Article

Friday, January 27, 2012

Everything You Wanted To Know About Sound Level Meters (SLMs)

The primer: what, how, why, what's available, techniques, applications and more

A sound level meter (SLM) is a device used to make frequency-weighted sound pressure level measurements displayed in dB-SPL.

0.0 dB-SPL is the threshold of hearing, and is equal to 20uPa (microPascals). This correlates to what one would aurally perceive when in a deep cave or in a large anechoic chamber.

Packaged as a single-function handheld test device, SLMs are intended to be held at arm’s length during measurements (to reduce the effects of the body on the measurements) or secured to a tripod stand for more stability.

All SLMs feature an omnidirectional measurement quality condenser microphone, a mic preamp, frequency weighting networks, an RMS detector circuit, averaging circuits, the meter display, AC and DC outputs used to feed other measurement devices or for recording (see Figure 1, below).

Most SLMs have the same set of user adjustments, including SPL range selection, A and C weighting filters, slow and fast detector response, and minimum or maximum SPL.

The SPL range switch provides a balance between minimizing the preamp noise level and measuring a wide range of sound pressure levels.

Most of the commonly available SLMs measure from about 30 to 130 dB-SPL and do this in 3-4 ranges.

 
The more advanced and expensive SLMs feature removable microphones, 1-octave and/or 1/3-octave filter sets, additional weighting filters including B, D and Flat or Linear (no filter), additional detector response options (Impulse and Peak), averaging (over time) and data logging or storage (either on-board, as computer files or both).

Almost all SLMs are designed and specified to perform to one of four internationally standardized levels of accuracy:

 
Note that these tolerances are at 1 kHz, the standard calibration frequency for SPL measurement. In order to ensure the flatness of the SLM there are additional tolerances specified for various frequency bands and microphone classes as well.

Class-0 SLMs are employed primarily to calibrate other SLMs and may be used for very high precision noise measurement in controlled spaces and/or for academic research.

Class-1 and Class-2 SLMs are most widely used by acousticians, sound system professionals, industrial designers/ manufacturers and researchers in academia and government. Measurements made with these levels of accuracy are generally acceptable as evidence in the resolution of legal disputes.

Class-3 SLMs are restricted to noise survey meters and dosimeters.

Microphone Sizes
Most general purpose SLMs are provided with a 1/2-inch free field microphone, permanently attached to the SLM body.

Higher quality SLMs with removable microphones may be outfitted with capsule sizes ranging from 1/8-in to 1-in.

The smaller microphone capsules (1/8-inch and 1/4-inch) have three primary advantages:
—They are capable of higher SPL measurements;
—They remain omnidirectional up to higher frequencies;
—And they present less disturbance of the air due to their reduced physical size (the mic itself has less impact on the measurement).

Larger 1-inch capsules exhibit less self-noise and are therefore best suited to measurement in very quiet spaces, of very low-level noise sources, and of low-frequency energy.

Microphone Classes
One can also choose between three different classes of measurement microphones: Random Incidence, Free Field, and Pressure.

In reality, these three classes of microphone are all fundamentally “pressure response” microphones and are designed to sense pressure - not pressure gradients or particle velocity.

The difference is primarily in how they pick up sound at high frequencies, and this relates to how they are positioned relative to the sound source.

At higher frequencies (where wavelengths become so small as to equal, or be smaller than, the size of the microphone diaphragm) the direction in which the microphone is aimed - relative to the direction of the sound being measured - will have an effect on the accuracy of the measurement.

Random Incidence microphones are intended for use in a diffuse sound field where the sound is arriving from all directions due to a high level of reflections.

This type of microphone must be angled approximately 70 degrees to the sound source in order to yield accurate measurements.

Free Field microphones are designed for use in an open space (free of reflections) and should be held at 0 degrees to (pointed directly at) the sound source.

Pressure microphones are designed specifically for measurements in couplers and closed cavities such as plane wave tubes, ear cavities in dummy heads and for noise measurements in air ducts.

Some SLMs provide a switched correction filter allowing one type of microphone to be used in place of the other.

Note: In general, a “measurement quality” microphone differs from a recording or sound reinforcement microphone in its flat response, the absence of an onboard preamp, and its less rugged construction.

Detector Averaging Time
The standard averaging time choices provided by the majority of SLMs are Fast and Slow.

The Fast setting has a 125milliseconds (mS) averaging time (developed in 1950 to match the ballistics of the moving coil meters used at that time) and is useful for steady state (not dynamic) sound.

For measurement of more dynamic sound (when the Fast setting results in fluctuations of 4 dB or more), the Slow setting provides a 1 second averaging time resulting in much more consistent and stable readings.

It is this setting (along with an A-weighting curve) that is most frequently used for measurement of pop music SPLs.

Some higher priced SLMs also provide Impulse response time (four times faster than the Fast setting) used for extremely dynamic and short term (impulse) sound and peak response time, which accurately responds to signal pulses as short as 50 microseconds.

Frequency Weighting Filters
Because we humans do not hear frequencies in a linear manner (as illustrated in the often referred to Fletcher-Munson curves, aka: equal loudness contours), sound level measurements made with a flat response do not accurately reflect how we perceive sound.

When measuring pop music sound reinforcement systems (with the program signal containing generous amounts of bass energy), broadband readings would be artificially high without the frequency shaping provided by the A-weighting filter.

C-weighting, with its more gradual low-and high shelving filters, is used for very high SPL sound system and noise measurements.

On SLMs without a Flat setting, the C-weighted setting may be used to feed a spectrum analyzer such as Rational Acoustics Smaart for frequency analysis of the broadband sound.

Table 2 below, and then Figure 2 below that, show the frequency response for A and C weighting filters, plus two other weighting filters employed for more exacting and specific noise measurements.

 
All weighting curves are equal in level at 1,000 Hz, the standard reference frequency that SLMs are calibrated to using acoustic calibrators.

Note On SPL Annotation: When communicating or recording (speaking or writing) SPL values, it is necessary to annotate what - if any - weighting curve has been employed. A-weighted SPL is written or stated as “XX.X dBA” and C-weighted measurements as “XX.X dBC”. Such annotation eliminates the need to provide the “SPL” suffix.

The absence of weighting notation implies that a Flat setting (no weighting filter) was used and in this case the suffix “SPL” must be used to differentiate from other, non-acoustic decibel values.

Historical Note: The A, C and B frequency weighting curves were first developed for specific sound levels based on equal-loudness contours.

Because equal-loudness contours were developed from research utilizing pure tones, it later became evident that the weighting curves chosen for SLM measurements at certain levels do not correlate very well to how we actually hear complex sounds.

 
More recently it has been found that the A-weighting curve provides a realistic approximation of how we hear over a fairly wide range of sound level measurements.

Display
With very few exceptions, modern SLM’s are provided with LCD displays with 3- or 4-digit readouts. Most of these provide 1/10-dB resolution except in the case of 3-digit displays when they go over 100 dB-SPL.

Prior to LCD technology, analog meters were employed and these suffered from all the inherent shortcomings of this method of display: slow rise and settle time, susceptibility to damage from pinging (over loading) and overall mechanical fragility.

Another shortcoming of analog meters is the difficulty in establishing stable readings due to the range selecting that is common in most SLMs. One must take the range that is selected and then add or subtract from the meter position accordingly.

Measurements made when the sound pressure level has changed and is out of range are prone to be inaccurate, as well. Some feel that fast impulse noise measurements are easier to read with an analog SLM than with a “digital” SLM.

Both styles of display are difficult to read in dim lighting but some SLM’s feature backlit LCD’s for this condition. And either type of display may be quite difficult to read while at arm’s length if the display size is too small.

Equivalent Continuous Sound Level
Although we live sound folk generally believe that we are able to accurately determine the overall sound pressure level when we view the displayed real time values on our SLM’s, these readings are actually no more than a glimpse, or snapshot, of the varying sound pressure levels that we are exposed to.

Due to the dynamic content of the sounds we normally work with (this applies to both speech and music) and our own human shortcomings, we are unable to mentally integrate the frequently changing SPL’s and determine an accurate mean SPL value.

Research into hearing safety and annoyance and the standards that have been developed from this research are based on the accumulative exposure to sound energy over periods of time.

To this end, integrating SLMs have been developed to provide measurements that are conducted over a selected time period and averaged so that they provide an accurate equivalent energy value, designated as Leq (LAeq or LCeq when weighting is employed).

Integrating SLMs also provide Lmax and Lmin values, and the user can choose the time period that the Leq averaging takes place (normally from several seconds to 24 hours).

More and more organizations and governments are adopting this as a more accurate means for establishing risk of hearing damage, community noise standards, annoyance issues, etc.

Although stand-alone integrating SLMs are considerably more expensive than basic SLM’s, some software is available to provide this statistical averaging from the AC output of any SLM.  and both the TerraSonde Audio Toolbox and Neutrik Minilyzer provide this function along with other SPL measurements and other useful audio system measurements.

Relative Vs Absolute Measurement
Absolute measurements are those made with calibrated SLMs that provide actual sound pressure levels referenced to 20uPa (0 dB-SPL).

Such absolute SPL measurements may be used with and accurately compared to those provided by other calibrated and equal quality SLMs.

Conversely, measurements made with un-calibrated SLMs or those with of low quality (including less then linear frequency response) do not accurately indicate actual sound pressure levels.

SLMs that have been incorrectly user-calibrated or have simply lost their calibration over time also do not provide accurate absolute SPL values.

Relative measurements are simply comparisons between sound pressure levels and do not require absolute calibration nor other “precision” performance characteristics such as a flat frequency response.

Non-calibrated and lower-quality SLMs do provide meaningful comparative measurements as long as the same SLM is employed for these measurements and the measurements are made in the same manner.

Example: A Radio Shack SLM, which is specified as providing +/-2.0 dB-SPL accuracy and which has a less-than linear frequency response, is still able to provide realistic indication of the changes in relative sound pressure levels even though it does not provide accurate measurement of absolute sound pressure levels.

Likewise, the SLM function contained in Smaart may be used “as is” (without calibration) for accurate relative SPL measurement.

One relevant and valid relative SPL measurement that can be made in sound reinforcement work is determining the even-ness of coverage provided by a loudspeaker system.

In this case virtually any SLM is able to indicate any deviations that may exist in the sound pressure levels as the SLM is moved about the coverage area of the loudspeaker system.

In a similar manner we can observe changes in relative SPL as the gain of a sound system is brought up and down and we can accurately determine changes in SPL as a noise source (machinery, for example) is turned on and off.

Available SLMs
As stated previously, SPL measurements made with Class-2 and Class-3 SLM’s provide sufficiently accurate (within +/- 1.0 dB) readings for what we require in sound reinforcement work.

Until recently, SLM devices that met this level of accuracy were prohibitively expensive ($1,200 and up).

Most of the old classic SLM’s are cumbersome to set up and, due to their inherently fragile construction, are seldom used for touring sound or on-site installation work.

Without question the most common SLMs used by live sound folks have been those made by Radio Shack. These appear to represent a very good value, are easy to use and are readily available almost anywhere in North America.

In researching for this article I have found that, along with all of the sound system operators and contractors who own a Radio Shack SLM (or two), there are large numbers of other users including university physics/acoustics and biology departments, municipalities, hi-fi enthusiasts, and theater system installers and property management companies.

And there are at least a few websites devoted to using the Radio Shack SLM for tweaking hi-fi, project studio and sound reinforcement loudspeaker systems and for simple noise measurement.

The original Radio Shack model 33-2050 utilizes an analog meter display and has been in production for the past few decades, and several years ago, the model 33-2055 was introduced, providing a “digital” (alphanumeric) LCD display, recessed and better-protected switches and a maximum/minimum measurement function.

Otherwise, these two models employ the same microphone, base circuitry and battery complement.

 
Alternative SPL Measurement Tools
One alternative to the stand-alone SLM’s described in this article is the SLM function included with some of the better portable RTA’s (real time analyzers) that are often employed for live sound and acoustic measurement.

