Article

Tuesday, February 07, 2012

One-Stop Shopping: Captain, What Does It Mean, This Term “Full Production”?

The sound company’s job is to advance the show with the artist and show up with a rig. Not so when the full production falls into your lap.

Sound companies handle “one-off” shows every day. It’s usually formulaic, and after a while, we do it by rote. 

But what happens when the client wants one-stop shopping? This is also known as “full production” or “turn key service,” and it’s quite a bit more involved than an average show. Generally months of planning and coordination are needed, as well as work with a number of subcontractors. It just can’t be done by the seat of the pants.

Normally, when a sound company is hired for a show, the client is a promoter or a venue. They provide the stage, they provide the power, and they provide the labor. The sound company’s job is to advance the show with the artist and show up with a rig. Not so when the full production falls into your lap.

Particularly for large, multi-stage festivals, hiring a single source to handle all the entertainment elements of the event is almost a necessity. The event director has too many other things to handle to have to worry about the details of his entertainment. 

Steve Rosenauer, director of the St. Mary’s University Alumni Association Fiesta Oyster Bake in San Antonio, Texas, once told me his definition of full production: “As a client, full production means working with a knowledgeable and experienced company that can produce a turn-key operation with regard to organizing, building and operating the necessary staging, sound, lights and equipment needs, with all meeting the negotiated specifications of the event as well as the bands. A company that does this can greatly enhance the quality of the event and provide a solid peace of mind to the entertainers and the event organizers.”

For the purposes of describing the process of a full production event, I will use the Fiesta Oyster Bake as my example. It’s a two-day, six-stage festival which kicks off San Antonio’s annual Fiesta Celebration every April. Fiesta has been ranked as the second largest party in the U.S. (Mardi Gras being first) by the National Meeting Planners Association. (And yes, they bake tons of oysters!) For years, our company, Sound Services, worked with this event. (Note that we recently chose to close the company for reasons completely unrelated to business.)

PREP MAKES PERFECT
In order to be ready by mid-April, we would start working in November. To be fair, we had been doing this event for nearly a decade, and had amassed a team of subcontractors with whom we were all very comfortable. Until a company gets to this point, preparations probably need to commence even sooner.

In November, we would begin talking about what our needs were going to be. Because city electrical inspectors were involved, we checked the City Code Compliance for any new electrical requirements. For example, one year (and for the first time), we were required to ground all of stages to the audio power distribution services, as well provide non-conductive covering of all power cables running in public areas. Not fun to discover things like this at the last minute!

We provided staging, sound, lights, backline, labor and all technical personnel for the festival. Because the client uses many more generators than just ours, they made those arrangements, but they used our generator provider so we were assured that power would not be a problem. The generator provider also stayed in contact on any change orders he received that might affect us.

Also by November, the client usually had more than half of the talent booked, so we got a vague idea of what to expect from headliners’ riders. By December, we started talking with our subcontractors, discussing what had changed from the previous year, giving them the firm dates, and requesting a firm price by January. 

After ringing in the new year, and still four months out, it was time to nail down the financials. Be very meticulous with this process!  Everything must be committed to paper, and math triple-checked in order to avoid any mistakes that could cost an entire profit margin.

It’s doubly vital to get this facet correct in the first year with an event, because the client will base future projections on those first year costs. Therefore, a mistake probably can’t be made up for next year.

Only after every cost is defined and listed, as well as those of the subcontractors, should the price be committed to the contract submitted to the client. Note: the one thing we found most often overlooked is the cost of a production manager. The hours and hours you spend working on this shouldn’t be done for free!

WORKING IN EARNEST
We would submit our contract on the first of February, with the understanding that requests on artists’ riders would probably cause an increase in total price. By this point, the client had all talent booked, so we could start working in earnest to learn just what those extra costs might be. My goal was to have all this information by the 15th of the month, still two months out.

There is a negotiation with contract riders and advancing the show that can - with some diplomacy - help reduce the number of additional line items for your client. Because most headliners’ riders are based on arena shows, for example, they will often concede some lighting instruments. 

On the other hand, you don’t want artist representatives to think your client is cheap, so know where and when to stop asking for concessions. It’s important to manage your client’s expectations in this regard as well. Most touring artists also understand that festivals differ from concerts, so if the stages are adequately stocked to begin with, most of the added line items will be for backline and spotlights.

Once we determined all of the additional artist-related expenses, we submitted a contract addendum. This addendum should include absolutely everything - a. client will begin to lose confidence if presented with more than one price addition. His budget is set in stone by this time, and your math errors and oversights are not his fault.

MINIMUM OF 40
Because Sound Services was responsible for the entire Oyster Bake Festival, not just the two stages we were physically covering, it was imperative that we advance the show with every artist. In this case, we’re talking a minimum of 40 bands, which made for a lot of work. But it accomplished several very important things. 

First, we got a thorough look at the requirements of every stage, and were assured that each subcontractor could adequately cover the entertainment line-up. If there was a particularly tough set change on a stage at a particular time, we could arrange to have extra help on hand at that time. 

Second, it gave each artist a feeling of confidence to know that individuals who care about their performances run the festival. Third, we established consistency in the way the artists were handled. The subcontracting sound companies all appreciated this.

And fourth, we could apprise artists of the “special quirks” of this festival. For example, it’s held on a university campus that is, itself, located in a neighborhood, not on a major thoroughfare. Getting to the venue is difficult when 80,000 other people are also trying to do the same, and there is no alternate route.

Sometimes when we told first-time performers to allow three hours to arrive, some balked, but we remained adamant. The ones who didn’t believe us were invariably late, which is a no-win for everyone. (By the way, returning artists were never late!)

Further, artists can’t drive to any stages except the main one, because they’re all positioned among campus buildings. For this reason, full backline was provided at every stage, and musicians were discouraged from bringing more gear than they absolutely had to have. To accommodate this, the university set up a team of volunteers to ferry musicians and their gear to the stages. It took several years to streamline this process.

Once all the advance work was complete, we created stage plots and input lists for every stage, and for both days. These were then dispatched to the sound companies working the festival with us.

GETTING CLOSER
A pre-production meeting with the festival committee and all stage managers was held six weeks to two months out. Each committee reported on their progress and, although we weren’t involved in things like pizza ovens and beer sales, it helped us to know what was going to be happening around us. 

Entertainment production is an important part of this meeting, and we made it a real bonding experience. Construction of “Stage 1,” for example, meant an entire campus parking lot has to be closed two days prior to the event, and thus it was critical that the timing be executed properly by the university security department. 

We also got to meet the stage managers and orient them as to what was expected of them. These folks are critical for smooth-running shows, and we let them know that. While their duties are light, the few things we needed from them are all important to the show.

Other things covered in this all-important meeting were issues of water, green rooms, use of volunteers (there are hundreds!) and getting musicians to the event and their respective stages. Over the years, and learning from our mistakes, we developed methods to efficiently accomplish these tasks, but until you’ve worked with an event for a long time, these issues are extremely important to thoroughly think through. For example, from experience we all learned that as much water as we thought we needed - double it!

At this time, we also walked the campus with the festival director, making note of things like trees that needed trimming or light poles tp temporarily remove. (Grounds and electrical departments need to be notified in advance to schedule work like this!)

WHO’S DOING WHAT
By one month out, we had a firm grip on exactly who was doing what. For example, if there was a sound company short a monitor engineer, this was the time to step in and lend a hand. Each subcontractor provided us with a list of personnel and how many vehicles (and of what type) they would be bringing on site. One aspect to double-check: be sure each contractor is providing enough people. For example, backline duties done properly for six stages requires more than two techs.

At this point, we would tally up all production people (including stagehands and spotlight operators) and provide the festival director with the number of parking passes and wristbands needed. Remember - on a multi-day festival, each person might need a fresh wristband each day. We also padded this number by a few more to replace ones that were inevitably lost.

Very key: the best technical person on staff must be in charge of production management. Even with the best preparations, all kinds of little things can go wrong, especially at multiple stages. One person not involved in production at any one stage has to be free to fight the fires, and this person should be well versed in technical knowledge as well as diplomacy. 

Our production manager for the festival spent each day traveling between stages, providing a break to a beleaguered engineer here, dealing with a power problem there, handling a recalcitrant band engineer somewhere else.  He also carried a radio for instantaneous contact. And, this person must have healthy legs – in a very crowded festival, a golf cart won’t work!

Three weeks out, we assembled packets for all of the subcontractors involved.  These included parking passes and wristbands, a map of the campus showing all stages and parking areas, a complete schedule of the event, and for the sound providers, stage plots and input lists. Load-in times were also provided.

Scheduling personnel is critical at this point. We staggered the load-in times so that we could make the best use of our stagehands. Stagehands have a four-hour minimum, and each is usually scheduled to work at more than one stage during a shift.  For load-out, we scheduled a much larger number of stagehands. This schedule was then filed with the labor company as a written work order, and note that this also included spotlight operators as well.

IT’S SHOWTIME!
Two days before the festival, we began to build the stages. The provider arrived with semi-trucks loaded with staging, and we again walked the site with the festival director, spotting the stages, front-of-house risers, spot towers and security towers.

The day prior to opening, we loaded in at our two stages, which then left us free to address the mayhem of everyone else loading in the next morning. The lighting contractor also loaded in with us in order to be out of the way, and this left the lighting directors free to work with headliners who might arrive early. On-site security was continuous at this point.

Day one of the festival would arrive, and we were free to conduct headliner soundchecks on our stages. Fortunately, the first act didn’t begin until 6 pm, so the atmosphere wasn’t too stressful.

The production manager was also available to address the various surprises that unfold, as they invariably will. This is where months of planning pay off and you can look really good to the client, who’s running around putting out all kinds of fires while his production people are calmly doing their jobs.

If all subcontractors are competent and well prepared, the event should run like an average one-off show. One caveat, however: it’s still a multi-day, multi-stage festival, with thousands of people swarming all over, so competent, well-informed stage managers become critical to your existence. 

They aren’t needed to get artists on and off the stage – we had already planned that out. They are most definitely needed to competently answer artist questions - “Where are our food coupons?” and “Where is our dressing room?” and the like. They also kept lots of water on ice, and plenty of ice in the ice chests.

The most important thing stage managers did, however, was manage the radios. Each stage had a radio, as did the production manager and the lead backline technician, and they were on a common channel with the event director. 

