronts appear to be flat (planar) and are called plane progressive waves. Plane waves are sound waves in the simplest form.

The opposite of a plane progressive wave is the standing wave. And yet another wave type is the cylindrical wave, due to a series of point sources radiating in-phase with each other that results in a line source.

SIMPLE MULTIPLES

Waves, regardless of whether spherical, cylindrical, or plane, can be considered to be simple or complex. A simple wave is a wave that comprises only one frequency, such as a sinusoid. A complex wave comprises a fundamental sinusoid and harmonics, with the harmonics either being simple integer multiples (2, 3, 4…etc.) of the fundamental sinusoid or non-integer multiples (1.35, 2.21, 3.05 etc.), as occurs for many percussion instruments.

Through a mathematical process called Fourier Analysis, we can decompose a complex wave into the fundamental and harmonic frequencies, their relative amplitudes, and phase relationships. Figure 2 shows a Fourier analysis of simple and complex waves.

Figure 2. Fourier analysis of simple (top) and complex waves (middle and bottom). Figures to the left show wave amplitude as function of time. Figures to the right show wave amplitude as a function of frequency (fundamental and harmonics). Top figure is for a tuning fork; middle figure is for a clarinet; and bottom figure for a trumpet. (click to enlarge)

Two or more simple or complex sound waves can combine with each other through an additive process called the law of superposition. The resulting complex wave, assuming that the waves are linearly related, is the sum of the displacements due to each sound source.

Sound waves are linearly related when each is directly proportional to displacement. Non-linear acoustic behavior typically occurs when the source sound pressure level exceeds 140 dB.

While conceptually simple, the law of superposition is complex since the particle displacement (ξ) and particle velocity (u) of each sound wave may not always be in the same direction because the sound waves can arrive from any arbitrary location. Remember too, that particle displacement and particle velocity are functions of time and frequency.

Thus, the law of superposition requires a vector summation of waves. Waves from opposite directions will add momentarily together at a finite point in space and then pass through each other as the wavefronts continue propagating in their respective directions.

Waves can add constructively, resulting in greater amplitude, or destructively, resulting in reduced amplitude. The law of superposition describes this. What determines the resultant amplitude through constructive or destructive addition is the relative phase of the waves.

Waves that are perfectly in-phase (0-degree phase difference) add together with no destructive behavior. Waves that are perfectly out-of-phase (180-degree phase difference) add together to result in effectively zero amplitude. Most complex waves have phase relationships that vary as a function of frequency and do not combine in such simple relationships as described above.

JUST BEAT IT

One interesting wave addition phenomenon is that of beats. Beats occur when two sinusoids of slightly different frequency, typically less than 15 Hz apart, combine at a point in space.

Because the two waves have slightly different frequencies, they will have varying phase relationships, resulting in times when the waves are partially in-phase and partially out-of-phase with each other.

Thus, the waves will add constructively and destructively resulting in slowly varying amplitude.

For example, if the two frequencies are 220 and 229 Hz, the sinusoids will be in-phase and interfere constructively 9 times per second and be out-of-phase 9 times per second, and interfere destructively. The sound level will vary from loud to soft at a rate of 9 Hz.

Most people can perceive beat frequencies up to about 15 Hz. Beyond this value the sensation of “roughness” occurs with no beating. Further separation of the two sinusoids results in perceiving each as a separate frequency. This is one basis for determining the critical bandwidths of the ear.

Figure 3 shows the generation of a beat frequency.

Figure 3. Generation of beat frequency (bottom) from two sinusoids of slightly different frequencies (top and middle). (click to enlarge)

When a sound wave approaches a boundary surface, a portion of the incident energy is reflected and a portion is absorbed by the surface. The absorbed sound is the sum of the dissipated losses within the boundary medium and the portion transmitted through the boundary.

The characteristic impedance of the boundary surface determines the ratio of absorbed sound to incident sound. The physical density of architectural materials is higher than air and results in most of the sound energy being reflected away from the boundary surface.

Two broad classes of sound reflections can occur: standing waves and specular reflections. Standing waves are the result of the law of superposition. Specular reflections are not based on the law of superposition. The sound absorption mechanism described above is applicable to both standing waves and specular reflections.