Models from Ivie, Audio Control, and GoldLine all provide accurate broadband sound level measurement (Class-2 or Class-1, depending on the microphone used) plus octave and 1/3-octave spectrum analysis.

This combination of measurements is very useful for determining both the SPL and the frequency components of the sound/noise under con sideration.

 
Most, if not all, of the current crop of computer-based electroacoustic measurement system provide an SPL function along with their frequency and time-related measurements. Of these, SIM from Meyer Sound and TEF from Goldline are sold with microphones and are calibrated at the factory.

Smaart (any version), SpectraFoo, SysSid, PC-RTA and other similar systems utilize third-party microphones and preamps that must be calibrated by the end-user, using an acoustic calibrator or piston.

Once calibrated, these systems provide very good accuracy and feature a number of variables (weighting filters, etc).

Unfortunately, they are not really “handheld” nor nearly as portable as stand-alone SLMs. And (most importantly) they do not retain their calibration from site to site and therefore require an acoustic calibrator for continued use in measuring absolute SPL’s.

Another product that is a variation of a SLM deserves mention here. SPL “management” systems have been available for perhaps a decade and are directly related to the issues of sound system volume levels.

These simply employ microphones and a software-based SLM system that provides either an obnoxious visual warning or more extreme reactive functions (such as muting of the sound system signal!) when the predetermined SPL limit has been reached or surpassed.

Calibration
In situations where the accuracy of sound level measurements must be recorded and/or verified, it is necessary to calibrate the measurement system both before and after the measurements are made.

Software-based electroacoustic measurement systems that feature an SLM function also need to be calibrated if used for absolute SPL measurements.

An acoustic calibrator provides the standard 94 dB or 114 dB (or both) 1 kHz test tone and should also include the proper size opening or adaptor rings for the microphone(s) you intend to employ.

As is the case with SLMs, calibration devices are manufactured to several levels of tolerance (see table near beginning of article) and they have historically been prohibitively expensive.

Recently there have been several acoustic calibrators introduced that provide very good accuracy for sound system and acoustics measurement.

SLMs may also be calibrated with a pistonphone, but the cost of this device is prohibitively high and the high level of accuracy it provides is not practical for most general sound system related measurements.

Why We Use Them
The simple and snappy answer as to why we use SLMs: so we can know what’s going on.

The need for accurate and repeatable sound level measurement is due simply to the tendency for humans to aurally perceive sound in a very subjective manner. This applies to all aspects of sound perception in the audio field: frequency content, mix “correctness”, system EQ, dynamics, etc.

At some point or another we will disagree on the relative loudness being produced plus these other subjective sound qualities.

Whereas other electroacoustic measurement tools remain very subjective in how we interpret the data they provide, a reasonably accurate SLM provides an irrefutable means to objectively verify how loud the sound is (or is not).

SLM measurements may also be used to verify exactly how much the sound level has been changed (as noted in the section on relative versus absolute measurement).

Anyone involved in live sound system work is aware that there is an ongoing and increasing problem with the issue of volume levels produced in public performance, and specifically, in pop music reinforcement.

It’s wise for anyone involved in this field to become familiar with the actual measured numbers, not to mention the need to continually monitor what we are doing. Issues of personal health and legal liability are an ongoing and increasing concern.

Measurement of unintended noise levels and/or the effectiveness of acoustic treatment are additional exercises that most live sound folks eventually will become involved in.

How We Use Them
For basic every day sound system level measurement, using an SLM is fairly straightforward.

The standard SLM employs a free field microphone and is intended to be held at arm’s length and tipped up slightly towards the sound source.

As you move away from the source and into the reverberant or diffuse field, or when measuring in a multiple source sound field such as on stage, the angle of incidence of the microphone becomes much less of a concern.

When measuring noise levels at installation sites, some care should be taken. If measuring the noise level of a rack room, for example, be sure that the SLM is positioned away from walls, floors and other large boundaries.

Signal-to-noise levels are seldom an issue for loudspeaker system measurement because we can simply turn the system up, more than far enough above the noise floor.

But when measuring the noise from a loudspeaker system (hum, hiss and/or the fan noise that may exist in self-powered systems), or from a mixing console and its power supply, etc., or from other common noises such as HVAC systems, vacuum cleaners, power tools, etc., the unintended noise must be at least 10 dB-SPL lower than the noise being measured.

When measuring and logging levels to help resolve a dispute, both A and C weighted measurements should be provided, as should the distance from the source and any other detailed notes.

Also don’t forget that a non-annotated measurement (“00.0 dB-SPL”) will be interpreted to be flat (unweighted).

General Measurement Guidelines
When conducting sound pressure level measurements there are a few basic rules that apply:

• Keep the SLM at arm’s length, aimed in the direction of the sound source(s) and positioned away from room boundaries. By the way, the mixing surface on a large (40-plus channel) mixing console will effect SPL measurement when the SLM is laid down on the mixing surface or angled against the meter bridge.

• For almost all sound system measurements, use the A-weighting filter and Slow response setting.

• Be sure that you’re measuring within the range that you have selected. SLMs with multiple range positions and those with analog displays, in particular, may provide incorrect readings when the SPL is over/under the selected range.

• Prior to making SPL measurements, ensure that the SLM has been turned on for a few minutes and has reached room temperature.

• Wind and air-blowers will effect SPL measurements.

• Be aware of where you are taking your measurements. For example: if you measure from within the area where the bass energy from the sound system has combined (aka “power alley”) or within a standing wave null, the readings will be non-representative of the SPLs for the majority of the audience.

Tom Young is principle consultant for Electroacoustic Design Services in Connecticut, and he has designed hundreds of systems for churches and similar spaces. Tom’s also the moderator of the Church Sound Community Forum here on ProSoundWeb.

More articles by Tom Young on PSW:
Time Is On Your Side (If You Want It To Be) With Sound System Alignment
Microphone Techniques For Drums In Contemporary Worship Environments
The Aux-Fed Subwoofer Technique Explained

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Tuesday, January 24, 2012

L-Acoustics U.S. Sets Training Dates for KARA, KUDO & SOUNDVISION

L-Acoustics U.S. has announced its first two product training sessions for 2012.

The first three-day training is set for February 20 to 22 in Red Hook, NY and will specifically focus on the new KARA modular line source system and SOUNDVISION version 1.9.

The second session, hosted in Oxnard, CA exactly one month later from March 20 to 22, will cover the large-format KUDO line source system and SOUNDVISION.

“We’re particularly looking forward to our KARA and SOUNDVISION session in Red Hook as it marks our first official East Coast training,” says L-Acoustics head of U.S. touring support Scott Sugden. “We’ve had a lot of interest in a regional event like this from our eastern customer base and we’re very happy to now make it a reality for them.”

Primarily designed for technicians, mix engineers and sound designers referred by L-Acoustics Rental Network agents and clients, the first two days of each training will offer a blend of theoretical knowledge and field procedures focusing on operating and optimizing either KARA or KUDO in a safe and controlled environment.

A third day, which can be attended separately or in conjunction with the KARA/KUDO training, will be dedicated to covering the manufacturer’s SOUNDVISION 3D acoustical modeling software.

Upon completion of these seminars, attendees will receive a certificate of attendance.

The number of participants for both the Red Hook and Oxnard training sessions is limited to 12 people and priority will be given to L-Acoustics Rental Network agents and system owners.

For additional details on the training seminars and their related costs, click on the Support tab at www.l-acoustics.com or contact .(JavaScript must be enabled to view this email address).

L-Acoustics

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Thursday, January 05, 2012

A Conversation With Audio Pioneers, SynAudCon Founders Don & Carolyn Davis

The life, times and contributions of two individuals who dedicated their lives to improving sound quality through education

When noting the contributions of Don and Carolyn Davis to the professional audio industry, it’s hard to know where to even start. Their book, Sound System Engineering, originally published in 1973 (and since updated), remains a standard audio and systems resource.

Founders of SynAudCon, Don and Carolyn established the industry’s pre-imminent and most respected (and independent) educational resource, teaching thousands the essential concepts of audio and acoustics that in turn has led to remarkable advancements in systems and sound quality that we all enjoy. Now consider that these accomplishments just scratch the surface of their crucial role in leading the industry to its current modern era…

I had the privilege of spending an afternoon with Don and Carolyn while attending a SynAudCon seminar and workshop in southern Indiana. They were gracious enough to travel to meet me, with the warm and at times reverential reception they received from attendees standing as a testament to the tremendous respect they’ve tirelessly earned in service. Our conversation was fascinating, spanning a wide range of topics and touching on crucial historical landmarks that lend perspective and understanding to the current state of the industry.

Now “retired,” they continue to travel extensively, staying in touch with an ever-growing network of friends and exploring new places. Like many long-married couples, they have the endearing trait of often finishing each other’s sentences or interrupting to take the conversation in new directions. Frankly, I didn’t have to interject much as the two shared the fascinating tale of their lives in pro audio. So without further adieu, let’s roll tape and simply say, “go”.

Keith Clark: Don, I understand you worked with Altec Lansing prior to the founding of SynAudCon.

Don: I worked with Altec from 1959 through the early ‘70s, marketing and, really, managing mostly. I was a field rep based in Chicago serving a big chunk of the Midwestern U.S. We weren’t exactly sales reps, but more comprehensive in scope. Prior to this point, Altec Lansing products were distributed through Graybar, and major installations were often headed up by the Altec Service Company, the theater service division.

Just at the time I joined the company, they decided to set up their own distribution with sound contractors. A guy named Mo Morris had seen the vision that sound contracting was a viable thing, that it was a good way to move inventory out of the factory and into the warehouses of the contractors, and that it was a good way to respond quicker to needs.

So my job was to go out and identify potential contractors, and then to set them up as dealers and make sure they were supported, providing any encouragement possible.

This led to doing a little bit of everything. I enjoyed this role a great deal, and in the process, I worked with some of the “old-time” guys who had been Western Electric contractors. They were superbly trained people and quite used to top-of-the-line equipment – a piece of Western Electric equipment cost more than anything else, yet they invariably got all the better jobs.

Nate Reese in Detroit is a good example of this. It was said that during his first couple of years in business, he lost almost every job he bid on. But then he followed up with these same customers a bit later, knowing that most would be unhappy. He’d say ‘hello, I’m Nate Reese and I was high bidder on your project. Are you happy with the work?’ And, of course he got most of them on board as permanent customers.

After a while he didn’t really have to be too involved with the bidding process, because if they wanted it done right, they came to him. Nate was probably the first guy to make himself a millionaire in audio. It was his integrity, and that of Western’s gear, that did it.

So in the background was Western Electric, and you went out and tried to find people that fit that mold. When Altec was formed after the dissolution of Western Electric in the late 1930s, a lot of the Western personnel came on board. They bought up the rights to the best Western products for pennies on the dollar and then proceeded to make themselves wealthy men.

KC: You were one of the pioneers of equalization…

Don: At Altec, I constructed a seminar program in 1968 to show people how to equalize systems. The initial problem was that while even the early equalizers worked very well, the systems in general didn’t. People put in EQs and discovered they hadn’t planned enough power, for example, because now they could raise the levels. And what had been adequate before in feedback constraints wasn’t even close to adequate any more. A 10 dB increase in acoustic gain meant a 10 dB increase in power.

This emphasis in training people for equalization is exactly what Pat (Brown) is doing here with SynAudCon. You’ve got to look at polarity, you’ve got to signal align, you’ve got to clean up all of the impedances, match all the levels, and so on.

The way I found out about the problems, initially, was that we had franchised a bunch of contractors to handle equalization, and they had to spend about $10,000 on specialized equipment – GenRad and Hewlett Packard test gear. But nothing good in the way of progress and improvement seemed to be happening, so Carolyn and I loaded the first HP Real-Time Analyzer (RTA) ever made into the trunk of our car -

Carolyn: - Don had talked HP into building the RTA for him, the first one ever -

Don: - and we went on the road to find out what was going on. We quickly saw that even the best contractors were building inadequate systems – not that they weren’t great compared to most others, but they still weren’t adequate in terms of the extra power that could and should be delivered.