As the production staff performed its various tasks, we didn’t have time to monitor a radio, but when we had a problem or needed help, we simply asked a stage manager to contact whomever we needed. Previously we carried individual radios, but learned that this alternative approach worked so much better for everyone, plus it gave the stage managers a sense of ownership of their jobs as well. 

The best advice: “be round.” Roll with the punches and don’t get too excited by the inevitable little surprises that spring up. Make the production of entertainment as smooth as possible and don’t create tension or problems. That’s a big reason you were hired!

THE AFTERMATH
When it’s all over, the results of diligent planning and scheduling should continue to pay off. We found that handling a large number of stagehands at the end of the festival worked best if we arranged for the crew chief to assemble all of them at a pre-arranged site and make assignments from there.

Stagehands were first dispatched to the stages manned by our subcontractors, then re-routed to our stages last.  We always got this show loaded out within our four-hour labor minimum, by the way.

The production manager continued to make a circuit of the stages, being sure each stage had its allotted stagehands and collecting any left-behind belongings. We later attempted to repatriate these items with their owners.

When all the dust cleared a week or two later, we sat down and created a recap of the event, and this went into the file for next year. We also sent this recap to the festival director. Included were a summary of any issues that came up, general incidents, what worked well and what didn’t, and suggestions for improving next year’s event.

By working with the client in this fashion, we made ourselves a part of the event team, and enjoyed a multi-year contract. We also ingratiated ourselves to our subcontracting partners, who appreciated the work and reciprocated when appropriate. 

It’s just good business to develop this kind of working relationship with your clients and fellow business people, and it leaves you feeling pretty good about yourself as well.

Teri Hogan is a long-time audio professional and was co-owner of Sound Services Inc., a sound company based in Texas.

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Thursday, February 02, 2012

Church Sound Files: The Reason For “Bad Sound” May Not Be The Sound System

Three factors, roughly equal importance, play the key role in good sound - and “two out of three” isn’t good enough

Many things around us are getting better. Computers are faster, televisions have more resolution, and dishwashers are quieter and more powerful than ever.

But with all of our digital wiz-bang processors, technology has been unable to eradicate “bad sound.” Why is this so? This short piece is an attempt to shed some light on three possible causes, two of which have been completely unaffected by the technological revolution.

The goal of most sound reinforcement systems is to deliver high quality sound reproduction to the listener. While we would like to think that a high quality sound system guarantees this, it does not.

The quality of the reproduced sound will only be as good as the weakest link in the reproduction chain. Let’s examine some of the major “links” individually.

The Room
The room is a major factor in the reproduction chain. Most large spaces are hostile environments for sound systems, unless they have received special attention from a professional and a considerable financial investment from their owner. Good acoustics doesn’t just “happen.” It is the by-product of careful planning.

A quality sound system may radiate an exceptionally high-fidelity sound field into the room. Unfortunately, most of the radiated energy will create acoustic events that detract from the listening experience. While small rooms have their share of acoustic problems, these problems pale next to the late reflections, reverberation, and energy build-ups encountered in large spaces.

If your sound system doesn’t sound good, ask yourself the question “What have I done to provide a good acoustic environment?” If the answer is “nothing,” then you got what you paid for.

The Sound System
Of course, a good sound system is a vital link in the reproduction chain. But this doesn’t just mean expensive equipment. It means that equipment that is suitable for the environment has been selected and implemented by someone who understands the compromises involved in large room reinforcement systems. Money can be wasted on “features” that offer no real benefit for the large room environment.

The vast majority of auditoriums that I have visited are not suitable for multi-channel formats such as stereo, surround sound, etc. since each channel must be delivered to all listener seats. Loudspeaker placements that are optimal for stereo reproduction are horrible choices for single-channel systems.

Even with monaural systems, “first choice” loudspeaker placements often create problems with sight lines and aesthetics, and are therefore ruled out by venue owners. Multiple loudspeakers must overlap somewhere, and there will be sound problems in these areas.

A properly designed system will often sound bad in the aisles – the very place where casual onlookers will stand to evaluate it. We all have good sound at home, but the rules change as the listening space grows. Intuition that is not filtered through the proper large-room principles leads to errors.

Sound system designers are often forced to compromise away the performance of the system to make it fit aesthetic concerns, budget limitations, and fashion trends within the industry.

The Operator
I’ve intentionally saved this one until last. The most overlooked link in the chain is the end user of the system. This includes the mixer operator and any supporting staff, such as those who run the monitors and place microphones.

A monitor system that is too loud will dump excessive energy (usually low/mid frequency) into the audience area. This excess energy will upset the spectral balance of house sound system, tempting the front-of-house operator to compensate by over equalizing (usually in the form of high frequency boost). This results in a reduction in gain-before-feedback and an unnatural sounding system. Microphone placement is equally critical, as is an understanding of the shortcomings of various miking techniques.

If a lapel mic could sound like a hand-held, then no one would use hand-helds. The overhead drum mic that captures the cymbals also captures the stage monitors and “spill” from other instruments, as does the vocal mic used at arm’s length. And that “mellow” bass guitar sound that the musician likes in the practice hall turns to “mush” in a large space, where increased definition provided by the use of a pick and brighter strings may be required.

These factors and many more “eat away” at the sound quality of the system as a whole. A good mixer operator will evaluate and optimize the sound of the instruments individually before allowing the band to perform as an ensemble. There’s no room for democracy here – effective mixer operators learn to say “no” and “be quiet.”

A question that I recommend for an interview of prospective mix personnel would be “What will you do if something starts to squeal?” If the answer is anything other than “Turn the offending channel down slightly until I figure out what the problem is” move on to your next applicant. Filters implemented in desperation do nothing to preserve sound quality.

Modern mixing consoles pack a considerable “wow factor.” It’s fashionable to sit behind a large one and move knobs all of the time. But doing so doesn’t make one an engineer. Completing an accredited academic program or piloting a locomotive does. The decision as to which console to purchase is often made with no consideration as to whether anyone at the facility will be able to operate it. The result? Bad sound.

I have personally witnessed the performance of many good sound systems ruined by bad rooms and incompetent operators. I have also seen skilled operators “salvage” the sound reproduction in situations where the room and system were less than optimal.

The performance of a sound system is only as good as its weakest link. Unfortunately, all of the links that I have mentioned are of roughly equal importance, meaning that “two out of three” isn’t good enough. Good sound requires all three.

Experienced, well-trained audio people realize this and are there to help you find your weakest link. Pay for their advice and follow it.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world. Synergetic Audio Concepts (SynAudCon) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, SynAudCon is dedicated to teaching the basics of audio and acoustics. For more information, go to http://www.synaudcon.com

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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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Wednesday, January 25, 2012

Loudspeaker Sensitivity: What’s A Watt Anyway?

Shedding some light on the sensitivity specification and how it may translate to the real world performance of a loudspeaker system

The specification of a loudspeaker’s sensitivity is probably one of the most common, yet perhaps one of the most misunderstood.

It’s common to see the magnitude response of a loudspeaker system reduced to a single number as a sensitivity rating.

This is perhaps at the heart of the confusion.

One would think that this metric should give some indication as to how loud a particular loudspeaker will be when reproducing a signal.

One may also think that two loudspeakers with the same sensitivity rating will be equally loud when reproducing the same signal. Each of these assertions is only partially true.

A loudspeaker’s sensitivity can give an indication of its output level but only for a signal with a specific bandwidth and spectral content.

Similarly, two loudspeakers with the same sensitivity may not output the same SPL when excited by the same signal if the frequency response limits of the two loudspeakers are different. Let’s look at the underlying cause of each of these effects, bandwidth, and the role it plays, and also look at why sensitivity may no longer need to be referenced to a watt.

According to the standard IEC60268-5, a loudspeaker’s sensitivity is determined by measuring its output when driven by a band limited pink noise signal with a Vrms equal to the square root of the loudspeaker’s rated impedance and referencing this SPL to a distance of 1 meter.

The bandwidth of the pink noise is limited as a function of the effective frequency range of the DUT (Device Under Test). This is done to ensure that the test signal is confined to a portion of the frequency spectrum in which the DUT has appreciable output.

If a particular loudspeaker isn’t capable of reproducing signals below 150 Hz it does no good to excite it with such signals other than to generate heat. The same holds true if the loudspeaker can’t reproduce signals above some high frequency limit.

A high-resolution transfer function measurement of the DUT can also produce an identical sensitivity rating when the average magnitude is calculated on a log frequency basis.

As an example, let’s look at Figure 1. Here we see the on-axis response of a loudspeaker. Its sensitivity rating is shown as the straight line.

The length of this line coincides with the upper and lower frequency limits of the pink noise used to measure the sensitivity rating.

The spectral content of this noise signal is shown in Figure 2.

If a signal with different spectral content, but the same broadband level were used to drive this loudspeaker, would it result in the same SPL as the sensitivity?

It’s impossible to determine this without knowing both the spectral content of the signal and the response of the loudspeaker. (Note that 20 Hz to 20 kHz, or in the case of Figure 1, 110 Hz-8.3 kHz, does not specify the response of a loudspeaker. A graph of the response curve really needs to be known.)

With knowledge of these, we can certainly make an estimate to answer this question.

The spectral content of three different signals is shown in Figure 3.

One is the band- limited pink noise signal used to determine the sensitivity of the loudspeaker. The others are speech and a shaped noise signal having approximately the same spectral content as the speech. This speech-shaped noise is used instead of speech as its RMS level is more consistent as a function of time than actual speech.

Thus, it will be easier to determine the SPL output by the DUT with this signal. All three signals have approximately the same broadband RMS level. From approximately 200-800 Hz the speech-shaped noise signal has greater level than the pink noise signal.

Above and below this frequency region the pink noise signal has much greater level than the speech-shaped noise signal.

Comparing this to the response of the loudspeaker in Figure 1 we see that the loudspeaker has limited output below 150 Hz. The greatest output in the response of the loudspeaker occurs in the 300 Hz-3 kHz region.

If the speech-shaped noise signal were used to drive the loudspeaker with the same broadband level as the noise we could reasonably expect the broadband SPL to be greater than when driven with the pink noise signal.

This is exactly what happens.

The sensitivity of the loudspeaker is 97.1 dB. When driven with the speech-shaped noise the SPL is 98.1 dB, an increase of 1.0 dB.