Standing waves result from interference of two or more waves that repeatedly pass through each other when traveling back and forth between the room boundaries. The result is a wave that appears stationary having regions of maximum amplitude (antinodes) and minimum amplitude (nodes).

ON THE SURFACE

For rooms, the standing waves are referred to as room modes. Three types of room modes occur: axial, tangential, and oblique. Each room mode type is supported by an increasing number of room surface pairs.

Figure 4. Axial standing waves fundamental mode (top), second mode (middle), and third mode (bottom). (click to enlarge)

Axial modes require two opposite room surfaces (one pair); tangential modes require four room surfaces (two pairs); and oblique modes require six room surfaces (three pairs).

Axial modes are the most audible. The tangential and oblique modes are respectively 6 and 12 dB less than the axial modes. Figure 4 shows axial standing waves (room modes).

A specular reflection occurs when the incident angle from the incoming wavefront at the boundary surface equals to the reflected angle from the boundary. This reflection phenomenon only occurs when the wavelength of the incident sound is less than approximately one-fourth the boundary surface dimension.

For the above conditions, the reflections can be approximated as rays and laws of geometrical optics apply. Figure 5 shows simple specular reflection. The wavelength for low frequency sound is often equal to or larger than the room dimensions. When this occurs, there are no specular reflections, and wave acoustics is used for analysis.

Figure 5. Specular reflection where angle of incidence equals angle of reflection. (click to enlarge)

One key concept to remember when sound is incident at a physical boundary is the particle velocity (v) and acoustic pressure (p) are 90° (π/2 radians) out-of-phase with each other. At a boundary, the particle velocity will be zero and the pressure will be a maximum.

This is important when considering sound absorption of materials: the maximum absorption at the lowest frequency of interest will occur at a distance equal to λ/4 from the boundary. At this distance the particle velocity will be a maximum for the frequency corresponding to λ/4. Since most “acoustical” materials are frictional absorbers, a maximum particle velocity will result in the greatest sound absorption.

Resonance is the reinforcement of sound by synchronous vibration. Every object in nature has a “preferred” or “natural” frequency at which the object will vibrate.  Imposing an oscillatory force of the same frequency as the object’s natural frequency will cause the object to vibrate at maximum amplitude will little energy input from the exciting force. Changing the “forcing” frequency by a small amount will effectively decrease the resonant response.

DO THE MATH

When examining a system at resonance, we will observe a maximum peak at the resonant frequency (fO).

The height of the resonant peak will depend on the degree of damping within the vibrating system.

The resonant frequency response can be either very sharp, centered around a high amplitude narrow frequency bandwidth (Δf), or quite broad with lesser amplitude.

The desired acoustical response will determine which characteristic is best.

Systems that have a sharp resonance characteristic are called “high Q”; those with a broad resonance are called “low Q”.

The Q term refers to quality factor and can be calculated by the following equation:

where,

Q = quality factor, unitless
fO = resonance frequency, Hz
Δf = frequency bandwidth, Hz, taken as the -3 dB down points about the resonant frequency

Figure 6 shows both high Q and low Q resonance response.

Figure 6. High Q resonance (without damping) and low Q resonance (with damping). (click to enlarge)

RULES OF THUMB

Try to remember the following, or key them into your PDA or computer “cheat sheet.”

- Most everyday sounds we encounter are complex waves comprising many frequencies.

- Simple point sources radiate sound as spherical wavefronts assuming the wavelength is smaller than the source dimensions. At greater distances from the source the wavefronts flatten out and become planar.

- Sounds combine due to the law of superposition and the resultant amplitude depends on the amplitude, frequency, and relative phases of each wave.

- Waves will reflect from room surfaces. Specular reflection requires the wavelength to be at least one-fourth the room surface dimension.

- Resonance is the response of a system when driven at its natural frequency. The sharpness of the resonance will depend on the damping within the system. Rooms have special resonant phenomena called room modes.

Neil Thompson Shade has 30 years of experience in consulting and teaching acoustics, noise control and sound system design. He is president and principal consultant of Acoustical Design Collaborative, Ltd., located in Baltimore, and he has also been taught acoustics, sound system design, computer modeling and related topics at the Peabody Institute of Johns Hopkins University.