We learned to look at a space and to understand that what it presented acoustically was the challenge. Fit and match the space with an array that could meet the criteria of the space, and then work backwards through the system to fill it out with power and other components needed to do the job right. At that point, system design was being done just the opposite, from the microphone out, rather than speakers back.

Carolyn: And you should mention at that time that HP had also just introduced the desktop computer –

Don: – and that was a huge help.

Carolyn: Yes, Don bought into it quickly.

Don: I was looking at all the “gimmicks” of the time. But in this case, specifically, I was always lousy with a slide rule anyway, and the ability to be able to program everything on this portable computer was great. These early computers were really nothing more than a big programmable calculator, but they were very helpful.

In the earliest computer, we had to go through about 2,000 steps to attain calculations. Reverberation time, noise control, acoustic gain – all of this and more was plugged in for calculation. Of course, we hadn’t discovered how to do intelligibility yet, this was still intuitive only.

A bit later, V.M.A. Peutz of Holland and some other smart people figured out that intelligibility could be designed into a system ahead of time. Peutz was a real genius, unlocking the whole intelligibility problem. While there are current “gods” of intelligibility, this is where it all came from, where it all started.

When Peutz took one of the early TEF analyzers and programmed it to measure intelligibility, essentially - everybody objects to the term “measurement” in this regard but its an estimate taken off the data, it provided a place and explanation as to why so many systems of the time were falling short. The numbers really proved it.

KC: I’ve also read that you were instrumental in bringing the first TEF analyzer to market.

Don: Cal Tech (university) came to us and asked if we’d take over the licensing of Time Delay Spectrometry. They had only one licensee at that point, after a decade, and we got them 120 or so licensees within a year. That was kind of an interesting experience, and when they said, ‘OK, now it’s going good and we want it back’, we gave it right back to them. We weren’t in the business to be wheeler-dealers.

Carolyn: Getting back to equalizers, in March of 1968, Don went to a convention and came back with this idea for equalizers. He went straight to Art Davis (an Altec engineer) and told him about it. Art wanted to do it a little differently, and Don said fine, I don’t really care, and he and Don were on the original patent.

Don: I spaced out what the filters had to do, and Art made a contribution I hadn’t thought about, to make frequencies combining, summing -

Carolyn: – and we had a prototype by September and went to the AES Convention that year and presented a paper on it.

Don: The chairman of the session had been involved in early equalization work as well, and when he read the title of the paper - “One-Third Octave Broadband Equalizer” - he kind of stopped and raised an eyebrow on the word “broadband”.

To him, what we were calling “broadband” was actually very narrow. Now, there’s nothing wrong with a filter being exactly the shape of whatever your problem is, but you can’t go after anything that isn’t the middle of the phase realm. There are things in there - “bumps” - that if you put an EQ on it, you only make the problem worse.

But if you put it in the minimum phase realm, then the EQ clears everything – it corrects amplitude, it corrects phase, it even corrects time. But it must precisely meet, and any divergence causes problems. There was a great deal to be said for a parametric equalizer, only nobody really knew how to make them at the time. Dr. Paul Boner was making these real narrow filter devices, trying to make the intrusion as minimal as possible. But one-third octave shaping filters could shape to the broadband nature of the problem beautifully, and they didn’t introduce any major phase anomalies as a result. You follow the general shape of the curve.

Now there might have been a little individual narrow-band anomaly, but these were so narrow that they were inside critical bandwidths, and thus they didn’t much matter.

Nowadays we have the correct parametric process and equipment, and there are also these beautiful programs that invoke the house curve and let you match to it. If you know what you’re doing you can get very refined equalization.

But in the meantime, one-third equalization dramatically improved loudspeakers of that time, and it also led to discovery of problems with signal alignment. This is still something no one has really pursued fully yet, at least that I’m aware of. I don’t think the equalization field and issues have been fully worked out yet.

Right now, with most of the current devices, you get further by improving the audible quality of sound systems with signal alignment than you ever do with anything else, particularly with the newer array concepts. It will always be a tough job to have more than one of anything in an acoustical system – nature doesn’t like that. So, you make your compromises.

The contribution that I felt like I made is that prior to this work, the acoustic environment was almost totally ignored. Yet all along that was the major tool to play with. And in fact, most rooms ought to be corrected by people doing sound systems. There’s an optimum match for every system to every room, so that you don’t add any more power than needed for maximum intelligibility and you don’t add any more absorption than necessary for maximum control of energy. This is what a good acoustical consultant should do, but it’s surprising how many of them don’t.

KC: What are the roots of SynAudCon?

Carolyn: By 1972, we could see that things at Altec were not going so well due to some management problems. About that time, Don was asked to establish the European market for them, and he said we’d go over and check things out before agreeing to do it. But at that time, the economy was under some dramatic changes and it just wasn’t feasible -

Don: - well, we had an acquaintance named Mr. Vorwig who had been in charge of truck production during the war (World War II), on the German side, and who also had been the engineer that originally tested the Volkswagen for Hitler. Mr. Vorwig had a party that we attended, and he and some of the guests, including a banker in Frankfort, laid out for us what exactly was going to happen with the economy, the deflation of the U.S. dollar that would occur. I had to tell Altec that I wouldn’t take their offer.

Carolyn: Don and I used to work for a few years and then take time off and go to Europe and travel for months at a time – we didn’t have children so we could do that. Through the ‘50s, the economy was great, but by ’72, we found that prices were already 10 times more than in the ‘50s. And, things had changed with Altec –

Don: - when a company is being torn apart by bad management, the talent leaves first. The ones that hang in there may be great workers, but that’s not where the talent lies and where the future and insight is. There were a lot of strange contracts coming across my desk that I didn’t want to sign, and this is what happens… I’ve often sworn I was going to write a book on mismanagement with all of it I’ve seen over the years. I resigned from Altec in December 1972.

Carolyn: Altec offered Don a year’s salary if he would not go to work for the competition -

Don: – Which I had no intention of doing anyway –

Carolyn: – we took six months to write our book, Sound System Engineering, because we had an income from Altec. Sams Publishing printed it at no cost and allowed us to buy it at $10 a copy. It was loose leaf at that time, and about three years later they decided to publish it as a book. Then a few years later, we revised it.

Don: We had a lot of lovely people help us with this, just like Pat (Brown) does now with SynAudCon.

Carolyn: GenRad and HP loaned us thousands of dollars worth of equipment for our seminars.

Carolyn: In 1973, the oil crisis started and things were not good in terms of starting a business, but we decided to anyway.

Don: We set out on the road with a Dodge three-quarter ton truck and a camper shell to house all the gear, towing a trailer behind it to live in. We toured the country and taught audio.

Carolyn: Don could see that the only way we would really be able to make it in doing this tour would be to set up a sponsorship program. He went to Shure - or they came to him, I can’t recall – and they were great in terms of support. That first year, Shure, UREI and Sun Music were our first sponsors.

Don: The point is that there were several of these engineering folks and their companies who were very supportive, who understood what we had and wanted to give.

Carolyn: Another interesting and critical thing at this point in time is that Altec pretty much owned the contracting business. RCA had a service company and could still do some things at that point. And, some other names that aren’t even around anymore were the big entities. At the time, companies like Electro-Voice, Shure, JBL and so forth were really still just independent gadget makers.

What we did that was unique at the time was to put together all of the elements offered by these companies into proper systems. These pioneer sponsors of SynAudCon could provide the quality components, individually, and then that equipment could be formed into quality systems.

Don: UREI, for example, was one of the first to make the equalizer, and they were a sponsor. Emilar would make the drivers that were needed. So we “filled the chain” with sponsors so that people would know where to go to fill out an entire system. That was a piece of serendipity that worked out well for both us and the sponsors. It wasn’t really a deliberate thought-out thing, but just something that happened.

Carolyn: The next year after we started the sponsorship program, Don wanted something to bind SynAudCon “grads” together, so he started a newsletter subscription, free for one year to everyone who attended a seminar, and then renew for $25, later raised to $35.

KC: I understand you had settled in California by this time?

Don: Well, we owned property there, up in the mountains. It was a place to park the trailer and basically camp out. We’d be on the road for nine months out of the year and then go back and spend part of the winter in California.

Then in the summer, when everyone was busy putting in school systems, we’d park the trailer out at the (family) farm in Indiana. The old house hadn’t been rejuvenated at that point. We were living a gypsy life.

KC: So how long did you operate SynAudCon as a “road show” concept?

Carolyn: Well, in 1992 we were still doing classes in the U.S., Canada, Europe, Japan and Australia. We were in Japan on one of these trips when Don woke up one morning and said ‘this isn’t the way I want to spend the rest of my life’. So we canceled everything at that point. Travel had gotten old.

We had moved to the farm in 1987, so we decided to take the “old farmhouse” - built in 1883 - and fix it up so we could hold classes there for 10-12 people at a time. This allowed us to keep teaching, because we still loved that part of it. We did this through 1995.

A good consultant and/or contractor – someone who worked daily in the industry - would present the hands-on, and Don would teach the theory. Now Pat can do both the theory and the practical. Don has more of an interest in the theory, never quite as interested in the hands-on side of things.

KC: So outside of your absolute dedication, why do you think SynAudCon thrived?

Don: The fascinating thing is that in the 25 years we ran SynAudCon, we hardly had a conflict with any sponsor about anything, and almost all of them are still with Syn-Aud-Con to this day.

We always tried to have a sense of integrity about our relationships with sponsors, and this was reciprocal. One time we did have to “fire” one prominent loudspeaker company as a sponsor, because they were unfairly attacking another party and presenting grossly incorrect information. This just couldn’t stand, and we refunded their money. So we always did our best to have a sense of integrity about what we were teaching.

Carolyn: We limited sponsorships to 20 and had a waiting list, and Pat has expanded that.

Don: The point is that you’re out there trying to teach people about what’s right and wrong from a technical standpoint and they’re being told so many other things making it that much harder. We’ve had people accuse us of being prejudiced, and that’s not the case.

Carolyn: We’ve always had a special appreciation of new ideas and talent, and have so much enjoyed the promotion of that talent. So much of the ‘70s was an accumulation of a lot of information, and then in the ‘80s, all of this began to be focused into new ideas and products.

Don: We got to the stage when we could recognize talent when it wasn’t perhaps all that obvious to others -

Carolyn: - Richard C. Heyser, Peter D. Antonio, V.M.A. Peutz, Dr. Eugene Patronis, Gerald Stanley, Ed Long, Ron Wickersham, Ken Wahrenbrock – these were the people that developed the concepts that were so important to us: TEF, QRD Diffusors, %Alcons, LEDE control rooms, PZM, signal alignment, etc. They conceived the ideas. We often brought their concepts to the attention of manufacturers. I was mentioning this idea recently to a friend, and he said that the ‘80s was an outpouring of everything we had learned. But this was more on an individual basis, and now Pat and Brenda are taking the entire industry upward in the same way.

KC: How did Pat and Brenda come to take the reins?

Carolyn: Each seminar that was scheduled at the farm had a consultant or contractor to work with us. A consultant scheduled to work with us in a seminar had to cancel at the last minute due to health reasons, and we asked Pat to come in and teach on an emergency basis – Pat lives only an hour from the farm. And he was great, pretty much teaching just as he does now, explaining things so clearly and so well and feeling very comfortable in front of a group of peers.

Don: He loves it -

Carolyn: - and a little later, Janine Masten, who was with EV at the time –

Don: - sharp lady –

Carolyn: - this was in 1995, and she called to ask if Don would break his rule and go to Europe to teach for them. I was sure Don would say no, but instead he turned around said, “I’ll do it if Pat Brown comes with us.”

Don: And they said yes -

Carolyn: - and when they finished the classes, Larry Frandsen (head of Mark IV Audio Europe at the time) invited us to come back the next year. Don declined, but as he did so, Larry immediately turned to Pat. Pat accepted – which was what Don had in mind when he asked for Pat to be included in the tour.