This results from the higher level of the speech-shaped signal in the frequency region where the loudspeaker has higher output capability compared to the rest of its pass band.

Conversely, if the low-frequency band-limited pink noise shown in Figure 4 were used to drive the loudspeaker it is reasonable to expect that the SPL would be less than when driven by the noise signal.

This results from the low-frequency pink noise signal having a higher level in the frequency region where the loudspeaker has lower output capability.

The SPL produced by the low-frequency pink noise is 94.9 dB, a decrease of 2.2 dB.

Now let’s compare two different loudspeakers. Figure 5 shows loudspeaker A compared to loudspeaker B. Notice that they both have the same sensitivity, 97.1 dB.

Loudspeaker B, however, has greater low frequency and high frequency extension than loudspeaker A.

Because of this the bandwidth of the pink noise used to determine the sensitivity of loudspeaker B is greater than the bandwidth of the noise used for loudspeaker A (Figure 6).

As a result, the mid-band level of the noise for loudspeaker B is slightly less than that of the noise used for loudspeaker A. It’s a bit difficult to see but upon careful observation the black trace can be seen to be an average of 0.5 dB below the red trace from approximately 100 Hz-10 kHz.

This is due to the greater bandwidth of the signal used for loudspeaker B (black trace). Remember that the broadband levels of both these signals are identical.

So what happens when each of these loudspeakers is driven by the broadband pink noise signal (20 Hz-20 kHz) also shown in Figure 6? As each of the loudspeakers used in this example are markedly not flat in their mid-band response there may be some tonal, and potentially measurably, differences in the SPL.

Hopefully, the reader can put these issues aside for the moment. All other things being equal, the loudspeaker with the greater effective frequency range (low- and high-frequency extension) should have greater SPL output.

Loudspeaker B should have slightly greater output when driven by this broadband pink noise signal. In fact, loudspeaker B measured 0.8 dB greater than loudspeaker A, 97.0 dB compared to 96.2 dB.

From these examples one should be able to see that the SPL generated by a loudspeaker is a function of both the loudspeaker’s transfer function and the spectrum of the signal being reproduced.

Several acoustical room modeling programs take this into account when calculating the SPL produced over an intended audience area. They may allow for the selection of pink noise, some sort of speech spectrum, or a user-defined spectrum.

This should aid the sound system designer, while still at the drawing board stage, to better understand the potential SPL capabilities of the sound system with the typical program material the system is likely to be reproducing.

The other item I mentioned at the beginning of this article was referencing sensitivity measurements to one watt being dissipated by the DUT. There are several reasons why I think that this is not beneficial with modern sound systems.

First, it is somewhat cumbersome to determine how much voltage is required across a particular DUT such that the input current drawn from the driving source yields 1 watt. This can be done using dual channel FFT measurement systems and an appropriate current monitor or probe.

But would this give us useful information for the design and/or specification of loudspeakers or sound systems?

We can simplify this measurement procedure so that we don’t concern ourselves with the dissipation of a real watt by the DUT.

Instead we apply a voltage across the DUT that would dissipate one watt in a pure resistance having the value of the rated impedance of the DUT.

This certainly is easier, but again, does this give us useful information for the design and/or specification of loudspeakers or sound systems? Perhaps.

My thought is that more useful comparative information would be gained by applying the same voltage across the DUT regardless of its impedance.

The majority of amplifiers used in sound systems today are of a constant voltage type. That is to say, their output voltage remains constant independent of the load placed on them. Of course, the load must be within the specified operational limits for a given amplifier.

The salient point is that for a given drive voltage, a lower impedance loudspeaker will have greater SPL output than a higher impedance loudspeaker; all other items being equal.

Shouldn’t this be reflected in the sensitivity specification of the loudspeaker? Why then would one want to use a 2.0 Vrms signal to drive a 4-ohm loudspeaker and a 2.83 Vrms signal to drive an 8-ohm loudspeaker to determine their respective sensitivities?

Think about it this way; let’s connect two virtually identical loudspeakers to an A/B selector switch driven by the same amplifier.

The only difference between these loudspeakers is that one is half the impedance (rated at 4 ohms) than the other (rated at 8 ohms).

When switching between these two loudspeakers the output voltage of the amplifier does not change, however, the current drawn from the amplifier does.

This results in the loudspeaker with the lower rated impedance producing greater SPL.

Measuring and specifying sensitivity with the same voltage, regardless of the impedance of the DUT, would accurately reveal the SPL differences that occur.

From these examples, I hope that it’s clear that the input signal and the magnitude (frequency) response of a loudspeaker will determine the SPL generated, not just the sensitivity rating of the loudspeaker.

It’s much better to have knowledge of the loudspeaker’s response in the form of a graph than a single sensitivity number. The latter may be derived from the former.

Charlie Hughes has worked at Peavey Electronics and Altec Lansing. He currently heads up Excelsior Audio Design & Services; a consultation, design and measurement services company based near Charlotte, NC. Charlie is a member of the AES, ASA, CEA and NSCA. He is an active member of several AES and CEA standards committees.

More articles by Charlie Hughes:
Using All-Pass Filters To Improve Directivity & Magnitude Response
Loudspeaker Measurement: An Overview Of EASERA SysTune
Using Limiters To Enhance LF While Still Keeping Things Under Control

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Friday, January 13, 2012

Properly Cleaning Your Microphones

Advice on cleaning and maintaining microphones to ensure their continued reliability

You’ve finally invested in a high-quality vocal microphone and your voice has never sounded better.

Unfortunately, the keyboard player in your band decides he wants to use your mic during his featured rap. You cringe as he practically eats the microphone.

You can barely watch as he encourages audience members to scream into the mic.

Afterwards he returns your mic, still operational but considerably wetter and unhygienic.

Microphones are subject to an inordinate amount of abuse, especially in live music. Grilles and foam windscreens can become saturated with saliva, clogged with lipstick, and will absorb the smell of cigarette smoke prevalent in most clubs.

Regular cleaning of your microphone will not only improve its performance, but is also good hygiene. This document provides several simple yet effective techniques for cleaning microphones.

Dynamic Microphones
The best way to clean a microphone is to remove the grille. Most vocal microphone grilles simply unscrew, e.g., SM58, BG3.1. If the grille doesn’t slide off easily, gently rock it back and forth while pulling it away from the cartridge. Do not pull sharply or with excessive force, since that could damage the cartridge or separate it from the microphone housing.

Once the grille is removed, it can be thoroughly cleaned without damaging the mic. Since most of the offensive material on the grille comes from the human body, plain water should be a sufficient cleanser. Adding a mild detergent (dishwashing liquid) to the water will act as a mild disinfectant and remove odors absorbed by the foam windscreen.

To remove lipstick and other material stuck in the grille, use a toothbrush with soft bristles. In some models, the foam windscreen can be removed from the grille, but this is usually not necessary since water will not damage the grille. Most Shure microphone grilles have a nickel finish that makes them resistant to rust, and replacing the foam windscreen can also be difficult and time-consuming.

The most important thing to remember is: let the grille dry completely before reattaching it to the microphone! Microphones don’t like water, and although dynamic mics can withstand small amounts of moisture, a soggy foam windscreen will introduce more than is acceptable.

Air drying is the best way to dry the grille, but a hair drier on a low-heat setting can be used. Care must be taken not to get too close to the grille as excessive heat can melt some windscreen material.

Cleaning must be done more carefully for microphones that do not have removable grilles, e.g., SM57, 545.

Using a damp toothbrush, hold the microphone upside down and very gently scrub the grille.

Holding the mic upside down will prevent excess moisture from leaking into the microphone cartridge.

This technique is also useful for cleaning the foam that covers the diaphragm inside an SM58.

Again, keep the mic upside down, and be very gentle.

In live situations with multiple acts, it may be desirable to clean the microphones between acts. Use a diluted solution of mouthwash (Listermint, Scope) with water. Using a toothbrush and holding the microphones upside down, scrub the grille of the microphone.

At the very least, this technique will make the microphones smell more pleasant to the performer. Also make certain the sound system is turned off before the cleaning begins!

Condenser Microphones
Due to the more delicate nature of condenser microphones, never use water or any other liquid for cleaning purposes. Even a small amount of moisture may damage a condenser element.

For microphones with removable grilles like the Beta 87 or BG5.1, the grille and foam windscreen may be washed as described above.

Again, the grille and windscreen must be completely dry before reattaching it to the microphone. To clean a microphone with a permanently attached grille like the SM81 or BG4.1, use a dry, soft bristle toothbrush and gently scrub the grille.

Keep the microphone upside down so that loosened particles fall away from it. Take care not to let stray bristles get caught in the grille. This technique also works well for lavaliers and miniature gooseneck mics.

For condenser microphones that will be subject to harsh conditions, such as vocals and theater applications, it is advisable to use a removable external foam windscreen.

This will protect the microphone from saliva and make-up, and can be removed and cleaned with soap and water after the performance. Remember, never get water near a condenser element!

(Provided by Shure Incorporated.)

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Thursday, January 12, 2012

RE/P Files: Construction Of A Live Echo Chamber

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine comes a wealth of knowledge about echo chambers which first appeared in the July / August 1979 issue.

A live echo chamber can be a considerable asset for any recording studio, that is providing that it is a good one.

That’s the problem — how do you construct a good echo chamber? When someone builds a chamber, they hope it will turn out great and pray it won’t turn out absolutely dreadful and good for nothing but storing echo plates.

The truth is there are a number of complex variables which will make each chamber unique.

These factors which effect the chamber include the type of wall construction and the selection of materials used on the inside surface.

Probably the most important consideration is the cubic volume and physical proportions of the chamber.

This leads to the first question to be asked before a chamber can be built. What space is available?

Space
Most times echo chambers occupy surplus space. The more space that is available to start with, the easier the construction job since the builder won’t have to deal with odd angles or cramped building conditions.

The size of the chamber can generally vary from 1,000 cubic feet to about 2,000 cubic feet of internal volume. 1,500 cubic feet seems to get excellent results within a workable space.

A small chamber won’t get optimum results and the largest chambers are a luxury since they occupy a space large enough to be usable for other purposes.

When determining if enough area is available, it is necessary to remember to allow for figuring the wall thicknesses.

There is also a need to consider a wide enough passageway around the structure for a hammer to be swung. If this isn’t planned for putting on the exterior sheeting is going to be a difficult proposition.