Related articles by Neil Thompson Shade:
Acoustic Fundamentals And The Nature Of Sound
Getting To The Basis Of Everything We Hear

Tuesday, April 10, 2012

EAW Launches Red Certification Training

EAW has announced the launch of the Red Certification training program in support of the company’s new Powercube power and processing modules.

EAW will award the Red Certification status after completion of a two-day seminar conducted at the customer’s location.

“Powercubes deliver tremendous capabilities to owners of EAW mobile production line array systems,” states EAW president Jeff Rocha. “Red Certification ensures that Powercube owners get the most out of these exciting, new products.”

The Red Certification training seminar was designed by EAW technical training manager Bernie Broderick, a long-time mobile production and concert touring veteran as well as an experienced instructor.

Broderick will deliver the training seminars in North America and will work with the company’s world-wide Application Support Group (ASG) members to deliver the seminars globally.

“EAW’s line array technology is far too advanced for users to just hang a bunch of modules and hope they sound good,” Broderick says. “After completing the Red Certification training, EAW line array owners will fully understand the suite of tools that EAW provides, and they’ll know how these tools work together to deliver consistent results from venue to venue.”

The Red Certification training seminar consists of several sections, including:
•        Point source physics
•        Array design fundamentals
•        Line array physics
•        EAW Resolution array design and modeling software
•        EAW Greybox settings
•        Hands-on array rigging

According to Broderick, the training seminars build toward a complete understanding of EAW’s approach to line array design, control and execution. By carefully integrating modeling software, rigging hardware and digital signal processing, users can build arrays for a range of venues and enjoy consistent results from various designs.

“Once users understand that the ‘basics’ of line array physics are incredibly complicated,” Broderick notes, “they’ll appreciate the crucial role that EAW Resolution plays in designing arrays for a given venue. As soon as they start to rely on Resolution, they’ll understand why Greybox settings are the best way to get the performance they need. If they can build the array with the correct angles between the modules and apply the correct Greybox settings, they should get the results that Resolution predicts. It’s just that simple.”

EAW

The Old Soundman: Nicknames And Advice For An Emancipated Minor

Dear Old Soundman:

Sincerely,
Scohen

Dude!

You absolutely need one! Everybody needs one! How about if I call you “Sco”? That is what aspiring young “yo-cat” fusion musicians at the Berklee School of Music in the 80s called guitarist John Scofield.

These are the same obsessive characters who smugly referred to Charlie Parker as “Bird” and John Coltrane as “Trane,” and sat around discussing “Chick” and “Herbie” as if they actually knew them.

The guy who mixed Primus was known for years as The Shamblin’ Bear. Isn’t that great?! A stage tech for Santana is named Stubby (probably refers to the size of his broccoli, obviously), and their lighting director draws cartoons of characters named Buttface and Tipsy Poodle.

The traveling mixers of the UK rule when it comes to nicknames, like Knobby, Spoon, and the absolute winner, my man Ferret!

C’mon, get on the nickname train!

Next…

Hello Old Soundman -

Hi, pal!

My name is Chris, and I live in Little Rock, Arkansas.

You can dream of becoming president of the United States!

I’m currently 14 - I know, a young whippersnapper, but I’ve been experimenting with bands and sound stuff for a couple years, and people started telling me it really sounded great.

You’re kicking butt! Many guys live until they retire without ever hearing that it sounded really great… maybe because it never did while they were at the controls!

Well, through many good friends in the industry telling me techniques and approaches to different situations, I’ve worked with a good number of bands, probably 75 in the last six months.

Are you an emancipated minor? Do you never go to school? And are there really 75 bands in Little Rock? (You don’t need to answer these questions, I’m just having fun here.)

I just do this because I love music, and I like to make bands sound good, and as a whole I like working with many of the musicians I come in contact with… and just disregard the jerks.

Can you teach me how to do that?

But now I’m interested in $$, not for personal pleasures, but mainly for gear so I can compress, enhance, effect, etc.

What’s wrong with personal pleasures?

Oh… you’re underage. OK—later for those!

I recommend buying the Stereo Aetheric Artifact Enhancer from OSM Audio Industries. It makes the music sweet and low-down, and we offer easy payment plans for any budget!