That trip lasted about three weeks, and during that entire period I didn’t call to check in with the office, and it was such a relief. It felt like it was time for us to move on, and we asked Pat if he would take over. He talked it over with Brenda, who was a very successful nurse at the time, and she gave her support. Gradually, she worked into the business more and more and now has taken on a full partner role with Pat.

Don: Well, Brenda’s very sharp, very on top of things, understands the technical part in addition to her business talents. They also have a very spiritual side to them, that we love -

Carolyn: - they don’t talk about it much. They’re so ethical and we take a lot of pride in that.

KC: What’s the biggest difference in SynAudCon now, in comparison to what you handed off to Pat and Brenda?

Don: Pat has computerized the teaching process, has brought it the rest of the way into the digital age.

Carolyn: At a seminar or workshop, everything you see Pat doing with the computers and video screens, Don used to do with slides and overhead projectors.

Don: We have preached digital revolution for 20 years, that it would be the way to go, the way of the future. It’s interesting to look at the space race – a lot of people think all of their money was just shot to the moon, but actually a very small amount of hardware went there. The big thing to come out of it is the ideas, the outflow of technical creativity.

Don: Pat and Brenda have done another vital thing, and that is to go places where we had never gone. Mexico, South America, Jordan, India, Dubai -

Carolyn: - and he’s invited to China -

Don: - and that’s invaluable. He’s spreading the knowledge. In the late ‘50s, Carolyn and I worked at the American National Exhibition in Russia, an exchange fair between them and the U.S. We were showing audio equipment.

Recently, I was interested to read a book written by a former top KGB agent who noted the most subversive thing that ever happened between the U.S. and Soviet Union was this exchange, that it changed more things in Soviet Russia than anything else. He was kind of tongue in cheek about the subversive part, but what he was saying is true.

As the SynAudCon attitude gets around, the idea that you share the information rather than hold it close, as that philosophy gets into new places, it’s fascinating to see what comes about. SynAudCon became a society, a family really, without meaning to, based on this idea of sharing information.

Carolyn: Along these lines, the web site and what Pat does with the list serve is unbelievable, and the newsletter keeps going strong. This all goes with being a society. It’s just amazing that top professionals in this industry will gladly tell everything they know through these channels, unselfishly and for the benefit of anyone willing to learn.

Recommended Reading:
Sound System Engineering
If Bad Sound Were Fatal, Audio Would be the Leading Cause of Death

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Friday, November 11, 2011

Call For Entries In NSCA 2012 Excellence in Business Awards

Submissions will be accepted through January 15, 2012, for self- and peer-nominated applications

Systems integrators with outstanding success from an effective business strategy are encouraged to apply for NSCA’s 2012 Excellence in Business Awards. Submissions will be accepted through January 15, 2012, for self- and peer-nominated applications.

Winners will receive one free admission (a $1,049 value) as well as recognition throughout the year in various NSCA and industry publications.

Staying ahead of the competition is one thing, but companies with solid business sense and creative tactics continually beat their competitors to the finish line.

Successful strategies in fiscal responsibility, marketing, training and strategic advancement cultivate increased numbers of customers and, consequently, increased profits.

Similarly, businesses that are smart about retaining their staff through unique training opportunities or an organizational philanthropic philosophy not only draw the best talent, but also keep their companies ahead of the competition. 

Doug Hall, CEO at The Whitlock Group, said he was proud of his organization’s fiscal policies that earned an Excellence in Business Award for fiscal responsibility in 2011. 

“We feel strongly that making sound decisions for our company and our clients helps to keep us all competitive,” Hall said. “This value-based approach, which includes ROI assessments and long-term support for A/V and VTC technology investments, is what distinguishes us from many of our competitors in the industry.”

Customers, business partners, manufacturers and peers are encouraged to nominate systems integrators who have excelled in one of the following categories:
• Growth Strategies
• Professional Development
• Strategic Advancement
• Project Development
• Marketing Strategies
• Education of Allied Professionals
• Recurring Revenue
• Fiscal Responsibility
• Philanthropic Contributions

Candidates may apply online at www.nsca.org/blcawards.

The NSCA Business & Leadership Conference Committee will announce the winners on February 1, 2012. Conference attendees can register now to save $100 on admission; rates increase on January 5, 2012.

The Excellence in Business Awards will be presented at the 2012 NSCA Business & Leadership Conference, March 1-3, in Dallas. Winners will be featured during the Opening Night Reception, when they will discuss their strategies and techniques in an open forum with their peers and key industry representatives. For more information or to register, visit www.nsca.org/blc

NSCA

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Thursday, November 10, 2011

Microphone Techniques For Drums In Contemporary Worship Environments

There are a number of misinformed concepts about drum sound in pop sound reinforcement and these are equally wrong when applied to sound reinforcement in contemporary worship music

Mic’ing drums in a contemporary worship environment is a learned craft and skill that relies on technical know-how, a musical ear, partnership with musicians (and the drummer in particular), favorable acoustics, well-behaved loudspeakers, reasonable financial investment, and lots of patience.

How’s that for mix of encouraging and disheartening words?

The need for sound reinforcement in contemporary worship and gospel music is similar to that in modern pop music (including R&R, R&B, soul, funk, electric folk, etc.), especially where musical content and electroacoustic needs are concerned.

The drum kit’s position in the music and the mic techniques employed are the same as for pop music and, unfortunately, the rather long list of potential problems that can plague sound folk and drummers alike are a good part of this equation.

There are a number of misinformed concepts about drum sound in pop sound reinforcement and these are equally wrong when applied to sound reinforcement in contemporary worship music. Here’s the shortlist of misinformed ideas:

1. In smaller spaces drums do not need to be mic’d - everyone can hear them just fine.
2. For “pure” drum sound you should use a minimalist mic technique consisting perhaps of an overhead (or two for stereo) and a supplemental kick drum mic.
3. Drummers rely on feel and this equates to striking at whatever force they need to
4. If drums are too loud, we can just throw up some absorbing materials and this will really improve things.
5. If drums are too loud we can just buy inexpensive plexiglass shields and this will minimize the volume levels.
6. If the drums are too loud, we can reduce the volume levels with properly designed acoustic gobo’s and then add monitors so the drummer can hear his drums at the volume he/she wants.
7. If the drums are too loud the answer is to buy an electronic set.

Where We Came From
Somewhere during the 1950s-1970s, during rock-pop music’s formative years, a number of practices were developed that shaped how we perceive drums in the music we are discussing.

Virtually all of these mic’ing techniques and their resulting trademark sound were derived in recording studios. Many of them were developed for very good reasons relating to recording technique and some were sheer accidents that someone was savvy enough to recognize as worthwhile and didn’t record over.

The basic aural attributes of the majority of drum sounds that we employ and love are clarity, distinctness, power, and controlled tone with pronounced attack.

In some cases, there have been intentional efforts to minimize one or more of these. (I think back to some of the seminal recordings from the 1970’s-80’s, wherein the drums sound like very well-recorded cardboard boxes – with no tone whatsoever.) But overall these are what we recognize as key ingredients to great drums sounds.

Another fact of life in drum recording and reinforcement is that seldom does the sound that we strive for have much to do with the acoustic sound of the kit. Let me put this another way: Great sounding drums in live sound are completely unreal. They are not natural. Period. What we hear at the kit is not what we want to hear out of the FOH system and it is not what the drummer wants to hear from his monitor, typically.

Now don’t get in a tizzy if you are thinking of jazz drums, ethno-folk, new-age drums or any other musical style in which it is desirable to faithfully “capture” the real sound and reinforce it. That’s not what we’re talking about here.

So much of pop music (think post early Elvis and Little Richard) is distortion, not just in the case of cranked-up guitar but also grossly exaggerated and manipulated vocals, keyboards, saxophones, organ, ambient effects, etc.

Why We Continue In This Fashion
Aside from the musical merits of providing the same “in your face” drum sounds that virtually all popular rockinfluenced music employs to very good effect, there are very good live sound-related reasons to pursue the techniques that we do.

No matter what level you and your church may be on, from the lowest of hodge-podge, hand-me-down, and donated sub-garage band sound systems to the most well-endowed megachurch with unlimited equipment and designer budgets plus technical crews that fell off the last Sting tour, we are all fighting the same live sound battles.

In order to retain good, clear microphone pickup we need to reject the ever-present leakage that permeates every stage and platform under the sun.

To this end we must employ close mic’ing and utilize large amounts of playing and mixing chops to ensure that each drum hit is captured with a minimal amount of noise. ‘Noise’ is employed here just as it is in other instances of signal-to-noise that we in audio always have to deal with.

Each drum that we have so meticulously mic’d and is so wonderfully wacked by the incredible drummer is also a noise source for any and all of the other mics typically employed on a drum kit. This is where musical technique along with audio technique come into play and are of equal value.

The basic fact of life here is that in the process of close mic’ing each drum so that we have in-your-face drums sounds, we create the potential for an electroacoustic time-smearing quagmire of multiple sources that get mixed together and result in indistinct and mushy drums sounds. How is this so?

Simple. Take that snare drum with the very closely placed Shure SM57 that is aimed down toward the drummers gut. Now look over there at the high rack tom with its equally close-placed mic that is pointed down toward that drum.

What happens when the snare is wacked? First, the snare mic captures that snare sound and it is fed into the mixer. But that same snare wack is also picked up by the high tom mic which is 30 inches away.

The SPL (sound pressure level) of the snare at the rack tom mic position is perhaps 1 dB lower (at best) than it is at the snare mic. This tom mic is also fed into the mixer and when the two microphone signals are combined into a mix bus, there is a significant series of cancellations (comb filtering) due to the timing disparity of the two signals.

But that’s just the tip of the iceberg.

We also have a low rack tom mic (36 inches away from the snare mic), a floor tom mic (42 inches away) and one or two overhead mics (about 36-60 inches away), plus the hi hat mic (which is perhaps 15inches away) that, despite its cardioid pickup pattern and being shadowed from the snare mic, still picks up plenty of snare wack.

Incidentally, this is one reason why sheer plexiglass shields are likely to do as much as harm as they might do good. Take the above scenario and add a huge number of reflections that each drum wack creates as it impacts the drumshield and bounces directly back into the drum mics.

What are we doing here?

Well for one thing, what we might try to do is go the minimalist route. Obviously, if we use only two overhead drum mics then all of this multi-sourced timing garbage is minimized to a great extent.

But there also goes the “in your face” drum sound we need and we are also subject to leakage from every other sound source on the platform, partly because these overhead mics will need to be run hotter and more “full-range” than they would be if they were there just for the cymbals.

Just to be clear, for acoustic jazz/folk/ethno stuff this may very well be the preferred method, but not for contemporary worship music.]

Another very good reason why we need to close mic and provide in-your-face drum sounds is that the music we provide is very complex and dense. In the process of keeping repetitive snare and kick drum hits distinct in the sea of competing sounds (including bass, keys, guitars, solo voices, ensemble voices, choral voices, horns, ambience, etc), we rely very heavily on sheer clarity and punctuation of the percussive drums sounds.

This also applies, by the way, to other percussive sound sources such as bass (walking or slapped) and piano. Clarity is achieved by close mic’ing and tuning the drum kit in the player’s style or skill.

Speaking of drum tuning, this is a very large component in how successful we can be in achieving musically pleasing drum reinforcement. It’s the old “garbage in, garbage out” scenario that permeates what we do in audio.

Yes, the quality of the drum kit is very important, but the best, most expensive drum kit can sound quite horrible if the player does not know how to tune it for the musical style and for the environment in which the kit is being played.

Starting With A Good Sound System
Offering advice on how to mic drums for sound reinforcement without first clarifying the sound system requirements is foolish, in my opinion. Nothing we might do to try to improve drum sounds will be as effective as first starting with a carefully designed loudspeaker system that is optimized for maximum performance.

Timing errors that occur between multiple drivers and that are not corrected or minimized are the key component in lousy phase response in a speaker system. Poor phase response equates to negatively impacting the sound put through a loudspeaker system and in particular percussive sounds.

Regardless of what style of music you’re dealing with and the methods you use for mic’ing (close, distant, area, etc), without a well-behaved speaker system you are limited in the clarity you can achieve.