It is also suggested that the space be large enough to accommodate a chamber with a minimum interior dimension of 7 feet. A chamber with a side shorter than this will usually give unsatisfactory results.

The Floor Plan
Once it has been decided that there is enough space, the next step is to design a floor plan (Figure 1). When laying out the sides of the room none of the walls should be parallel or even near to parallel. The ceiling should not be parallel to the floor. This is very important if the room is to have maximum random reflections and a smooth decay. It’s at this stage that a bit of math should be introduced (1).

T = Const X M/α S

where:
T = Reverberation time
V = Total volume of the room in cubic units
S = The total combined surface area of all sides in square units
α = The average absorption coefficient Constant:
.049 if measurements are in feet.
.161 if measurements are metric.

The American National Standards Institute (in S-l. 1-1960) defines reverberation time of a chamber as the time it takes for the mean square sound pressure level to decrease to 60 dB after a steady state signal has ceased (2). Generally this level is referenced to 500 Hz although some information relates the level to 1 kHz.

The following equation for figuring decay time was developed by Wallace Clement Sabin (1900). Since his time there have been a number of alternate equations developed but the original equation continues to be the most popular. This is due partly to the simplicity of the computation and the similarity of the resulting data.

Sabin determined that the reverberation time was related to the volume of a room, its surface area and the total amount of absorption.

As the formula would indicate, the time is directly proportional to the volume and inversely proportional to the surface area of the room. With this being the case, the longest echo will be obtained when the required volume is achieved with a minimum of interior surface area.

Trying to achieve more diffusion in the chamber by building accordion type splays will have the effect of cutting down the delay time because the total surface area will have been increased.

Another point to be remembered when figuring out the length of the sides is that none of the dimensions between any two opposing walls should be the same or a multiple or fraction of any other two opposing walls.

There are a number of ratios which serve as guides when figuring acceptable proportions, including the Golden Section; (5^1/2 + 1): 2 :( 5^1/2 - 1).

This relationship was proposed by the Greeks and divides a line in such a manner so that the smaller dimension is to the greater as the greater is to the whole. There are four or five other ratios which have been proposed and accepted to varying degrees, (3) but the one most often used is Sabin’s 2:3:5 relationship. (Figure 2) (3)

Wall Angles
The angles used for the intersecting corners should not be severely acute. In practice, the simplest way to arrive at the wall angles is to build the sides so that two of the joining sides meet at right angles.

Their opposing walls are constructed similarly, but without permanently nailing down the floor plate. Once these walls have been framed, the entire unit can be angled inward. For the average chamber moving one end of each of these two walls in by a foot or so should be sufficient.

Rigidity And Isolation
Two things which are important in making a good live echo chamber are maximum interior wall rigidity, and the total isolation of the chamber from its surroundings.

The more rigid the wall is, the less energy dissipated when a sound wave hits it, hence the surface is more reflective. The isolation is important since it’s essential that the rooms broadband ambience be very low.

Echo chamber wall systems which achieve these goals can be constructed from a variety of materials. There are a few chambers that have been built with walls, floors and ceilings of poured concrete. Such a design if properly executed will get very good results, but will be expensive to build, very permanent and extremely heavy. It will be there forever, providing you haven’t underestimated the strength of your sub- floor.

The second most popular approach is concrete block walls. They are easier to build, and a bit easier to tear down, but once again you have the weight problem.

The most popular and cheapest type of construction is a wood frame design made of 2” x 4” and 2” x 6"s. It is the easiest to build with hand tools and more importantly is relatively light compared to concrete and block.

Walls
In recording studio construction it is very popular to use 2” x 4” staggered stud construction since the results achieve a better transmission loss than a conventional wall. (Figure 3) A staggered wall uses two rows of studs 8” apart (2” x 4” alternating 16” centers) on 6” top and bottom plates. Every other stud is flush to the opposite edge of the plates. In this way the wall sheeting of the two sides is only connected at the top and bottom plate.

A standard stud wall uses a 2 x 4 plate. The sheet rock covering both sides is connected through each of the common studs. The transmission loss between these two types of construction with insulation is STC 36 dB for the standard wall and STC 49 dB for a staggered wall.4

The staggered wall would be the preferable wall design if transmission loss were the only consideration, however, rigidity is a more important factor in the design of the inside shell, hence standard construction is preferred. (Figure 4)

The reason for this is that staggered construction does not allow enough space for cross bracing. The interior shell should have a cross brace splitting the span of every stud, joist, and beam. There should be no unbraced span longer than eight feet.

The preferred layout is to use a staggered exterior wall, and a standard interior one. These two walls should not be coupled in any way and hopefully will not only be decoupled from each other but from the rest of the building.

Floating Walls and Floors
A very important part of the de-coupling of an echo chamber is isolating it from the building it sits in. What sort of isolation is needed will depend on the specific situation and the funds available.

The drawings shown use Celotex for de-coupling. As can be seen, the outer shell is de-coupled from the floor of the building, while the echo chamber has a completely floated floor.

If the location requires extreme de-coupling then it might be necessary to construct the outer shell on its own completely floated floor. It is also likely that something other than than Celotex will be needed.

Machine rubber is a good alternative to Celotex. It seems to work acceptably when used in a quiet environment but it does seem to compress a great deal and might break down with time. A thickness of or more is necessary.

In an extreme isolation situation involving low frequency vibration more severe measures will be necessary. The worst case may need a floated concrete floor on spring isolators, but once again you have considerable weight and expense.

The best material which has been found for most situations are Fiberglas decoupling blocks. As shown, they are two inch cubes, covered with latex (to keep the moisture out) and are specially designed for floating floors. They can be used for both concrete and timber designs.

When they are used with concrete, a sheet of plywood is laid over them and a border runs around the plywood to create a pouring form. Be sure all cracks in the form are sealed so that none of the concrete will seep and re-couple the floating floor to the structural one.

When the blocks are used with wood construction, they should be set under three or four 2” x 6” headers. The timbers form the base for the echo chambers floor joists. Isolators should be placed about a foot or so apart along the entire length of the headers.

The only problem with the block spacers is the gap it leaves between the floating floor and the structural floor. The solution is to fill the space with Fiberglas and run the sheetrock down to %” away from the floor caulk.

These blocks are available from a number of suppliers. The blocks can also be used to isolate ceilings from walls. For this application the blocks seem to work best if they are split in two.

A chamber built on a concrete ground floor can be further de-coupled by using a concrete saw to cut a slot around the perimeter of the chamber. This helps quite a lot in isolating the rest of the building from the chamber. The newly formed slit can be stuffed with Fiberglas but a hard sealer should be avoided since it will slowly harden and compress and re-couple the slabs.

Insulation, Sheetrocking and Sealing
All the walls should be liberally stuffed with foil backed Fiberglas insulation (3%” #R-11). Note: when working with the glass, be sure to always wear gloves, goggles, a mask, and clothes that won’t allow the glass to touch your skin.

Both sides of each wall are covered with sheet rock. Two layers of y2” sheet rock is specified for the interior wall but two layers of 5/8” will work just as well or better. When two layers are used one should overlap the other so that none of the seams coincide.

All seams should be taped and sealed so the room will be airtight. In addition to standard seam Hydroseal which is a black gummy adhesive is highly recommended.

It is normally used as a roofing sealer and will stick to anything. It should be used liberally at every structural intersection of the sheeting or the stud construction. You end up using gallons of this stuff and practically glue the building together.

The purpose of the Hydroseal is to close every crack or seam. The smallest crack should not be ignored. If you have a 1/32” crack along the floor, the actual total area of that hole is considerable.

A final note on Hydroseal. It is suggested that trowels are unnecessary for the application. Wood shims made from construction debris works a lot better. These are good for only about one or two applications. It is almost impossible to clean Hydroseal off anything, including the applicator.

The Chamber Surface and Its Application
After the walls have been built and the last inside layer of sheet rock has been taped and sealed, the chamber is ready to have its reflective surface treatment applied. The reflective walls of the echo chamber can be made from a number of different materials.

The key to how good a particular material is for this application depends on how rigid it becomes after installation and what its absorption coefficient is. Referring back to the earlier formula, the length of the echo is inversely proportional to this coefficient.

The absorption coefficient of any material is defined as “the ratio of sound energy absorbed by a given material to that which arrives at the surface from the source.” (5)

Hence, a porous surface will have a much higher number than a reflective one. This figure will not only be different for each material but vary widely depending on the frequency. Figure 5 shows an absortion coefficient chart of materials versus frequency. As can be seen, plaster has a remarkably low number and is relatively flat. The plaster which Scott uses is Keen Cement (absorption coefficient of .015). He says, “when it is properly applied, it is as smooth as a baby’s bottom.”

Before plastering the surface, the sheet rock has to be prepared so that it will hold the plaster. This is done by nailing on the actual buttonboard.

The actual plastering involves the application of two different layers. The bottom one is a brown coat layer and is put on as a preparation for the top coat of Keen cement. It is strongly suggested that the actual plastering be done by a very good professional.

The mixing of the two cements is nothing more than following the directions on their respective bags. One of the most important things to remember about the plastering process is the amount of time required for proper curing.

As often happens the chamber is complete after the final coat of plaster is applied and there is a desire to use it right away. If the chamber doors are closed prematurely, ventilation will stop as will the drying of the walls. How long it takes for a wall to properly dry changes with the temperature, ventilation, and humidity of the environment, and can vary from a few days to a few weeks.

The longer you can continue to ventilate a new chamber, the better. Scott suggests hooking up a dehumidifier in the room during the drying. He added, “you’ll be surprised how much water comes out of those walls.” This is a crucial part of getting the best end results.

“I have been in chambers that were built years ago that have walls that have never completely cured.” Needless to say such a room has a poor reverberation.

Some chambers sound bad only because the plaster coat was improperly applied. In such cases all that might be necessary is a re- plastering. It is also likely that an additional layer of buttonboard will be necessary so that the new plaster will have a firm base.

Connections And Doors
The chamber should have an IN and OUT opening so that the speaker and microphone lines can be kept separate. The pipe used should be flexible and have an I.D. of 3/4”.

Increase the diameter if there is a need for more than two or three lines per pipe. The smaller it is, however, the less possibility there is for any leakage. Plastic hose works well because it is flexible, has gentle curves and there are no ridges inside for wires to hang-up on.