Whatever you do, don’t buy the Gagger 9000 from Eerie Zombco of Daly City, California—that thing is a rip-off!

I was just wondering how I go about charging for shows? (And when taxes start.)

Thanks,
Chris

Chris, the contrast between your youthful sincerity and nasty, cynical, fault-finding smartasses –- it’s almost too much for me. I need to face away from the camera for a moment, and shed a quiet tear.

Taxes usually start when a law-abiding entity pays you, but I’m not an accountant so do your own homework on that one.

Now, about when and how to charge? Each region has its own market realities. My advice to you is to speak to business persons in your area, such as nightclub owners (not that I would ever advocate you working illegally in such an environment), band managers (a simple wash with a hospital disinfectant such as Betadine after meeting with them should suffice to protect you), and sound company people, and ask them how things work in their worlds.

Keep checking out the Live Audio Board and Study Hall. Learning is good. Working is good. Knowledge is good (thank you, Emil Faber).

Best of luck!

Luv,
The Old Soundman

There’s simply no denying the love from The Old Soundman. Check out more from OSM here.

Monday, April 09, 2012

SurgeX International Joins Growing List Of SynAudCon Sponsors

SynAudCon is pleased to announce that SurgeX International has joined the organization’s sponsorship program.

SurgeX International was created by the original SurgeX founders after North American distribution of SurgeX products was sold to Electronic Systems Protection (ESP) in 2009. SurgeX International now serves the global marketplace with patented 230V AC power protection, conditioning and management products.

The SurgeX and SynAudCon relationship began in 1995 when Michael McCook and Andy Benton founded SurgeX, while simultaneously Pat and Brenda Brown became the proud owners of SynAudCon. The SurgeX group attended SynAudCon’s first seminar and brought with them two ‘new technology’ pieces of electronic equipment. Pat immediately recognized the need for that equipment within the audio industry.

Both companies had an exciting vision, however at that time they were also experiencing many of the same trials and tribulations which naturally go along with a new entrepreneurial focus. This created a natural bond and from that day forward they commended each other’s advancements and accomplishments in the professional AV industry.

SurgeX International’s Michael McCook states: “Education is key for any active individual, integrator or manufacturer in our industry. We know that SynAudCon has always led and continues to lead the way in a fundamental and highly advanced educational role. SurgeX International is proud to support AV professionals by way of SynAudCon sponsorship”.

Through the years, SurgeX has grown to become a leading manufacturer of AC power protection, conditioning and management. Their products are in almost every professional AV system. SurgeX International is now making unprecedented strides in the global marketplace.

“We are extremely pleased that the owners of SurgeX International have decided to sponsor our efforts,” says Brenda Brown, co-owner of SynAudCon. “We have had a rich history with the company. It is an honor to have that same group of people return to support our program.”

SynAudCon

Thursday, April 05, 2012

Science Or Snake Oil? The Facts Behind The Hype About Loudspeaker Wire

Too many good folks have been separated from their hard earned money by hyperbolic claims about loudspeaker wire. There will always be people with more dollars than sense, but they don’t last very long in professional audio.

I speculate there aren’t many (if any) of you who would pay thousands, or even tens of dollars per foot for speaker wire.

A very basic practice in merchandising is called differentiation. Marketers must come up with reasons for why you should buy their wire. To claim that their wire is better, they must first identify, in some cases invent, a difference.

This search for a selling proposition has sometimes focused on “skin effect.” It’s a real effect and describes how at very high frequencies, electrons travel in the outer layer or “skin” of signal conductors.

Another related property is that high frequency signals travel faster than low frequencies through the same cable.

These phenomena are dealt with appropriately in very high frequency applications with several techniques. For example, one particular type of botique wire is made up of a large number of very small conductors braided or woven into one cable, producing a large surface area or “skin” for a given cross sectional area.

Another approach for high power high frequency power transfer is to use a hollow conductor, resembling a section of copper tubing. If the electrons are going to ignore the center of the conductor, why pay for it?

This is not an issue for audio professionals, working at mere audio frequencies of 20 Hz to 20 kHz. Perhaps it would be if we were sending audio over many miles, like the telephone company in its pre-digital days. They had to periodically correct for waveform smear.