Tom Young is principle consultant for Electroacoustic Design Services in Connecticut, and he has designed hundreds of systems for churches and similar spaces. Tom’s also the moderator of the Church Sound Community Forum here on ProSoundWeb.

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Friday, October 28, 2011

Download The Latest PSW Webinar Featuring RF Consultant James Stoffo On Wireless Systems

Covers the latest on the "white spaces" issues, coming wireless technologies, and a lively Q & A session

Noted RF consultant James Stoffo’s recent PSW webinar, featuring the latest on the “white spaces” issues, coming wireless technologies, and a lively Q & A session with the audience, is now available for viewing. The webinar was sponsored by Shure.

Go here to play back and/or download the webinar.

The founder of Professional Wireless Systems (PWS), James is noted for wireless and frequency coordination work with countless high-profile performances, events and venues such as the Super Bowl, Broadway theatre, the World Cup, theme parks, NBA All-Star Weekend, the Radio City Rockettes and many others. He’s also currently involved with the development of new wireless technologies.

To check out the earlier webinar featuring James and Mark Brunner of Shure entitled “Solving Wireless Challenges Now & In The Future,” go here.

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Wednesday, September 28, 2011

Mythbusting: The Truth About Loudspeaker Wire

More dollars than sense?

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. “Litz” 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 speaker wire will exaggerate some real but insignificant parameter far out of perspective while compromising the real deal.

Forget the hype, what’s important for speaker 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” speaker 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 speaker’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 speaker 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 speaker 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.

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 speakers. (See Figure 1 at right.)

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 speaker 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 (above right) and Figure 3 (below right) shows 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 5 ohms. Meanwhile, Figure 3, while more complex, covers a similar impedance range, with a maximum around 16 ohms and a minimum around 6 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 1 (17/5) -1.09 dB -.57 dB -.12 dB Spkr 2 (16/6) -.81 dB -.42 dB -.09 dB

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 to 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 3-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 wire 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. {extended}

Tuesday, September 27, 2011

Components & Techniques For Getting The Best Results With Portable PA

So many options, and so many of them great. Here's what to look for and how it can work in meeting your specific application needs

Technically, lot of sound reinforcement systems are “portable.”

Get together enough hands and/or the right heavy equipment, and almost anything is portable.

But in pro audio, the term portable PA refers to compact systems that can be easily transported in a small truck, van, or even the trunk/back seat of a car, and then hand-carried (or hand-trucked or rolled) into a venue and set up quickly by one person (if need be).

Most often, we’re probably first think of loudspeakers - compact, 2-way models that are also sometimes referred to as “speakers on sticks,” but there are other components such as powered and non-powered mixers in the genre, as well as packaged systems that come complete with stands, cables and even microphones. 

Portable PA systems have been around for decades, and while the basic form has remained the same, they’ve come a long way in terms of performance capability and feature sets.

A popular model in the 1950s was the Knight system, a total package that included a 32-watt power amplifier, a choice of microphones, and even the option of a 4-speed record player!

By the 1970s, the genre had really moved forward in terms of quality and output, with many of the larger manufacturers offering portable components and systems.

One that stood out was the TAPCO Entertainer system from Electro-Voice, which earned a well-deserved reputation as an excellent portable PA system, comprised of the 100M powered mixer and 100S loudspeakers.

In fact, Entertainer systems can still be found performing today, and the heritage of the system continues at EV, exemplified by the new ZXA1 loudspeaker introduced earlier this year.

The 2-way ZxA1 is powered by an integrated 800-watt, 2-channel amplifier module, and it also has built-in steep crossover slopes and woofer excursion protection as well as a switchable high-pass filter allows for use with a subwoofer.

Further, it’s outfitted with XLR microphone and line level inputs, along with an XLR line level output for daisy chaining several amplified loudspeakers or a sub.

Staying with modern portable loudspeakers, the JBL EON has undoubtedly become one of the world’s all-time best-selling models over the past decade or so, and these days it seems like every other event has a stage flanked by Mackie SRM loudspeakers on stands.

Other portable components continue to evolve to a higher standard as well. Soundcraft just released the Notepad Series of multipurpose mixers, with all models outfitted with the company’s GB30 mic preamp and four stereo line inputs.

The Notepad 124FX also has an integral digital effects processor, which has a feed from every input and over 100 effects, including a pink noise and test setting.

Meanwhile, QSC Audio GX Series amplifiers have become a staple in driving portable systems, featuring Class H topology that’s based on key elements of the premium PLX Series, while the recently introduced Crown Audio XLS Series amplifiers integrate advanced crossover, limiting and DSP. 

When it comes to packages, there are a plethora of packages.

The Yamaha STAGEPAS Series offers passive loudspeakers, powered mixer and cables. A nifty facet is that the powered mixer is actually housed in one of the loudspeakers, and is detachable.

There are dozens of choices in this genre, ranging from professional caliber down to more “prosumer” models that even incorporate CD and mp3 players and a wireless microphone receiver within the loudspeaker system cabinet.

Sometimes, just a powered loudspeaker about the size of a loaf of bread equipped with a mic input will do the trick, and these can also come in handy for spot monitoring in support of larger systems.

Plenty Of Choices
Whether it’s independent components or a package, every portable PA system usually includes these basic items:
• Microphones (and sometimes direct boxes)
• Mixer and power amplifier, or powered mixer, or mixer and powered loudspeakers
• A pair of loudspeakers
• Loudspeaker stands, microphone stands
• Mic, line-level and loudspeaker cables
• Optional items include floor monitors, direct boxes and a mic snake

Let’s look more closely at each component.

Microphones. Step one is figuring out how many mics (and mic inputs) are needed. Unidirectional dynamic mics work well for speech, singers, guitar amps and drums, while unidirectional condenser stand-mounted mics are a solid choice for acoustic instruments as well as singers. Presenters and singer can also be outfitted with headphone mics, and acoustic instruments with mini clip-ons.

Power Amplifiers. How much power do you need? If you’ve already got loudspeakers or are going to purchase them, follow the recommended power guidelines provided by the manufacturer. Levels of a powerful system can always be tuned down to match the application, so be sure not to go too light in terms of output.

That said, here are some very loose guidelines:
• Speech-only system in medium room: 50 watts continuous per channel
• Folk music in a coffee shop with 50 seats: 25 to 250 watts
• Folk music in a medium-size auditorium, club or house of worship with 150 to 250 seats: 95 to 250 watts
• Folk music at a small outdoor festival (50 feet from loudspeaker to audience): 250 watts
• Pop or jazz music in a medium-size auditorium, club or house of worship with 150 to 250 seats: 250 to 750 watts
• Pop or jazz music in a 2,000-seat concert hall: 400 to 1,200 watts
• Rock music in a medium-size auditorium, club or house of worship with 150 to 250 seats: at least 1,500 watts
• Rock music at a small outdoor festival (50 feet from loudspeaker to audience): At least 1,000 to 3,000 watts

As previously noted, some power amplifiers now also offer built-in DSP, and these packages are increasingly competitive from a price standpoint.

The digital processing can serve as a substitute for active crossovers and delays.

Crossover-filter presets for specific loudspeakers can make it a snap to set up a multi-way system.

High-pass filters prevent harm to loudspeakers, often due to powerful lows in music, mic-stand thumps or accidental DC at the amp output.

Mixers. Self-powered or not? A mixer that incorporates power amplification is usually easier to carry and set up - it’s more plug and play.

An increasing number of powered mixers also include onboard effects, and some offer a graphic equalizer, which is useful for tuning the frequency response of the loudspeakers in different environments.

An advantage of a separate mixer and power amp is that if either one fails, only one component needs to be replaced.

Stand-alone mixers also tend to be more ergonomic for the user, and there are more size options - no need to purchase a unit with 8 channels if the maximum number of channels needed isn’t going to exceed 4 or 5. 

Be sure that the mixer has enough balanced XLR mic inputs to handle any possible application you have in mind for the system. RCA jacks for CD and MP3 players come in handy.

Loudspeakers. Full-range loudspeakers for portable PA are usually 2-way designs, either powered or passive. Most applications require just two loudspeakers, but there may be times when additional loudspeakers are needed.

Flexible connectivity is something to keep in mind.

For example, the latest generation of JBL EON portable loudspeakers have an XLR output where the output signal is selectable, either the whole mix may be looped to another loudspeaker (or sent to a mixing console), or simply the primary input for traditional “daisy-chaining” of additional loudspeakers.

These also offer one XLR/ quarter-inch combo connector and additional quarter-inch inputs providing input flexibility and the ability to mix multiple sources.

Some applications may require a stage monitor or two, in addition to two or more main/full-range loudspeakers. An increasing number of portable cabinets offer a side that angles the drivers more steeply upward toward the performers.

Typically, full-range loudspeakers are available with a choice of 12-inch or 15-inch cone woofer that is ported, joined by a driver on a horn or waveguide.

Horn dispersion (6 dB-down points) is commonly 40 degrees vertical x 90 degrees horizontal, or 40 degrees vertical x 120 degrees horizontal, but again, we see an increasing variety of coverage patterns available.

Keep in mind that the overall goal is to focus as much direct sound on the audience (and off of surrounding hard surfaces). Thus the polar pattern is important, particularly for indoor applications and especially in highly reverberant spaces like gymnasiums and some worship sanctuaries.

The frequency response of the loudspeakers should be wide enough to reproduce the sound source accurately.

For speech only, 100 Hz - 12 kHz is usually sufficient, but a guitar-singer application is better served by 80 Hz - 15 kHz and a rock band really should have 40 Hz - 15 kHz or higher.

Those frequency limits are typically 10 dB down or less from the level at 1 kHz.

Of course, the flatter the response over the passband, the more accurate the reproduction.

Boosting The Boom
This brings us to subwoofers. Dynamic music performances can really benefit from extended low-end energy.

The majority of full-range portable loudspeakers also offer at least one companion subwoofer, typically loaded with a 15-inch or 18-inch woofer.

These usually include mounts to accommodate stand poles, and sometimes have wheels on their back floor edge for easier transport.

An option for music applications is a subwoofer/satellite configuration, where one or two subwoofers on the floor provide deep bass, while two compact satellite loudspeakers on stands provide the rest of the spectrum.

Because our ears don’t localize extreme low frequencies, all sound “appears” to come from the satellites. The advantage of this approach is not having to lift large, heavy loudspeakers for positioning on stands.

Column loudspeakers are having an influence on portable PA designs, and that’s no surprise considering that the Shure Vocalmaster system, one of the most popular portable systems in the 1960s-70s, also featured a column approach.

A recent example is the Fishman SA220, which incorporates six 4-inch mid-woofers in a vertical line topped by a 1-inch soft-dome tweeter, housed in a cabinet measure just over 5 inches wide by 6 inches deep.

It’s self-powered and comes equipped with two mic/instrument channels with high-quality preamps, each with 3-band EQ, phantom power, built-in reverb, effects loop, notch filtering and phase controls.

A popular configuration is to place the SA220 behind the performer/group, where it can serve as both mains and monitors. Another common set is one SA220 per performer.

The advantages are that the performer(s) hear the same mix that the audience hears, and the sound level is more constant with distance than with a woofer/horn system.

Further Considerations
These days, most powered PA loudspeakers are bi-amplified: they have one amplifier for the woofer and another for the tweeter.

Advantages of bi-amplification include:
•  Distortion frequencies caused by clipping the woofer power amplifier will not reach the tweeter, so there is less likelihood of tweeter burnout if the amplifier clips. In addition, clipping distortion in the woofer amplifier is made less audible.
•  Intermodulation distortion is reduced.
•  Peak power output is greater than that of a single amplifier of equivalent power.
•  Direct coupling of amplifiers to speakers improves transient response—especially at low frequencies.
•  Bi-amping reduces the inductive and capacitive loading of the power amplifier.
•  The full power of the tweeter amplifier is available regardless of the power required by the woofer amplifier.

Loudspeaker cabinets come in both wood and molded plastic.

Wood cabinets are covered with either scratch-resistant heavy-duty paint or durable fabric that’s often called, in slang, carpet.