Make the hole through which the hose is run as small as possible since you want to maintain the integrity of the wall system as much as possible. Once the hose has been installed, any gap between the hose and the wall should be completely caulked with Hydroseal.

A door is obviously needed somewhere in the chamber. Be sure the access to the door and the passageway leading to it is large enough to get a good sized speaker in and out of the chamber.

The two doors used should be of solid construction, not hollow. They are mounted on completely separate frames and jambs to coincide with the de-coupled inner and outer walls. (Figure 6) Machine rubber should be used between the two frames where they almost touch (%” to between the walls.

Rubber should also be used completely around the jambs of the two doors. When they close the rubber should compress and make a tight seal. Since these doors are generally used infrequently, an elaborate closing is unnecessary.

Additional Construction Notes
A light should be installed somewhere inside the room. The switch for it can go anywhere convenient including right on the fixture. Be sure that de-coupling practices are maintained while running the AC conduit and mounting the fixture.

All the walls are built with studs on 16” centers. When laying out the studs, be sure to take into consideration that the drywall is 8’ or 12’ high and 4’ wide and the centers of every third stud needs to line up with the edges of the drywall. #16 nails are used for all end nailing and #8 for toe nailing.

Drywall nails are used on the first layer of drywall, but because of the added thickness #8 nails are used on the second layer. The buttonboard should be nailed on with #8 nails.

Be sure to estimate lumber lengths thoughtfully, to limit the amount of scrap.

However what is left should be used for blocking. The greater the waste, the more the chamber (and for that matter any type of construction) will cost.

Speaker(s) and Microphone(s)
Changing the speakers and microphones or altering their placement will change the sound heard in the control room. Deciding what type of speaker sounds best or what microphone should be used becomes a matter of taste.

This is equally true as to where they are placed in the room.

Providing there was adequate space to start with and care was taken with the design, construction, and isolation considerations, it is probable that the chamber will end up sounding very good.

With a little bit of luck, it might turn out to be the sort of chamber that gains a reputation for itself as well as the studio that has it.

References:
1 - Sound Recording Practices, J. Borwick; “The Acoustics,” by Alex Burd, Oxford Press, page 27.
2 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 25.
3 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 87.
4 - Handbook of Multi-Channel Recording, F. Alton Everest, Tab Publications, page 261.
5 - The Audio Encyclopedia, H. Tremaine, Sams Publications, page 44.

Scott Putnam has been extensively involved in studio construction, having designed and built a number of echo chambers including two for Kaye- Smith in Seattle, and a pair for the Record Plant. As a builder he has collaborated on various projects with a number of acoustic consultants and architects including Jack Edwards, George Augspurger, and Scott’s Father, Bill Putnam.

Downloadable Media
Original Article (pdf)

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

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Tuesday, January 10, 2012

Backstage Class: Developing The Sound Of A Rock Show

My sonic vision should be well in line with the way the artist wishes to be presented

Beauty in art revolves around the realization that there is no “correct” way for something to look, sound or feel. I believe this to be also true about the way audio is presented at a rock show.

In fact, there’s a fairly wide range of possible sonic footprints which a sound engineer can offer the music to the audience while still maintaining an impressive auditory presentation.

An even bigger challenge is to find a “sound and mix” that optimally compliments the artist’s vision and management’s expectations while fueling audience immersion.

So let’s take a look at some of the various factors in play.

First we have the way the artist wants to sound. Awkwardly, the humans that create and play the music rarely get to hear the way their own show actually sounds, so they must rely upon the opinions and reactions of other people.

I smile when chatting with a band after the show and they ask me “how did the show sound?” when ultimately it is them who should be telling me whether my mixing skills and choices rocked or not.

The driving force behind the confidence that stage performers gain in their sound engineer’s skills tends to be based heavily on the opinions of band management, spouses and close friends. Concert reviewers, fan club message boards and real-time audience reactions are also very important aspects of the equation. To reach a level of harmonious success as an engineer, it’s important to also be aware of your own personal preferences, biases and opinions.

I’ve developed a bit of a strategy to balance out the sometimes conflicting pressures in order to end up with a mix that is a solid fit. Though I often do not have the luxury of following the complete process, I’ll share the steps here.

Defining Roles
Meeting the band for the first time is like any personal or business relationship: first impressions are crucial. If possible, I’ve already listened to some of their recordings and asked whoever hired me some basic questions. Early on I really want to determine their expectations. Am I helping a young band get their sound dialed in? Am I temporarily filling in for another engineer? What were the issues and assets of my predecessor? Did he/she leave, get fired, or is it just a logistical choice to use an engineer in this geographic region?

It’s pretty much a fact that every band wants to sound as good as they can - but - are they willing to spend some money to hire in high-quality gear to help achieve this? Or perhaps they want me to squeeze better sound out of whatever gear I happen to encounter?

Persuading artists and management to approve an adequate sound budget can be extremely frustrating. One of the methods I use in order to surround myself with the gear I desire is to say, “if you give me the tools I need to do my job, I will make every show sound great.” This is a very powerful statement because it establishes a self confidence in skill.

Further, it institutes a level of accountability and value in that the expenditure will achieve results. If they do provide the gear you ask for, then you must perform, and they get what they truly desire: a great sounding show. Additionally, the more money they spend on the gear you request, the higher their expectations in results will be.

Focus On Playing
Returning now to meeting the band: “Ooh, that’s cool, how long have you played through that amp?” “What did you play through on the most recent album?” “Is there new gear in your setup?”

By asking them questions along these lines, I want to determine how set they are in their stage sounds. Are they happy and comfortable, or flexible and searching for some new solutions?

I don’t like to change the sound of a band on stage, what I want to do is stabilize it. I want to help them create an acoustic environment that works well for them so they can focus on playing the show instead of messing with the gear.

The next adventure is hanging out at some rehearsals. For me, this is the most important interaction. My mode is watching, listening, and wandering. I will stand near each of the players and hear what they hear when playing. For example, I’m more interested in the tone of the guitar amp where the guitar player stands than what is coming from the amp.

And, how similar are the instrument sounds to the recorded material? I make mental notes of any discrepancies and address them later with the artist. Do you prefer the sound on the album or the rehearsal sound? What about the vocal effects? Some album effects are nearly impossible to do live. How much focus should I put on emulating the backwards guitar solo?

Minimize The Changes
Also in evaluating rehearsals, I start building a mental picture of how I think the show should sound. Factors that are taken into account include: Is there a single person that is the driving force behind the band, or is it balanced between two or more members? Which instrument will reproduce the lowest frequencies? Will the kick sit tonally below or above the bass? Will vocal sibilance create breathy high frequencies above the cymbals?

In addition, there are many ways to overlay two guitars. There is the “wrap around” with one mid-range guitar and the other guitar with lower and higher totality and the mids scooped out a bit. There is the “high low” with a heavy chunky guitar and an edgy bright guitar that sort of combine to form a whole guitar sound. And then there is the “overlap’” with both sounding similar and relying on stereo panning and width to offer spatial differentiation. These also can be combined and altered based on the song or part of a song.

A big goal is trying to minimize the changes I actively need to make during a show so that the primary focus is on distilling several “sonic scenes” that suit particular songs or song tempos. Slow songs work well with extended low frequencies, crisper highs, and longer reverb times. Fast songs light up with a tighter kick and bass, as well as more snare bottom.

About Those Levels
I also pay very close attention to volume levels. Experience has taught me that when a band has a well-balanced stage volume, it makes everything else easy.

By well-balanced. I mean that when I stand center-stage and all of the stage monitors are off, I should hear a well-balanced mix of all amplified instruments meshing well with the acoustic drum sounds.

If things are amiss, I open a discussion about refining stage sound, ideally with each player individually. Since there may be past resentments between band members over volume levels, the last thing I want is to be seen as taking sides.

I stay away from suggesting changes in the volume levels of the amps; instead I discuss physical placement distances and tilting upward, inward or outward of the speaker cabinets. Another thing I avoid is directly broaching the subject of turning down amp volume unless I know the artists well and a strong trust has been developed, and further, that there is no doubt that a distinct improvement will be realized.

Quite often, I’ve found that once the artists realize there is a truly functional and logical stage volume to strive for, they will adjust amp volumes on their own. I also try and get each band member and backline tech to stand stage center at some point and listen.

Speaking of backline techs, I can’t count the number of times that a musician would gladly play at a lower volume, yet the tech, in an effort to please, finds turning the rig up as loud as possible to be irresistible. It’s not uncommon for the amp sounds at rehearsal to be quite good and then at the actual show, everything gets turned up and the sound falls apart.

If all goes as planned, working with the musicians and techs will result in dialing up a desirable stage volume. Whether it’s during rehearsals, sound check or maybe directly after a particularly good show, as soon we reach that happy balance, I take photos of all the rigs and the drum set. They provide a great starting point or somewhere to return to.

Second Nature
If permitted, I will also grab a recording of some rehearsals as well. From this point forward it’s all about complete immersion into the band’s music. in my car, at home, and in my headphones while traveling.

My goal is to commit the music to sub-conscience memory. I want it to be second nature, where my hand automatically moves to push a guitar solo. I also start figuring out which songs have backing vocals, and/or unique effects, and whether I hear any other instruments beyond what I’m aware of on stage.

Notes are jotted in my phone, ready to be asked the next time I see the band. Hopefully, whether this process is a day or two months in duration, by show day, I have a strong mental image of exactly where I want to go with the sound.

My sonic vision should be well in line with the way the artist wishes to be presented. The amount of time I’ve spent with them, in addition to demonstrating a high degree of attention to detail, will ideally establish a confidence in my skills so they can focus on purely playing the show, while I can focus on connecting the music created with the audience that desires to experience it.

Dave Rat (www.daverat.com) heads up Rat Sound Systems Inc., based in Southern California, and has also been a mix engineer for more than 25 years.

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Bringing Clarity To Loudspeaker Power Ratings & Their Relationship To Performance

It’s unfortunate that misunderstandings about power ratings have precipitated an arms race to provide large numbers

One of the most confusing subjects in audio? Loudspeaker power ratings.

It’s generally accepted that a large loudspeaker power rating is a sign of quality and something to be desired.

And it’s the performance metric that probably has the greatest influence on the consumer’s buying decision.