But at the speed that electricity travels, our typical path distances are much too short to be an issue.

Out Of Perspective
Wire is not very sexy or easy to create real marketing hooks for, but it can actually make an audible difference. The dominant mechanism is simple resistance.

It’s perhaps ironic that the “snake oil” markers of loudspeaker wire will exaggerate some real but insignificant parameter far out of perspective while compromising the real deal.

Forget the hype, what’s important for loudspeaker wire is that it exhibit low impedance that is resistive in nature. If the wire has a significant impedance component (reactance) that changes over the audio frequency spectrum, this can form a simple divider with the loudspeaker’s resistive impedance and cause a frequency response error.

In addition, since loudspeaker impedance will vary quite a bit over frequency, even a perfectly resistive speaker wire will cause errors. The magnitude of this frequency response error will increase proportionately as the wire’s resistance increases.

Purveyors of “funny wire” don’t bother to make claims about useful metrics like resistance since that is already defined by the wire size or gauge (known as “American Wire Gauge” or AWG for short). That would be like advertising how many quarts were in their gallons!

However, frequency response errors caused by wire resistance are one of the very real things that people actually do hear. I find this following anecdote instructive.

From a discussion with one individual who was certain that he heard a significant improvement when using his “Snake-O Special” loudspeaker wire (name changed because I don’t remember it), I determined that the wire gauge he was using was marginal for the length of his run. The wideband loss of volume caused by a wire’s resistance will be very difficult to hear without a side-by-side comparison.

But the difference in amount of loss caused by the loudspeaker’s changing impedance at different frequencies can easily cause a frequency response error that is probably what he heard. It’s easy to imagine how a rising impedance at high frequency could cause a pleasant sounding treble boost. Just listen to how clean and clear these “Snake-O Specials” sound!

There are several strategies to manage these real losses from wire resistance. The obvious one is to throw more copper at the problem. Heavier gauge wire with lower resistance will exhibit lower losses for a given run length.

Another fairly obvious approach is to locate the amplifiers as close as possible to the loudspeakers to keep the run length as short as possible. A third less obvious approach is to scale up the intermediate signal voltages.

Constant Voltage
There are cases, such as in large distributed sound systems where neither of the first two approaches is cost effective.

You can’t afford to put a separate amplifier at every loudspeaker location, and sending sound sources over long distances with acceptable losses would require very heavy gauge wire.

The solution borrows a strategy from high voltage power distribution systems such as the one used by utilities to bring electrical power to our homes.

The power developed within a given load increases with the square of the terminal voltage (E^2/R). However, wire’s losses only increase linearly with current flow, because the voltage developed across the wire is a simple function of its resistance times that current. 

Power engineers determined that by raising the voltage carried by transmission lines they could increase the power being carried exponentially while simultaneously reducing the losses due to current flow.

The utility company accomplishes this magic with step-up/step-down transformers. By “transforming” a typical 100-amp at 240-volts residential service, up to tens of thousands of volts at the transmission line the 100-amp draw is reduced to the far more manageable level of 1 amp or so. Wire losses are 1 percent of what they would otherwise be.

Similar manipulations go on in “constant voltage” distributed sound systems but rather than stepping up the voltage to thousands of volts the standard for U.S. systems is 70-volt, with Europe using a slightly higher 100-volt standard. The rest of the world tries to conform to one of those two standards.

Of course, the audio signal isn’t actually held constant. The voltage at rated power is. Both 5 watts and 500 watts constant voltage systems deliver the same nominal voltage for distribution.

The goal in any effective distribution system is to deliver as much power as possible to do useful work in the load and waste as little as possible heating up the wire. In a simple distributed sound system sending a few watts of announcements across a few hundred feet of factory floor, the typical low voltage system could drop as much power in the speaker wire as would reach the loudspeakers.

By stepping up to 70 volts and back down again at each loudspeaker the balance of power delivered versus lost is more respectable.

To put numbers to this concept, say we are trying to deliver 1 watt each to two loudspeakers located 100 feet distant from an amplifier using 24 AWG wire. Because we must count wire losses from the feed coming and going, 200 feet total of 24 AWG exhibits resistance of approximately 5 ohms.