Molded plastic cabinets have also proven durable, can feature attractive styling, and are often lighter than their wood counterparts.

For example, a quality 12-inch, 2-way model with plastic cabinet might weigh about 25 pounds, while it’s wood counterpart could be double that figure. (You can come to appreciate lower weight after several nights of muscling loudspeakers on stands.)

Some plastic cabinets, however, have a tendency to “leak” low frequencies or resonate, possibly degrading both frequency and time response.

Most cabinets include a pole cup on the bottom that accepts a 1-3/8-inch or 1-1/2-inch pole for mounting.

The pole cups are vertically aligned, with some having a second angled cup to aim the loudspeaker down toward the audience when raised.

Two models in the new QSC Audio KW Series offer a proprietary Tilt-Direct pole cup mounting system.

A turn of a dial engages a 7.5 degree downward tilt of the loudspeaker, directing more of the acoustic energy toward the audience.

Moving & Mounting
Manufacturers of portable loudspeakers typically provide stands, either included or available as an option, and of course, these work well for the vast majority of applications.

Be vigilant about stands - they should be constructed of metal (usually aluminum), offer a solid tripod base and a secure collar, and be rated to comfortably handle more than the weight of the loudspeaker.

There have been some interesting developments in the world of stands as well.

The Ultimate Support Systems Air-Powered Series have an internal shock that lifts and lowers loudspeakers weighing 50 pounds and less with virtually no effort.

The Ultimate TeleLock Series has a collar that gives the user the ability to safely raise or lower the stand while a loudspeaker is on it. 

Both Air-Powered and TeleLock poles are also available for subwoofer mounting.

Portable PA loudspeakers can also be a good choice for permanent and semi-permanent installations.

For example, a church may find that its portable loudspeakers on stands are doing a fine job, so well in fact, that there’s the desire to get them out of the way and permanently mount them.

If you suspect this might be a future possibility, look for loudspeakers outfitted with mounting points that bolt onto steel cables or yokes that can be aimed as needed.

Don’t skimp on loudspeaker cables. Relatively low gauge cable is good practice, with #14 or #12 zip cord or zip cord, PVC or SJO cable of the same gauge seen most often.

Neutrik Speakon connectors are both more reliable than phone plugs (1/4-inch TRS), they lock into place, and pass more current.

To avoid tripping that could result in injury to a person or a loudspeaker or both, tape cables securely to the floor, using gaffe tape.

Extra cable length can be coiled and stored under the mixer or under the loudspeaker stand.

A very helpful item is a dolly, wheeled cart or hand-truck to transport equipment into venues. Consider lightweight tubular carts such as those from Ultimate Support Rock n Roller and Kart-A-Bag - being collapsible, they store easily in your car or truck.

Any additional racks and trunks should have casters for rolling, and lockable casters will keep them in place once positioned. Some trunks have trays and lid storage to make it easier to organize items.

Protective covers (usually available from the manufacturer or from Under Cover or Cloud 9) should be used to prevent damage of components, including for the loudspeakers. No one likes to see ugly, scratched-up cabinets, and it’s simply not a professional look. 

Placement Strategies
Typically, main left and right loudspeakers on stands are placed to either side of the stage/presentation area, aimed at the audience. (Figure 1, Option 1).

Make sure the loudspeakers are “in front” of the microphones, positioned toward the “dead” rear of the cardioid mic pattern, thus reducing the potential for feedback.

Raise the loudspeakers high enough to clear the crowd. Otherwise, people in the back will hear muffled sound because the crowd attenuates the high frequencies.

Also, raising the loudspeakers prevents blasting the front row of the audience.

Articulation is best if the direct-sound level is high relative to the reflected-sound level. This happens if you place the loudspeakers closer to the audience and take care to aim them.

An alternative to loudspeakers at either side of the stage is suggested by veteran audio consultant Ray Rayburn: Mount a single loudspeaker at one front corner of the audience, shooting across to the opposite corner (Figure 1, option 2).

Further, stack the two loudspeakers vertically (horn to horn) which narrows their coverage angle in the vertical plane. Clamp them solidly together.

The advantages of this arrangement include no comb filtering from hearing two loudspeakers at different distances, and a clearer overall sound with less reverb because of reduced ceiling reflections.

For sporting events, try to place loudspeakers so they aim across the playing field at the bleachers. This way, the players can hear what’s going on, and the people in the bleachers will absorb some of the sound and reduce reflections.

Final tips: wind screens on microphones outdoors really help eliminate noise, pop filters on microphones help prevent breath pops, which are particularly common among less experienced presenters, and a headworn mic allows a presenter to turn his/her head without getting off-mic.

Bruce Bartlett is a microphone engineer, sound mixer, recording engineer, and audio journalist. His latest books are “Practical Recording Techniques 5th Ed.” and “Recording Music On Location.”

{extended}

Thursday, September 22, 2011

Church Sound: How To Avoid Seven Common Mute Mistakes

You wouldn’t think one little button with one simple function would cause many problems but it does!

 
How many mute mistakes have you made? 

You wouldn’t think one little button with one simple function would cause many problems but it does! 

Time after time, people fall victim to its simplicity. Now you can find out how to minimize your mute mishaps.

The first mute mistake I ever made was when I was working at radio station 89.5 WFCI – Indy’s New Rock Alternative. 95 percent of the music was on CD with 5 percent being on vinyl. The CD players were set to stop after a track was played. As a DJ, that was great because I never had to bother with fading or muting a CD channel on the mixing board.

But then came the Eurythmics.

Their album was on vinyl and it was the first cut on the A-side.

The song lyrics ended and the last 20 seconds were instrumental. I announced the standard segue with the radio station call letters, the name of the song I’d just played and then did the pre-sell for the next three-song set. Somewhere along the way, I also started the next song.

I turned off my microphone and then something caught my attention. The song on the air didn’t sound right. It was a new song, the format was alternative rock (when it was up and coming in the mid 1990s), and I’d never heard it before. Even with all that being said, I knew something was wrong. Out of the corner of my eye, I saw the record player still spinning.

The second track on the vinyl album was playing at the same time as the one in the CD player. I’d been so used to the auto-stop of the CD player that I’d forgotten to mute the record player!

What is muting? 
The idea of muting is simple. Looking at each channel, your mixer will either have a mute or an on/off button. For simplicity, I’ll reference this as the mute button. Using the mute button, you silence that channel completely. This includes anything going to the monitors.

There are also assignable group/VCA mutes. In this case, you can route multiple channels to one control for volume and mute-ability.

Getting technical in 3…2…1…
A VCA-enabled mixer works differently than a non-VCA mixer. Without getting overly technical, the difference between the two is the point in which the “group control” takes control of the individual channels. 

A VCA-enabled mixer has a VCA (voltage controlled amplifier) on each channel that pulls the signal before the channel’s fader and controls the signal separately from the fader.  A non-VCA-enabled mixer gets the signal after it goes through the channel fader.

The more components a signal has to traverse, the more noise that can be introduced into the signal. Therefore, the VCA-enabled mixer doesn’t require the signal to go through the additional fader component of the channel.

Let’s talk common mute mistakes
There are seven common mute mistakes:

1) Not muting unused open microphones.  An open microphone that’s not being used can introduce noise and unwanted stage sounds into your mix. Also, in the right scenario, they can become the cause for feedback. Mute every unused open channel.

2) Muting instead of fading. Picture this, a soloist is singing to a recorded track. She stops singing and the CD track begins fading out. Suddenly, the music stops. “NEXT!“ That’s about how it comes across; “you’re done, I stopped the music, please exit the stage.” You are producing the whole service, so everything should flow together. Therefore, fade music all the way before you mute the channel.

3) Muting without fading. The best example of this is that you are playing the recorded track from #2 and you let the CD music fade naturally as per the recording. Then you mute the channel once the song is over.  Fifteen seconds later, you hear something coming from the stage. You have muted the channel but the monitor is still active and the CD is now playing the next track. Therefore, when playing media, remember it’s best to use both the mute and the fade in case you forget to hit the STOP button.

4) Un-muting with the volume up. The pastor starts talking and you’ve forgotten to un-mute his channel. Therefore, you un-mute it with the fader already at the nominal level. POW! His voice bursts through the air in a rather unpleasant fashion. [Unpleasant fashion?  Where did I pull that phrase from?] Instead, move the fader down, un-mute the channel, and raise the fader up to the proper location. This one isn’t so much about not making a mute mistake but how to best recover from it.

5) Incorrect muting with mute groups. Going back to the mute group / VCA comments I made earlier in the article, depending on your setup, you may or may not be muting the monitor sends as well. Make sure you have your pre/post fader channel options selected correctly and test them by muting the group during rehearsal.

6) Unassigned mute group channels. Do you have the background singers in a mute group? Are you sure you have them all in a mute group? If not, one background singer will quickly become a soloist. Double-check your routings and settings regarding groups.

7) Missing a cue. You didn’t un-mute a channel in time. I almost didn’t list this one as it seems more about paying attention and less about the mute button…but when it comes to that lil’ ol’ mute button, it’s front and center. You need to bring your A-game each time you are behind the mixer. Do that and missing a cue won’t be a problem.

Summary
Mute mistakes are the biggest mixing issues the congregation notices during a service.  One or two people might notice a mistake in the drum mix but when you miss a cue or jar them by suddenly muting an active channel, everyone will notice.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.

{extended}

Wednesday, September 21, 2011

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Curt Taipale heads up Church Soundcheck, a thriving community dedicated to helping technical worship personnel, and he also provides expert systems design and consulting services with Taipale Media Systems.

More articles by Curt Taipale on PSW:
Humor Files: Unintended Amendments To The Laws Of Physics

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Wednesday, August 17, 2011

SynAudCon Digital Seminar Slated For September 26-28 In Indianapolis

A rare opportunity to attend an invaluable seminar

SynAudCon will be presenting its much-lauded Digital Seminar this coming September 26-28 in Indianapolis.

The seminar, approved for 24 CEUs, is designed to shorten the learning curve while receiving a comprehensive introduction to digital audio, digital signal processing and digital audio networks.

Further, it provides a depth of understanding of everything from data formats to networked audio systems.

The seminar staff is a team comprised of SynAudCon leader Pat Brown, Steve Macatee of Rane and Bradford Benn of Crown Audio. The three of them form a tag-team approach to present SynAudCon Digital in a visually effective way.

Together they make learning digital audio fast, friendly and fun. This team not only has the theoretical grounding but also has applied these concepts in the field, so it is not just theory but also real world experiences that are being shared.

Price for the three-day seminar is $895. The seminar site is the Wyndham Indianapolis West, which is offering a room rate $92 a night (plus tax).

For more information about the SynAudCon Digital Seminar and to register online, go here.

Registration can also be handled via phone with Brenda Brown of SynAudCon at 812-923-0174.

SynAudCon

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Friday, July 29, 2011

Church Sound Recording: Start By Making The Right System Connections

A discussion of the correct procedures for connecting a recording device to your church sound system

At first glance, recording the audio of worship services might appear to be a relatively simple procedure.

Just connect a recording device to the sound system, set levels, press “record” and you’re ready to roll, right?

If only it were that simple.

Still, it’s true that capturing quality recordings is not rocket science. The most important thing is to get off to the right start.

Therefore, let’s start at the beginning and take a walk through the essentials.

Every sound system is a group of audio devices, and electronic signals flow through each device, from one device to the next. Each device is equipped with inputs and outputs.

Simply stated, inputs are connections that receive incoming signals, while outputs are connections that distribute outgoing signals.

A basic rule: the output of a device always feeds the input of another device.

Another basic rule: cables link outputs to inputs, and these cables need to be outfitted with correct connectors at each end.

It’s vital to understand the correct procedures for connecting a recording device to your sound system.

For our purposes here, we’re talking about two-channel recording devices such as CD recorders. However, note that the interconnection information presented here applies to any two-track recording device, such as memory recorders, PDA recorders, hard drive recorders, and others. (Even good ol’ cassette tape recorders!)

Making Connections
To add a two-track recording device to a sound system, connect the mixer (mixing console) output to the recorder input.