But a closer look reveals that power rating is far less significant than other metrics regarding the performance of the loudspeaker.

The term “power rating” requires further explanation to avoid misunderstanding. It’s tempting to associate it with the acoustic output of the transducer, or even the recommended amplifier size.

But it has little to do with either.

First, let’s expand the term to make it more meaningful.

How about “maximum input power dissipation?” The term “input power” is appropriate because the loudspeaker presents a load to an amplifier.

Assuming negligible effects from the cable (a safe assumption if the correct wire selection criteria are used), the output power of the amplifier becomes the input power to the loudspeaker.

And because bigger amplifier power ratings are accepted as better (i.e., a sport utility vehicle versus an economy car), it’s assumed that larger loudspeaker power ratings indicate a better product.

Amplifiers that connect directly to loudspeakers are called power amplifiers, because their output is a higher voltage and current facsimile of the input voltage to the amplifier. (Figure 1)

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Power amplifiers are rated for power generation. A bigger number is generally better as it indicates the potential for the amplifier to do more work.

Loudspeakers are rated for power dissipation. Their power rating describes the amount of continuous power that can be dissipated in the form of heat without damage to the loudspeaker.

While at first glance it may appear that more power dissipation is better, this is only true if the method used to achieve it does not compromise the efficiency of the loudspeaker.

Modern power amplifiers act as constant voltage sources to the loudspeaker. This means that the output voltage of the amplifier is essentially independent of the load placed on it by the loudspeaker.

If you drive an amplifier with a signal and measure its output voltage with no load connected to the output terminals, and then connect a loudspeaker to the terminals, there is no significant change in the reading on the voltmeter.

The difference between the no load and loaded case is that with the load present current will flow from the amplifier terminals through the loudspeaker.

Lower load impedances (more loudspeakers in parallel) draw more current from the amplifier, increasing the total power transfer from source to load. (Figure 2)

This is why the total output power of the amplifier generally increases when driving more loudspeakers. Note that the output power of the amplifier increases, but the power is distributed among the connected loudspeakers.

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So, if one loudspeaker is connected in parallel with another, the total power output of the amplifier increases but the power per loudspeaker does not.

In fact, it probably drops a little. It is best to keep amplifier loads above 4 ohms to minimize cable effects and avoid excess current demands on the amplifier.

The Arms Race
The power drawn by the loudspeaker from the amplifier is found by multiplying the voltage times the current.

Conservation of energy says that all of the power from an amplifier must be accounted for. Part of the power produces the mechanical movement of the loudspeaker, and the rest of it becomes heat.

The mechanical movement of the cone produces the sound from the loudspeaker. The heat is a waste by-product, and like any waste quantity, it must be disposed of.

Unfortunately the conversion of electrical power to acoustical power is an inefficient process (less than 10 percent is typical) so most of the amplifier power is wasted (heat) and must be dissipated.

The power rating of the loudspeaker describes the capacity of the loudspeaker to dispose of the heat produced by the inefficiencies of the conversion process – so back to our expanded definition of “maximum input power dissipation.”

As such, it’s a mistake to associate the power rating of the loudspeaker with its sonic performance. Higher power dissipation ratings simply mean that the loudspeaker is better at cooling itself.

But power ratings by themselves give no indication of efficiency in producing acoustic power, which is the purpose of the loudspeaker.

It’s possible to increase the power dissipation rating of the loudspeaker by reducing its efficiency. One could simply add some resistive elements internally.

The result is a very large power rating but very little sound – not what we’re after!

The sound pressure level (SPL) produced by a loudspeaker is more closely related to the applied voltage than the applied power.

This can be seen by plotting the on-axis SPL against both. (Figure 3)

The power drawn by the transducer varies with frequency, and while the SPL is often referenced to the input power, it actually tracks the input voltage quite closely.

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It’s desirable for the loudspeaker to have a flat voltage response, so that equal drive voltage per frequency produces a flat magnitude response on-axis.

The ideal loudspeaker could produce the desired sound pressure level using as little power as possible. There would be less heating due to the higher efficiency.

So there is nothing impressive or inherently beneficial to driving lots of power into a loudspeaker.

It’s more impressive to get lots of sound with less applied power. Think of mileage ratings for automobiles, and you have the right idea. It’s more about efficiency than consumption.

Horn loading and boundary placements are methods of increasing loudspeaker efficiency, allowing more sound per applied electrical watt.

Proper Perspective
The same misconceptions about power ratings in loudspeakers occur when we choose a light bulb. The wattage rating is often associated with the light output – more watts, more light.

Bulbs have a luminosity rating that describes their light output, but few consumers ever consider it.

So, if we need more light in a room we buy a “bigger” bulb (higher wattage rating).

It’s only natural to apply this assumption to loudspeakers. Next time, shop for the highest lumens output for a given power input and you’ll get the best value.

A very high power rating on a loudspeaker doesn’t mean that it will be very loud.

Rather than saying “Wow, the Killsound 5K loudspeaker handles 5,000 watts!” it would be better to ask “Why do I have to feed 5,000 watts to the Killsound 5K to get 100 dB SPL in the audience? On the hand, the Efficienator 1 loudspeaker can produce that level and only have to dissipate 100 watts!”

A more meaningful loudspeaker rating would be that of maximum SPL. This rating can be found by scaling the loudspeaker’s sensitivity rating by the maximum input power rating.

It allows a loudspeaker with a lower power rating – but higher sensitivity – to compare favorably with a loudspeaker with a higher power rating but lower sensitivity.

It’s unfortunate that misunderstandings about power ratings have precipitated an arms race among manufacturers to provide large numbers.

Big power ratings are an easy sell, but high efficiency is a better goal.

Power Test
Many methods exist for determining the maximum input power to a loudspeaker. All of them have their merits, and all have similar attributes.

A meaningful power test must include:
- A broadband noise stimulus that is band-limited for the device-under-test.
- A method of determining the power transfer between the amplifier and the device-under-test.
- A time metric that describes how long the loudspeaker can dissipate the applied power.
- And (ideally) a measurement of SPL from the loudspeaker.

Figure 4 shows a useful way of plotting the results of the test. The noise stimulus is often pink noise (equal energy per 1/n-octaves).

Some methods use flat pink noise and others use a weighting scale to simulate the spectral content of music.

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The latter type can produce higher power ratings since more of the electrical energy is shifted toward the lower frequency bands where a transducer can usually dissipate more heat due to its heavier construction.

To determine power transfer, both the voltage and the current applied to the device-under-test must be monitored.

It’s not sufficient to calculate the power transfer from the applied RMS voltage and the nominal impedance of the load.

The load impedance will increase when the device-under-test heats up, reducing the power drawn by the load (power compression).

When a loudspeaker is operated near its power dissipation limits it is not unusual to increase the power applied to the load by turning up the amplifier, but with the result of no additional sound pressure level and even a reduction in power transfer.

It’s best to consider power ratings on a decibel (proportional) rating scale. Wattage ratings can be extremely misleading with regard to the performance of a device.

Consider the fact that a loudspeaker with a 500 watt continuous power rating will only be slightly louder than one with a 250 watt rating (+3 dB), assuming that the efficiency of both are the same.

This means that there is little practical difference between the two, even though there is an apparent large difference in their ratings.

Most power tests modify the pink noise stimulus to have a lower crest factor – the peaks in the program material are reduced by a clipping circuit.

The practical reason for clipping the waveform is to allow the amplifier to deliver more power to the load.

The maximum output power for unclipped pink noise is about 1/10th of the amplifier’s sine wave rated power. Clipped pink noise can produce about 1/2 of the amplifier’s sine wave power rating, allowing power testing with reasonable amplifier sizes.

The clipping artifacts do not contribute significantly to the heating of the loudspeaker, but the lower crest factor produces more power (higher RMS voltage) into the load.

A continuous power test feeds 6 dB crest factor pink noise to the loudspeaker for a specified period of time (i.e. two hours). This is a demanding test for the loudspeaker, since there are no breaks in the program material to allow cooling.

Program power ratings attempt to simulate music or speech by reducing the duty cycle of the waveform. If the noise is pulsed, some cooling can occur between bursts and more short term power can be applied prior to failure.

Many manufacturers estimate the program power rating by doubling the continuous power rating (+3 dB or 2x is a reasonable assumption).

The actual recommended amplifier size will be larger than either of these ratings. A reasonable estimate is the continuous power rating +6 dB (4x).

Given these definitions, a complete and meaningful power rating for a loudspeaker might be: Maximum Input Power – 200W/400W/800W (continuous, program, recommended amplifier size)

Apples To Apples
Once can easily see the problem with comparing loudspeaker power ratings.

It takes a lot of research to assure an “apples to apples” comparison, and many specs simply don’t include enough background information to allow this. Feeding a loudspeaker with less than its rated power presents no danger.

In fact, it will have a longer, happier life with less power. I recommend limiting the input power to no more than one-half (-3 dB) of the continuous rating for reliable operation.

In the preceding example, this would mean using an 800-watt amplifier, feeding it typical program material (10 dB to 14 dB crest factor) and driving it just to the brink of clipping as a maximum.

Under these conditions, the amplifier will be producing about 80 watts or less into the loudspeaker, which is safely below the continuous rating.

Since the amplifier has a potentially large output, care must be taken to assure that low crest factor program material is not turned up too loud as it could damage the loudspeaker.

Finally, it’s important to realize when the point of diminishing returns is being reached when turning up the volume on a sound system.

Each 40 percent increase in applied voltage to the loudspeaker produces twice the input power, and a slight (+3 dB) increase in sound level.

Remember that with audio it is the proportional chance that matters. As the volume of the system is increased in 3 dB steps, eventually the limits of heat dissipation are reached and the next 3 dB “breaks the camel’s back.”

A loudspeaker is very near its maximum loudness at one-half of its rated power. There is nothing to be gained by going further and the loudspeaker will likely suffer permanent damage.

Advancements in automotive technology have produced vehicles with greater efficiency and lower operating cost due to reduced waste.

The audio industry should have a similar goal – achieving the desired SPL using less amplifier power.

As efficiencies increase, the need to dissipate lots of power should diminish – as should our fascination with high power ratings.