Figure 1: Two different ways of realizing one watt at two loudspeakers. Click to enlarge.

To realize 1 watt at each loudspeaker, there would need to be more than 4 watts into the wire at the amplifier end. (Over 2 watts gets wasted as heat in the wire).

 
If we first step up the audio to a nominal 70-volt level the current drops to such a low level that the same wire would only waste 0.14 watts while delivering the same 1 watt each to the two loudspeakers.

As useful as constant (high) voltage systems are for managing wire losses, they don’t make sense for point-to-point runs in sound reinforcement systems. The main drawback is the size of the step-up and step-down transformers required.

To put this in perspective, the size of the transformer has to double every time you drop the frequency an octave. To cleanly pass 20 Hz both step-up and step-down audio transformers would have to be three times the size of a conventional amplifier’s 60 Hz power supply transformer.

Keep It Short
The good news for most live sound applications is that we don’t have to tolerate extremely long wire runs. By locating power amplifiers near the loudspeakers we can keep wire runs reasonably short. At these shorter distances we can easily afford heavier gauge wire.

While power losses are now manageable, it is worthwhile investigating the next dominant consideration in sizing loudspeaker wire.

Frequency response errors will be caused by the voltage divider created between the wire’s fixed resistance and the loudspeakers changing impedance versus frequency.

Figure 2 and Figure 3 show two representative loudspeaker impedance plots, pulled from the Internet.

These are not offered as either worst case or typical.

From the impedance plot in Figure 2, if we ignore the extreme low frequency, this loudspeaker exhibits a maximum impedance greater than 17 ohms, with a significant region of the upper bass down around five ohms.

Figure 2: This loudspeaker exhibits a maximum impedance greater than 17 ohms. Click to enlarge.

Meanwhile, Figure 3, while more complex, covers a similar impedance range, with a maximum around 16 ohms and a minimum around six ohms.

 
To derive a frequency response error we need to compare the drop at maximum impedance to the drop at minimum impedance. The equations below calculate that drop for a given wire resistance.

Note: To simplify this analysis we will assume all loudspeaker impedances to be resistive. While not strictly accurate, loudspeaker impedances will typically be resistive at impedance minimums and any errors caused by load phase angle at the impedance maximums will not be significant for the sake of this analysis.

Minimum Voltage drop= V max = Z max /(Z max +Z wire) 
Maximum Voltage drop= V min = Z min /(Z min + Z wire)

Frequency Response deviation= FR max = -20 Log10 (V min/ V max)

Solving for 1-, 0.5-, and 0.1-ohm wire resistance we get:

Loudspeaker….......1 ohm…...... 0.5 ohm….. 0.1 ohm

Spkr 2 (16/6)....... -.81 dB…..... -.42 dB…... -.09 dB

Figure 3: While more complex than the loudspeaker in Figure 1, this covers a similar impedance range, with a maximum around 16 ohms. Click to enlarge.

Another related consequence is how wire resistance degrades effective damping factor.

While damping factor is usually though of as a power amplifier characteristic, in reality the wire selection can easily dominate actual damping available at the loudspeaker.

In the above examples the 1-ohm wire would by itself cause a rather weak damping factor of 5 or 6 (regardless of the amplifier’s rated damping factor).

Using the 0.1-ohm wire predicts a more respectable 50-60 damping factor, with some small additional degradation due to the amplifier’s output impedance.

Damping factor deserves a more extensive discussion, but for this exercise we will assume that the amplifier’s output impedance is small with respect to our wire’s resistance.

Gauging Gauge
It’s difficult to predict a precise threshold for audibility of frequency response errors.

Controlled listening tests have suggested that differences as small as a few tenths of a dB can be audible.

To satisfy the dual goals of minimizing frequency response errors and not degrading damping factor for the example loudspeakers selected, I am comfortable with targeting a total wire resistance on the order of 0.1 ohm.

Wire’s resistance varies linearly with length. To keep the total resistance below our target limit of 0.1 ohm we must first project the length of our desired wire run, and then select a wire gauge whose resistance per unit length keeps us within the total resistance budget.

Don’t overlook that the wire length is actually twice the run distance as we must consider the feed to and return from the loudspeaker as effectively in series. We must also add in contact resistance for the connections at all ends.