But which output?

Take a look at the back of the mixer, and you’ll see some connectors that are labeled with one of these names:
Master Output (Master Output)
Mix Out (Mix Output)
Stereo Bus Out (Stereo Bus Output)
Stereo Mix Bus
Line Out (Line Output)
Channel 1 Out, Channel 2 Out
Program Out (Program Bus)

Want to know a deep, dark secret? These terms all mean the same thing! (Audio people love to make life more complicated with extra jargon. Why agree on one name for something when several will do the trick?)

But for purposes of our discussion here, I will call these connectors Master Output, which is the most commonly used term.

The Master Output is a sum of all input signals feeding the mixer. And, the Master Output is already connected to the rest of the system - power amplifier (s), equalizer(s), etc.

Thus, because the Master Output is already in use, find a spare Master Output. What? There is no pare on your mixer?

No problem.

Purchase a Y cable, available for a very reasonable price from many music retailers and electronics stores.

As shown in Figure 1, a Y cable is true to its name - a short cable assembly shaped like a “Y.”

It splits one signal into two, so that a mixer’s single output can feed both the sound system and a recording device. If your mixer is stereo (with two output channels), two Y cables are required - one per channel.

Other Options
Some mixers offer dedicated “Recording Output” or “2 Track Out” connectors, and a recording device can be linked from there.

And note that some mixers include a built-in graphic equalizer. A recording devices should NOT be connected to the output of this equalizer. You don’t want your recording to be equalized!

Therefore, keep in mind to always use connector(s) in the signal path that come before the graphic equalizer.

Note that many mixers offer more than one type of output connectors. Again, why settle on one when more can serve to confuse and complicate things.

Possible connectors include:
RCA (also called Phono)
XLR (also called Three-Pin)
1/4-inch (also called Phone Jack, TS for “tip-sleeve” or TRS for “tip-ring-sleeve”)

All three types are shown in Figure 2. Each cable must have a connector that mates with your mixer (on one end of the cable) and a connector that makes with your recorder (on the other end).

Some Examples
Suppose your mixer offers a Master Output already connected to the sound system, as well as a spare Master Output, which is an RCA connector. Your recording device input is also an RCA connector.

Thus the cable connecting the spare Master Output and the recording device should have RCA connector at both ends. More specifically, a cable with RCA “male connectors on both ends.

If the mixer has a stereo Master Output, and the recording device has a stereo input, the connection can be made with individual RCA cables (one per channel) or a stereo RCA cable.

Another example: your mixer offers Channel 1 Out and Channel 2 Out, and these are 1/4-inch output connections.

Your recording device offers RCA input connections.

The solution is two Y cables, one for each channel, and each with a 1/4-inch “male” plug on one end and a 1/4-inch “female” phone jack on the other.

Plug an adapter cable into one of the phone jacks as shown.

Figure 3 shows this configuration.

And for clarity, note that only one channel is illustrated.

What if your mixer has one mono Master Output, but your recording device offers two inputs for stereo recording?

Time again for a Y cable, with the single connector end plugged into the mixer output, and the double connector end plugged into both inputs of the recording device.

Where To Go
I highly recommend purchasing cables already outfitted with the correct connectors for your system, rather than trying to make them yourself. This saves time and hassle, in addition to insuring a rock-solid, reliable connection.

Another option is to use a cable with two identical connectors; say, 1/4-inch connectors at both ends. Then add adapters (i.e., a 1/4-inch male to RCA male adapter) to make the correct connection.

One other option that should not be overlooked is to purchase the correct cables from your system contractor.

Keep in mind that with cables, like everything else, you get what you pay for. Inexpensive cables usually live up to the negative definition of “cheap - they can add noise to a system and fail earlier. Best to spend a little more for quality.

A final note: always be sure to label both ends of each cable according to what they plug into, so you can easily tell what is plugged into what.

Making labels is easy - just use masking tape and a felt-tip marker.

Congratulations, you’re connected.

Next time we’ll talk about hooking up devices with XLR or TRS connectors.

AES and SynAudCon member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.

More Church Sound articles by Bruce Bartlett:
Microphone Strategies That Produce Great Results With Church Choirs
Preventing “Hollow” Sound With Microphone Techniques
Identifying & Solving Microphone Problems

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Wednesday, July 20, 2011

What’s Under The Hood? Power Amplifier Sections, Connectors & Classes

A power amplifier is not just a black box that makes signals stronger; rather, this complex device has a number of functional sections, connectors and circuit classes that differentiate one model from another.

As audio professionals, the more we understand what’s under the hood of modern power amplifiers, the better we can make a wise buying decision.

What are the main sections or parts of a power amp?

Every power amplifier includes a power supply, an input stage, and an output stage. Most amps also have protection mechanisms; some have DSP, and a few have networking capability.

Let’s explain each feature…

Power Supply
Basically, a power amplifier uses the input signal to modulate DC from its power supply. This supply receives 120 volts AC from the mains outlet and converts it to DC to operate the transistors, FETs and MOSFETs and so on in the amp circuitry.

Two types of power supply are analog and switching. A typical analog power supply rectifies the incoming 50 or 60 Hz AC and low-pass filters it to create DC for the power amplifier circuitry.

A switching power supply converts the incoming AC to DC, switches it on and off at an ultrasonic rate, runs those pulses through a small, lightweight transformer, then rectifies and filters the waveform to produce DC. The switch-mode supply can be smaller and lighter than the analog supply, but is more complex.

Some amps have a separate power supply for each channel so that high demands on one channel don’t affect the other. A few also have a separate power supply for the input stage, which is the part of the amplifier that does not drive the loudspeakers.

It’s important that the power supply have enough power reserve to supply power for transients or signal peaks. That happens when the supply uses big filter capacitors that store energy, and releases it when needed.

If the amp is heavily loaded down (that is, it is driving a low output impedance), the power supply voltage may drop or “sag,” causing distortion. Using a separate power supply for the input stage prevents distortion in the input stage caused by the output stage’s supply voltage sagging.

Amplifiers of very high power draw lots of current through the power cable from the AC outlet. To avoid limiting the current that can be drawn, the AC power cable has to be heavy gauge and short.

And the circuit breakers feeding the amplifier’s AC outlet need to be 20 A or higher rather than 15 A. Low-current AC outlets can prevent the amplifier from reaching its maximum power output.

Input Stage
The input stage or “front end” accepts the input signals and feeds them to the output stage to be amplified. Here you’ll find connectors that mate with the input cables.

Level controls and any plug-in modules are part of the input stage as well. The level controls do not affect the gain of the amp; rather, they affect the input sensitivity – the input voltage required to drive the amp to full power.

Turning down an amp’s level controls does not make it less powerful or reduce its wattage rating. Instead, this requires the amp to have higher input signal to drive it to full power.

Put another way, turning down the level controls reduces the level to the output stage of the power amplifier. If you send the amp a high enough signal level, you can drive the amp to its full rated power even with the level controls turned down from maximum.

In fact, it’s standard practice to set the amp’s level controls for proper gain staging. Set up the sound system’s mixer so that signals peak around 0, then gradually turn up the power amp’s level controls until the sound is as loud as you want it. This results in the best system signal-to-noise ratio and headroom.

If you turn up the amp’s level controls to maximum, you’ll often hear mixer noise through the system loudspeakers because the mixer will have to be run at levels well below 0 on its meters.

Let’s look at other parts of the input stage. LED’s on the front panel indicate signal level, clipping, and overheating, so they can be used for diagnostics if you hear no sound or distorted sound.

Connectors in the input stage are on the back panel of the amp. You’ll see these types of connectors:

• 1/4-inch phone jacks: These are most often seen in portable PA or small band PA systems. TS (tip-sleeve) is unbalanced; TRS (tip-ring-sleeve) is balanced and is preferred for its rejection of hum and noise.
• Female XLR: This three-socket locking connector mates with a male XLR and provides a balanced connection. It’s used in portable PA and touring sound applications.
• Terminal block (terminal strip, screw terminals). This type is intended for permanent installations. It lets you eliminate connectors and their cost because the input cable is hard-wired to the terminal block.
• RCA or phono connectors: These are used for background music systems and home stereos.

All XLR inputs, terminal blocks and most phone jacks are wired balanced which rejects hum and noise on the input cable.

Output Stage
This stage amplifies, or increases the power of, the input signal up to a level sufficient to drive the loudspeakers.

In this stage are the power transistors (output devices), which tend to generate a lot of heat. Also in this stage are the output connectors which are wired to the loudspeakers.

Four types of output or loudspeaker connectors are phone jacks, five-way binding posts (banana jacks), Speakon connectors and terminal blocks (screw terminals).

• Phone jacks are inexpensive connectors for low-power applications. They are often seen in portable PA systems.
• Five-way binding posts provide a temporary or permanent, high-power connection to banana plugs, spade lugs or stripped wires. You’ll find these connectors in amps for touring sound.
• Speakons are a high-power, locking, cylindrical connectors used in touring sound.
• Terminal blocks are mainly used in installed sound applications to eliminate connectors and their cost.

Figure 1 is the back panel of a Crown I-Tech HD power amplifier, showing XLR, Speakon and five-way binding post connectors.

The best power connectors have low contact resistance. As contact pressure and contact area increase, contact resistance goes down. High-pressure contacts increase current flow by helping the current to penetrate through the surface films. They also increase contact area by flattening out the contact surfaces.

So when you use banana plugs, it helps to “stretch out” the ribs in each pin to increase contact pressure. Use a small screwdriver to bend the ribs.

Protection
The better power amps include circuitry that protects the loudspeakers and the amp itself from overheating and burning out. Some include a limiter to prevent the output power from getting too high and causing clipping, which can destroy tweeters.

Others prevent DC and ultrasonic signals from reaching the loudspeakers in the event of amp failure. Low-end units just blow a fuse or trip a circuit breaker if the current draw is too high, while high-end amps limit the output power so that the music doesn’t stop.

Cooling
The main cause of amplifier failure is overheating, so most amps include heatsinks and fans to keep the amp cool. In some units the fans come on only when needed. Some Class D amps tend not to get hot, so they don’t need fans.

DSP
A few models include built-in digital signal processing: compression, limiting, EQ, filtering, and so on. The advantages are:

• There is less gear to lug around in racks.
• EQ and limiting presets can be set up in DSP to work with specific loudspeaker models. Just select a preset that works with your chosen speakers.
• Some DSP includes diagnostics such as load monitoring to check for blown speakers, error logging, and so on.

Networking
Another feature in many modern amps is networking. Network-capable amps can be part of an interconnected audio network, so they can be controlled and monitored from a central computer. This beats walking around on stage trying to figure out which amp has shut down.

Amplifier Class
Let’s turn now to another aspect of power amplifier design. Amplifier class refers to the circuit design of the output stage, such as Class A, Class AB, Class D, and so on.

As a background for this section, remember than an audio signal has a positive half of the cycle and a negative half of the cycle (Figure 2).

Transistors are basically rectifiers; they can conduct (pass current) only during the positive or negative half unless they are biased by a certain amount. The bias can create a DC offset in the signal.

Here are the features of the most common classes:

Class A
• Has enough bias (DC offset) to shift all of the audio signal into the positive region in the output devices (Figure 3). As a result, positive/negative signal halves become more positive/less positive changes.

• Because the output transistors are always on, current flows at all times. This design generates a lot of heat. Some power is dissipated even when there is no music playing.
• Lowest distortion.
• Least efficient (typically 20 percent); wastes a lot of energy.
• Typically used in audiophile applications up to 300 W per channel.

Class B
• The output devices are in push-pull pairs: one device amplifies the positive half of the sine wave signal, and the other amplifies the negative half. Each device of the pair is on for half of the signal cycle (positive or negative voltage) and off for the other half of the cycle. Each device conducts for a half cycle.

• Much more efficient than Class A (typically 60-70 percent).
• Less heat.
• There is discontinuity at the transition point between transistor signals near 0 volts (Figure 4). This results in high “crossover distortion” or “switching distortion” – low sound quality.
• Typically used for pocket radios or clock radios.