Pat & Brenda Brown lead Syn-Aud-Con, conducting audio seminars and workshops around the world. Synergetic Audio Concepts (Syn-Aud-Con) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, Syn-Aud-Con is dedicated to teaching the basics of audio and acoustics. For more information, go to http://www.synaudcon.com

More articles by Pat Brown on PSW:
Specification Sheets: What Do The Charts & Graphs Really Mean?
The Vital Impact That Acoustics Can Have On Sound At Your Church
How To Illuminate The Audience With Beautiful, Consistent Audio Coverage
Proper Loudspeaker Placement: How To Avoid Lobes and Nulls
Ten Reasons Why Church Sound Systems Cost More
What Makes A Quality Loudspeaker?

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Monday, January 09, 2012

Understanding Winches Of All Varieties

The devil is in the details when we're talking about motorized winches and lift machines.

Motorized winches. Lift machines. Line shaft winches. Cable drum winches.

Whatever you want to call them, they’re everywhere these days, with more coming.

To ride the wave, however, a lot more understanding has to be brought on board. (Get it? Wave…board?)

Winches that are designed to lift scenery, lighting or other heavy stuff is what we’re talking about.

Please don’t confuse them with machines used for opening and closing curtains on traveler tracks. They are a whole different animal and are not what we are discussing here.

The basic components of a winch are: Electric motor with brake, gearbox (or speed reducer) drum, and control. We’ll take ‘em one at a time.

Electric motors with brakes are simple enough. The motor makes a steel shaft go round and round and the brake stops it.

If your motor does not have a brake then you don’t have a lift machine. (You have either a curtain machine or a boat anchor.) Most motors in our industry have output shafts that run about 1700 RPM.

The important part here is how much muscle they have. Muscle, in this case, is defined as horsepower. More horses, more power.

Now, the gearbox. This is where things can start to get interesting. The gearbox, or speed reducer, is that box-like thing bolted to the motor. Inside there is a shaft connecting the two.

When the shaft comes into the gearbox it is spinning like a bat out of hell. When it leaves it is walking. Maybe something like 30 RPM. It really depends on the desired end result.

And how does the gearbox do this, you ask? Magic? No.

Gearboxes work on a simple principle: different size gears working in conjunction will not only slow down that pesky shaft, but also develop enough muscle (torque) to turn the drum. Different gear configurations get different results. It depends on what the desired result is.

But all gearboxes have two things in common: They all have gears and they all need lubrication. (There’s a joke in there, but I’m not touching it.)

Manufacturers of gearboxes have specifications for lubrication. Buyer beware, however, for these specs are written for industrial users, not us theatre folk. Their specs call for replacing oil after about a zillion hours of use.

In an industrial application this may be once a month. In the theater it could translate to once every ten years. Inaction, as we all should know, can be just as dangerous as action.

If you don’t use that motor very often the oil begins to turn to sludge at the bottom. Less and less oil gets to the gears when you turn it on. Replace your gearbox oil at least once every two years. Check it every year. If it looks dirty, or you see stuff swimming around in it, change it.

Okay, now we move on to drums. Unfortunately, cable drums are not nearly as exciting. (Buddy Rich, now he was exciting.) Cable drums, when lifting heavy stuff over people, (especially when it’s ME down there) must have some common characteristics.

First, they must be grooved. This way the cable will wrap on the drum in a dignified and controlled manner every time. No overwrapping or jumping around.

Second, the drum must be long enough to accept the entire travel distance of the cable on a single layer, including some extra “dead” wraps for safety.

Series winches are designed for continuous duty pulling and their compact design makes it difficult to get cable caught between the drum flange and end support housing.

For example, if you are lifting a piece 20 feet into the air with 3/8-inch cable, the drum must be at least 4 inches long. 20-feet x 3/8-inch cable plus three dead wraps on a 12-inch drum. (At least that’s what Peter Scheu said, and I always believe Peter.)

Third, the drums have to be connected to the gear box. The only right way to attach a drum to a gearbox on a standard lift machine is via direct coupling. (What an image!)

No belt drives, no chain drives, and no flexible couplers - the fewer the parts, the less potential for failure.

Reprinted with thanks to Sapsis Rigging

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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

{extended}

Tuesday, January 03, 2012

Inexpensive Studio Monitoring Upgrades

While the studio monitors themselves are probably the single greatest cause of this problem, there are a number of other related factors that can have a significant impact. We’ll address three of them here.

“I recently purchased a really good (expensive) set of studio monitors for my studio, but I still find that my mixes sound drastically different as I listen to them on other systems. What am I missing?”

While the studio monitors themselves are probably the single greatest cause of this problem, there are a number of other related factors that can have a significant impact. We’ll address three of them here.

1) Acoustic Treatment: It’s safe to assume that the decisions you make while tracking, editing and mixing are based on what you hear. If you’re mixing in a room that has detrimental reflections like standing waves, flutter echoes and low frequency room modes (and nearly EVERY rectangular room does!), then it’s nearly impossible to hear the mix without being influenced by these common sound issues.

In other words, you might over or under compensate low frequencies, or the placement of lead vocals in a mix - and that’s just the start! The good news is that by implementing the proper acoustical treatments, you can make even the worst sounding room good enough to be useful. Controlling reflections yields truer sound and allows the “real” sound of an instrument or voice to come through.

Beyond the physical construction of the room, which at this point we’ll assume you don’t have a lot of control over, the two methods of controlling sound are sound absorption and sound diffusion (we shouldn’t forget about bass traps, but those technically fall into the category of absorption). Companies like Auralex and RPG have world-class acoustic treatment solutions that are both affordable and amazingly effective.

2) Decouple Your Monitors: Everything in your studio that vibrates contributes artifacts to the sound of your mix. Typically our monitors are acoustically coupled to a shelf, a rack, stand, or the meter bridge of our mixing boards. Doing this automatically degrades the accuracy of what you’re hearing. And once your monitors cause whatever they’re resting on to vibrate, everything else that’s in physical contact with your monitors also starts to vibrate, which just adds to the problem. Auralex makes a nifty and inexpensive product to address this called the MoPAD.

MoPADs provide sonic isolation between your monitors and whatever your monitors are resting on - instantly improving the accuracy of your entire monitoring system. Using the MoPADs allow you to hear the sound of the monitors more directly, without the interference caused by the way they interact with their resting surface.

3) Listening Position: At the listening position, constructive and destructive interference between your monitor’s direct sound and reflections from adjacent boundaries causes severe peaks and dips in the frequency response.

In addition, acoustic resonances or room modes cause substantial acoustical gain or attenuation at frequencies determined by the room’s dimensions. While it is important to provide uniform modal frequency distribution by acoustically treating the room, the degree of acoustic gain at each frequency depends solely on the location of the listener and loudspeakers with respect to the room’s sound pressure distribution at that frequency.

In other words, the placement of your monitors and the listening position has a tremendous effect on your ability to hear your mix accurately. Room Optimizer by RPG is a program that utilizes modern geometrical image model prediction techniques along with powerful multi-dimensional optimization to achieve the smoothest and flattest bass response in a rectangular room - it tells you where to place your monitors relative to the listening position.

This result is accomplished quickly, effectively, and automatically by properly positioning the listener and loudspeakers.

You may find that a different layout of your studio will significantly improve the sound at the listening position, and thus your overall results. With the wide variety of racks, desks, and stands available today it’s easier than ever to reconfigure your studio into not only a better sounding environment, but one that ergonomically allows you to be more productive.

Most people who put studios in their homes or other non-conventional spaces go through at least two or three iterations of their setup before arriving at something they can be happy with for any length of time.

{extended}

Thursday, December 22, 2011

Twenty Questions: Dynamic Details

Test your knowledge of compression in various applications in our latest pop quiz.

Go ahead and see how much you really know or don’t know about Compression.

This mini-test gets progressively more difficult and is a good indicator of where you stand in this area of pro audio science.

Rick Kreifeldt, Harman Professional SDIG Vice President, has graciously guest authored a dynamics processing inspired Twenty Questions.

There is one correct answer for each question. Good luck!

 
1.) What does a Compression Ratio of 4:1 mean?

A.) Once past the threshold, the input signal must increase by 4 dB for the output to increase by one dB.
B.) Once past the threshold, the output signal will be 4 dB less than the input signal.
C.) Once past the threshold, the compressor will have 4 dB of gain reduction.

 
2.) Compressor attack time is defined as:

A.) The time it takes for the compressor to come out of gain reduction.
B.) The time it takes before the compressor starts gain reduction.
C.) The time it takes for the compressor to enter full gain reduction.
D.) None of the above.

 
3.) Signals below the compressor threshold pass through without being compressed:

A.) True
B.) False

 
4.) If a compressor is showing 10 dB of gain reduction and you add 10 dB of makeup gain, lower level signals below the threshold will be:

A.) Reduced by 10 dB.
B.) 10 dB louder.
C.) 20 dB louder.

 
5.) Signals below the threshold of a downward expander are:

A.) Passed through without any gain reduction.
B.) Reduced by the downward expander ratio.
C.) Made louder by the expansion ratio.

6.) A soft knee compressor:

A.) Starts compression before the threshold.
B.) The compressor will not be at the full compression ratio until after the threshold.
C.) Has a lower output level at the threshold than a hard knee compressor.
D.) All of the above.

 
7.) Gate attack time is defined as:

A.) The time it takes for the gate to enter gain reduction.
B.) The time it takes for the gate to close.
C.) The time it takes for the gate to come out of gain reduction.
D.) None of the above..

 
8.) A parametric EQ with a bandwidth of 1/4 octave inserted into the side chain of a compressor with a boost of 6 dB at 1 kHz will cause the compressor to compress a signal at1 kHz more than a signal at 100 Hz.

A.) True
B.) False

 
9.) An optical compressor uses what for the gain element:

A.) Photo cell and LED.
B.) Photo diode and LED.
C.) Fiber optic VCA.
D.) Flux Capacitor.

 
10.) A compressor is set to a 3:1 ratio. The threshold is set at +3dBu. If the input signal is at +12dBu, what is the level of the output signal?

A.) +4 dBu.
B.) +5 dBu.
C.) +6 dBu.
D.) +7 dBu.

 
11.) A gate has a look-ahead time of 600uSec. Which of the following is true:

A.) The gate will open 600uSec faster if this feature is turned on.
B.) The gate looks 600uSec ahead in time to see if it will need to open.
C.) The gate will open in 600uSec plus the attack time.
D.) The main signal will be delayed by 600uSec so that the gate will have time to open before the signal passes through the gain element.