Lets look at how this works out for a practical example of a 20-foot run. First, we double that to 40 feet to establish the true signal path length.

Then we need to account for contact resistance. I’ve seen Neutrik Speakon (or copies of that connector) rated as low as 1mOhm (1/1000th ohm) per contact when new, and guaranteed

< 2 mOhm over life.

Because there are four connections in our total path lets budget .008 ohms for connections. Subtracting this 0.008 ohms from our 0.1-ohm target leaves us .092 ohms for wire. Dividing this 0.092 ohms by the 40-foot length calculates out to 0.0023 ohms per foot.

Plugging this into the equation for wire gauge:

AWG = 10 ×log 10 R +10 (note R is per 1000 feet)

We get:  AWG = 10x log 10 (2.3) +10 = 13.6 gauge

This is a little cumbersome, but once you have established an appropriate gauge for a nominal run length with your specific system. This gauge can be scaled up or down for other run lengths.

Wire resistance changes linearly with length. It changes non-linearly with gauge. A convenient property of wire gauge is that the wire’s resistance will double for every 3-step increase in gauge (AWG). Conversely the resistance will drop in half for a three-step decrease in gauge.

Based on this same example and rounding off to 14 AWG, we can expect similar performance from a 40-foot run using 11 AWG wire, and a 10-foot run would only need 17 AWG. This numbering convention gets a little unusual below “0” AWG.

One step below (larger than) “0” is “00”, and “000” is two steps larger than “0”. I don’t expect to see speaker wires this large, as they would be very difficult to effectively interface with amplifiers and loudspeakers.

Using this example to size wire for your system will get you in the ball park, but it will be more accurate to use actual impedance specifications for your loudspeakers. Manufacturers of professional loudspeakers routinely publish this information.

Remember, use only the impedance max/min deviation within the audio bandwidth of interest. It doesn’t matter what a tweeter’s DC resistance is or a woofer’s 20 kHz impedance, since you won’t be listening to them there.

You also may want to tighten or relax the acceptable frequency response deviation. Better yet, look at your loudspeaker’s typical frequency response and determine if the response errors caused by your wire losses are additive or corrective.

While I don’t suggest trying to dial in corrective equalization using wire losses, if the error is making your system flatter you can afford to be less aggressive in sizing your wire AWG as long as you keep damping and power losses under control.

John Roberts is a long-time professional audio product and system designer and has been writing outstanding technical articles over the past two decades.

Wednesday, April 04, 2012

Microfiles: Altec Lansing (and Western Electric) 639 A/B

The Altec Lansing 639A/B microphone made its debut in the early 1940s, and was originally sold by Western Electric (WECO) under the same model number.

When the U.S. government forced the breakup of WECO in 1947, Altec spun off to continue manufacturing the WECO sound reinforcement and related products, and they continued to make/offer the 639 for many years afterward.

The microphone earned the nickname of “birdcage” because of its size and body design.

It’s an early unidirectional microphone, and what made it unique was use of two microphone elements: ribbon and dynamic.

The ribbon offered a figure-8 pattern; the dynamic, being a pressure microphone, provided an omnidirectional pattern.

By internally blending the two microphone element outputs, the omni cancelled the rear lobe of the figure-8, creating a cardioid pattern.

Both models included a pattern selector switch. The “A” model offered three settings – ribbon, dynamic and cardioid, while the “B” model supplied three additional settings to modify the cardioid characteristic.

Design credit for this 639 goes to William R. Harry and Robert N. Marshall of Bell Laboratories. They received patent number 2,227,580 in early 1941.

Model 639 Condensed Specs
Sensitivity: -84 dB re 1v/dyne/cm2
Power Output Level: -56 dBm at 10 dynes/cm2
Frequency Range: Uniform from 40 Hz -10 kHz
Impedance: 40 ohms (average value, intended for use with equipment having a rated source impedance of 25 ohms to 50 ohms)
Dimensions: 7 inches by 4.4 inches by 3.4 inches
Weight: 3.25 pounds

References: A History of High Quality Studio Microphones. AES Journal, December 1976; Vintage Broadcast Microphones web site; and Dorrough Electronics web site.