Class AB
• Both output devices in each pair are biased slightly on which reduces crossover distortion. Each transistor operates slightly more than half the cycle but is off for a fairly long time, which reduces heat dissipation. (Figure 5).

• About 50 percent efficiency.
• Low distortion.
• Typically used for home stereos and pro audio amps up to 600 W per channel.

Class G or H
• Both of these classes include two DC rails (DC supplies) of low voltage and high voltage. The high-voltage rail is switched on only when the input signal demands it, which reduces the amount of heat in the output devices. The power supply is signal-controlled.

• In Class G, one output stage is fed by the low voltage rails and another stage is fed by the high-voltage rails. The low-voltage stage is always on, and the high-voltage stage turns on only when the signal exceeds a threshold level. Class H uses only one output device stage which is fed variable supply voltages depending on the amplitude of the signal.
• Fewer output devices and less heat sinking are required, which reduces the weight and size of the amp.
• Tends to have elevated distortion at high frequencies due to the switching.
• Typically used for LF or MF applications from 400 to 3000 W per channel.

.Class D
While the other classes operate in a linear (analog amplifier) mode, Class D amplifiers run the output devices as switches (on or off) at an ultrasonic frequency to create a continuous series of square waves or pulses.

The input signal is made to modulate the pulse width, an operating principle called pulse width modulation. The pulses’ high frequencies are filtered out, and this filtered output (an analog signal) drives the loudspeaker.

• Very efficient (90-95 percent). That’s because the transistors are either on (with high current and no voltage) or off (with high voltage but no current). So there is very low power dissipation compared to linear amplifiers.
• Tends to be small and light because of less cooling and fewer output devices.
• Typically used for high-power car stereos and pro applications where heat, weight and size are considerations.

Class I
• Class I or BCA (Balanced Current Amplifier) design developed by Crown is an efficient system based on Class D switching amplifier technology. The amp needs only a little AC power to operate.
• According to the manufacturer, a BCA amp generates one-tenth the heat of conventional amplifiers so it can work with much less air movement. This reduces fan noise, heatsink size, filter maintenance, and failure due to heat, so the amp can be small and light.
• A BCA amp re-uses the energy returned from the speaker rather than dissipating it as heat or forcing the amp into premature current-limiting. This helps BCA models handle 2-ohm loads without shutting down.

I hope we’ve clarified some of the differences among power amplifiers. They are another component in the chain of audio equipment that we need to understand well.

Take the PSW Real World Gear Tour of the latest power amplifiers.

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

 

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Wednesday, June 29, 2011

Everything You Wanted To Know About Microphone Splitters

As your sound system expands, it will eventually be necessary to provide additional mixes from locations other than the main mix position

If church sound system has evolved along a familiar path, what started out as a pretty simple, small group, “sound-on-a-stick” has gradually become more and more sophisticated.

Many have replaced that powered mixer with separate components and added a snake to allow for a mix position in the listening area.

Words like “direct box”, “balanced”, “low impedance”, “crossover”, etc. have become part of the sound team’s vocabulary as they strive to provide today’s expected level of sound quality and production – for both the listeners and the performers.

In this article we just might add some additional terms to your audio vocabulary as we discuss microphone splitters.

When To Split?
As your sound system expands, it will eventually be necessary to provide additional mixes from locations other than the main mix position.

Although it’s possible to provide a separate monitor mix from the main console, a person located nearer to the performance area can hear what the performers hear, see their cues more easily and just generally be able to provide a better monitor mix.

Or you may be called upon to provide a separate mix for recording or broadcasting your performances. That mix will be at its best if the person providing it is isolated from the confusion of hearing the live sound. In any event, you’ll most likely need to split your mic signals and feed more than one mixing console.

Impedance
Proper design of signal flow in an audio system dictates that low impedance outputs (mics) feed high impedance inputs (mixers). When a signal is split to be sent to more than one mixing console, the input impedances of those consoles provide additional paths for the electrical current.

This actually increases the overall load presented to the mic signal and limits how many times it can be split without degrading tone or introducing distortion.

Microphones can usually be split to up to three, and in rare cases even four, destinations without the use of electronics. The number of splits that can be accomplished depends on the application, impedances present in the system, length of the cables and the quality of the components used in the splitter. This is called passive splitting – no power required.
Active electronic splitters will most likely be required when splitting microphones to four or more consoles.

There are two types of passive splitters: parallel and transformer isolated.

Parallel Splits
The simplest form of splitter is the parallel type split. This involves taking a mic cable and simply “Y” connecting the plus, minus and ground wires to two or three other cables.

Although this method successfully connects the mic to multiple mixing consoles, be aware that it connects the consoles directly to each other as well.

Most modern consoles will behave rather nicely configured this way but sometimes, mostly in older units, the consoles will interact with each other. This can happen when making adjustments at one console causes the electrical characteristics of its input circuitry to change.

These changes may reflect through a parallel split and possibly upset the input circuitry of the second console.

For example, if adjusting the gain on console A causes the DC voltages of A’s input to change, this will show up at the input of console B. In some cases, this might cause the gain to change on console B, definitely a bad situation.

If you are considering a parallel split, you can test for this by making up a microphone Y cord, attaching a mic to both consoles and trying it out.

Make adjustments on each console, especially to the trim or gain section, while listening to the other. Turn the phantom power on and off each console. If you don’t notice any changes in volume or quality of sound, then a parallel split should work OK. (Adding the actual lengths of cable to each leg of the split can present other issues, such as added capacitance and increased susceptibility to RF interference.)

However, if the consoles are interacting, or if you are traveling and will be splitting to unfamiliar consoles regularly, then it would be wise to accomplish the split by using transformers.

Transformer Isolated Splits
In a transformer splitter, the microphone is wired straight through to a “Direct Out” and also to the input of a splitting transformer. (See Figure 1 below.)

This transformer has a 1:1 turns ratio and its output side is connected to the second or “Isolated” split output. (Transformers with two or more secondaries are used for achieving more than one iso split.)

The transformer will pass the microphone’s AC audio signal but will block DC voltage in either direction. This helps prevent interaction between the consoles.

One of the outputs is usually wired as a direct connection because the transformer will also block phantom power (DC). Remember to plan on connecting this direct leg of the split to the console that will be providing the phantom power.

Another benefit of using a transformer split is that it increases each leg’s ability to reject interference by improving the “balanced” characteristic of the line (called “Common Mode Rejection” or CMR).

A disadvantage of this type of split is the added expense of the transformers. High quality transformers are essential for providing proper shielding and for preserving the frequency response of the mic signal - don’t cut corners here!

Ground Lifts
Not all grounds are created equal. In fact any time two pieces of audio gear are plugged into grounded outlets, their actual resistance to earth ground can vary quite a bit - even when the outlets are on the same circuit.

This can be due to the designs of the power supplies, the length of the cable from the outlet to the service box, poor or oxidized connections within the outlet boxes and service panels - anything that can affect the resistance in the electrical path to ground.

Even when using a transformer split, a problem can arise when the consoles’ grounds are connected directly to each other and here’s why: If console A “sees” a lower resistance to ground through its connection through the splitter to console B, then part of its AC ground return current will take that path of least resistance and AC current flows in the shields of the cable, through the splitter, and over to console B. This is called a ground loop.

Now, instead of the shields providing a defense against unwanted interference, they are carrying 60 Hz AC and radiating it directly into the signal conductors that they are supposed to be protecting! This is why ground loops produce hum.

Although it might solve the hum problem, you should NEVER use a three-prong ground lifter on the AC power cable of either console! This is not safe and can present an electrical shock danger to the people using the system.

A better solution to this problem is to break the ground connection of one or more channels between the consoles. This is accomplished by disconnecting each offending ground connection at one end (usually the splitter) and leaving it connected at the opposite end. The shield for that channel will continue to work because it is still grounded at one end.

Some technicians will clip all of the split grounds, leaving them permanently disconnected but it’s better to install ground lift switches for each channel or use lift adapters when necessary.

This way, the ground can normally be left connected but lifted if there’s a problem. Also, if the main console is unplugged or disconnected, the grounds can be left connected to the split console making it usable.

This is sometimes needed when churches or schools with large format main consoles decide to keep some of the mics split to a smaller, less complicated mixing console. This allows a less technically proficient person to operate the system when basic sound is needed for a small service or assembly.

Remember that a direct out or passive split will not pass phantom power with the ground lifted at either end and a transformer isolated split will not pass phantom power even with the ground connected at both ends

Active Splitters
In the above examples of parallel and transformer isolated splitters, the signals are split without using any powered electronic circuitry.

As discussed, there are issues involved regarding impedances, frequency response and console interaction. These problems can essentially be eliminated by splitting the signals with active electronics.

In an active splitter, the mic signal is applied to an amplifier circuit. This circuit is actually a mic preamplifier similar to the input of a mixing console.

It can be designed to provide a wide and flat frequency response and present an optimum and constant impedance to the microphone.

Gain adjustment can also be provided at the splitter to boost weak signals before they have to make that long trip down the snake. This improves signal to noise ratios.

Phantom power can also be provided for at the splitter which eliminates the issue of deciding which console has to provide it when designing passive splits.

Sometimes an active splitter is used to feed line level signals to several destinations such as a bank of amplifiers, tape deck duplicators, headphone monitors, etc.

In this case, it is usually referred to as a “Distribution Amplifier” or “DA”.

The output of this mic-pre is fed to several more amplifiers that feed the various split outputs required. These separate amplifier circuits prevent interaction between the input and outputs (called “buffering”) and can be fine tuned to produce the best possible results.

Separate, buffered amplifier outputs also eliminate problems associated with the consoles interacting with each other.

Also, all outputs can be separately transformer coupled which greatly improves their balancing quality or Common Mode Rejection (CMR) of the output line. This makes them less susceptible to the effects of outside interference.

Electronic drive to each transformer can be designed to be extremely low impedance in nature which further improves noise rejection and response, especially bass frequencies.

These advantages over a passive circuit allow a signal to be split to more outputs and with better overall frequency response.

Al Keltz is a technical writer who works with Whirlwind USA.

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Friday, June 24, 2011

Sennheiser’s RF Wireless Sound Academy Seminar Journeys To Three Major US Cities

Unique single-day workshop led by top industry experts covers creating trouble-free RF wireless operation in even the toughest environments

Sennheiser has announced that it will be offering its popular RF Wireless Sound Academy Seminar in three major US cities: Tuesday, June 28, 2011, in Richmond, Virginia; Thursday, July 21, 2011, in New York, New York; and Thursday, September 15, 2011, in New Orleans, Louisiana.

This single-day workshop is designed to teach attendees how to plan for trouble-free operation of multi-channel wireless microphones and wireless personal monitoring systems in even the toughest environments. Alongside its popular presenters, Joe Ciaudelli and David Missall, the event in New York will feature special guest Volker Schmit.

Schmit is the RF engineer who spearheaded the creation of Sennheiser’s most successful and innovative products, including the popular evolution wired, evolution wireless, MKH, MKE, and 3000 and 5000 series.

The RF Wireless Sound Academy event will allow attendees to learn or enhance their skills through classroom and hands-on instruction on the following key elements: transmitter and receiver technology, antennas and distribution systems, wireless monitoring systems, system planning, frequency coordination and how digital television and new FCC policies may affect wireless microphone users.

Ciaudelli and Missall are industry veterans in RF wireless operations. Ciaudelli began his career 20 years ago as a Sennheiser applications engineer, and has provided frequency coordination for large multi-channel wireless microphone systems used by Broadway productions, major theme parks and broadcast networks, including the frequency plan used for NFL football games.

Missall is a National Market Development manager for Sennheiser Electronic Corporation and has provided RF wireless solutions and trouble shooting support for several broadcast networks and organizations including CNN, ESPN, Univision Network, Fox Sports, CBS, Speed Channel and NASCAR.

Registration costs $149 and includes, lunch, workshop materials and a $50 coupon toward any Sennheiser product.

For more information and to sign up, visit the Sennheiser RF Wireless Sound Academy registration page.

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