12.) What will be the result of running the kick drum into the key input of the bassist’s gate?

A.) When the kick drum is played, the bassist’s sound will be compressed.
B.) The bass will only be heard when the kick is played.
C.) The bassist will finally be in time.
D.) The drummer will finally be on time.

 
13.) An acoustic guitar player switches between strumming and finger picking in the same song. Which of the following compressor settings will make his finger picking 8 dB louder if his strumming peaks at 8 dBu?

A.) Adjust the threshold to ­8 dBu and the ratio to 2:1.
B.) Adjust the threshold to 0 dBu with a ratio of Infinity:1.
C.) Adjust the threshold and ratio until you have 8 dB on the gain reduction meter when he is strumming and no gain reduction when he is finger picking. Add 8 dB of make-up gain.
D.) Adjust the threshold 0 dBu and the ratio to 8:1; add 8 dB of make-up gain.

 
14.) You are using a digital limiter. The A/D of the digital compressor reaches it’s maximum output when the input signal is at +22dBu, and the box is unity gain.

Assuming no other gain through the unit, at what threshold must you set the digital limiter so that the output signal will not exceed +10dBu?

A.) ­0 dBFS
B.) +10 dBFS
C.) +12 dBFS
D.) ­10 dBFS

 
15.) If a downward expander is set at a ratio of 1:1.5, a threshold of ­40dBu, and a maximum attenuation or depth of 20dB, how much gain reduction will there be if the input signal is at ­50dBu?

A.) 10
B.) 12.5
C.) 15
D.) 20

 
16.) Compression ratio of 2:1. Compressor threshold of ­10dBu. Limiter threshold of 0dBu. If the input signal is at +10dBu, how much gain reduction is being contributed by the limiter:

A.) 10 dB
B.) 5 dB
C.) 20 dB
D.) None
E.) 15 dB

17.) If you want to duck music when someone talks into a mic, you should:

A.) Use a gate with the music going through the gate and the mic plugged into the key input.
B.) Use a mixer to add the two signals, and then use mixed signal into a key input of a gate.
C.) Use a gate set to duck and have the voice in the sidechain, music in the program pathway.
D.) None of the above.

 
18.) A Hi-Shelf EQ is inserted into the side chain of a compressor. The EQ is set to cut 10dB at 5kHz with a slope of 18 dB/octave. The compressor is set to 2:1 at a threshold of -10dBu. If the input signal is +10dBu at a frequency of 10kHz, what is the level of the output signal?

A.) ­5 dBu
B.) 0 dBu
C.) +5 dBu
D.) +10 dBu

 
19.) Which compressor settings will most accentuate the transient attack of a snare drum?

A.) Fast attack and slower release.
B.) Fast release and slower attack.
C.) Fast attack, fast release, long hold time.
C.) 15 dB
D.) Slow attack, slow release, short hold time.

 
20.) Assuming you have a unit which is a multi-band compressor / limiter, which of the following statements is NOT true:

A.) A multi-way crossover is used to divide up the frequency bands.
B.) This unit can be used to de-ess full program material.
C.) For each band you have a separate VCA and RMS detector.
D.) Setting the threshold of each band’s limiter to 0 dBu will ensure that the output will not be above 0 dBu.

Be sure to let us know how you scored in the comments!

More Fun Quizzes on PSW:
Take Our “Real World” Stage Monitoring Quiz
Test Your Knowledge Of Power As It Relates To Sound Systems
Quiz To Rate Your Audio Skills, Knowledge & Personality Type

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Monday, December 19, 2011

What Really Defines Good Bass In Sound Reinforcement?

"What most people think of as boominess in the midbass is really the sound of distortion harmonics from the bass speakers operating a couple of octaves below." -- Edgar Villchur

These days, concert loudspeaker systems can sound very good over most of the frequency range. Things have improved a lot in the last five or ten years.

But to my ears, bass sound hasn’t made nearly as much progress.

How often have do you go to a show, particularly a fairly loud one, and feel that you’re listening to a war between the upper and lower halves of the spectrum?

It all sounds pretty good at low levels, but when things get cooking, up comes the roar from below. 

To preserve some semblance of transparency, the poor sound man has no choice but to produce an offsetting screech from the high mid, and the fight is on. The music (remember the music?)  is lost somewhere in the middle.

At the end of the show, you might hear people making excuses: saying that the room acoustics were bad, or that the band was too loud on stage, or that there wasn’t enough PA, or who knows what else.

Although there are lots of bad rooms and other issues out there, I think that a lot of the problem simply comes from bad bass in the sound system.  In what follows, I’ll describe what makes bad bass, and how NOT to have it.

Harmonics
One of the first people to think hard about bass was a man named Edgar Villchur. Villchur founded the famous home hi-fi loudspeaker company Acoustic Research (AR) in 1954.

AR built the world’s first compact low-distortion woofer, the AR-1, which had a clean, tight, musical bass sound, in contrast to the big boom boxes that were popular at the time.

About the sound, Villchur said something that’s still the biggest single key to good concert bass: “What most people think of as boominess in the midbass is really the sound of distortion harmonics from the bass speakers operating a couple of octaves below.”

Back in 1991, I was developing a new woofer for Jasonaudio’s predecessor, Jason Sound of Vancouver. The new woofer replaced an earlier model that had the same output level and same frequency response. 

The only difference, other than a more convenient box shape, was that the new box had about 6 dB less distortion than the old box. The old box had about 4 percent distortion at high power, while the new one had about 1 percent. 

When we compared the two models, the main difference we noticed wasn’t in the bass. The bass was about the same. 

But with the new box, the entire sound of the PA below 1 kHz was cleaned up.

The midbass and lower midrange were more transparent and less confused-sounding, and the sense of separation between bass and midrange frequencies was greatly increased. 

Why? Because the new woofers were not spraying ugly harmonics across the whole lower half of the musical spectrum.

If you’ve ever hung around with people who tune PA systems using modern measurement tools like Meyer SIM or Rational Acoustics Smaart, you might have noticed (as I have) a curious phenomenon:  when you look at the system’s frequency response after it’s tuned, you often observe a broad valley between about 200 and 400 or 500 Hz. 

Why have they set the tuning like that? 

One of the reasons is that when the system gets loud, the woofers fill up that range with harmonics, so the REAL program has to be reduced to maintain overall musical balance.

That’s a shame, because a lot of very compelling music (piano left hand, cello, saxophone, lower strings of guitar, floor tom…) lives in that range.

Here’s what good bass sounds like:
* Bass impulses do not mask the rest of the music.
* You have a sense that the bass is part of the music, not just some rhythmic sound effect.
* The bass has color and texture that vary from moment to moment—it isn’t all just the same monotonous drone all the time. Unless, of course, that’s the way the music is intended to sound.
* You can hear the pitches of notes, not just vague roars, rumbles, bonks, and thuds.
* Percussive sounds are realistic, and do not sound like giants coughing or rugs being beaten or huge pillows being whacked with sticks.
* Transients are strong and hard, but when a sustained note is called for, it is clearly and musically produced.
* There are no spurious harsh or honky noises added.
* When a really loud, really low note is played, it hits you in the stomach and lower body, not just in the upper chest and head.

Good bass is transparent. This may sound strange - how can bass sound transparent? If you want to hear the answer, go to a pipe organ recital in a fair-sized church or cathedral. No PA system on earth has ever produced the quality of bass that most decent pipe organs put out every Sunday. 

Regardless of what kind of music you like, if you care about bass you should seek out a pipe organ concert every couple of years, just to remind your ears what it can be like.

In PA systems, good bass is rare. In fact, many mix engineers have never heard it. Whey they finally do get a chance to mix on a system that does have it, they often need a little time to get used to it. 

But after they do, they almost never want to go back to bad bass.

Is A Little Bad Better?
Sometimes people say that a little bass distortion is good, because it makes bass sound louder and adds punch to transients. I think the jury’s still out on the value of that approach. 

There’s no doubt that distorted bass does sound louder, heavier, and more oppressive. There’s also no doubt that, if properly mixed, occasional clipping of bass amps can add some crunch to bass transients. 

However, I think those effects are hard to handle and can easily get out of hand.

My vote: If you want more bass, get more (or better) woofers. If you want crunch on a bass line, insert an appropriate effect into the mix channel where you want it, or maybe just use a cheaper mic.

I don’t think it makes sense to subject the whole mix to a bunch of nonlinear, intermodulating woofers just to get a few channels tweaked. Bad might be better if you’re on a hot date, but not if you’re a woofer. 

We’re talking about bass instincts here, not base instincts.

An Exception
I have to mention that there’s one exception to all of the above: trance music. By this I mean dance music, rave music, some kinds of techno music, and generally any kind of music that’s designed to produce an altered state of consciousness by physical bombardment of the body and senses.

If you’re into that kind of thing, then you want your bass to have a stunning effect. Such an effect is produced when each bass transient is accompanied by a blast of harmonics that shakes the room, dims the lights, masks the music, and makes people’s eyes go blurry.

If that’s what you’re after, then you’ll probably want high-distortion bass. The easiest way to get that is to use nonlinear woofers. To get more control over the phenomenon, you can use bass-distorting devices, including subharmonic generators and other evil machines.

Lots of Bass
Even outside the trance community, lots of people like lots of bass. Me, for example—I like lots of bass. But if we’re not using distortion to create the illusion of powerful bass, what then do we do?

The answer is: have lots of bass headroom. If you have plenty of bass output capability, then when a real bass note comes along the system can put it out with integrity. 

In other words, if you’ve really got it, you don’t have to fake it. But how do you get enough bass headroom and still stay within reasonable space and budget limits? 

The answer: use efficient woofers. Most concert woofers are not really efficient enough to provide sufficient headroom for today’s music with today’s budgets.

Frequency Response
How low should your bass go? 

Here are low-frequency limits of some common musical instruments:
Four-string electric bass—40 Hz
Five-string electric bass—30 Hz
Standup (double) bass —40 Hz
Normal grand piano—27.5 Hz
Bass singer —62 Hz
Ordinary large pipe organ— 16 Hz
Kick drum (approximately)— 60 Hz

What this table doesn’t say is where the most important frequencies of these instruments are.

For a lot of bass instruments, the harmonics are louder and more important than the fundamental frequencies. 

Being of higher frequency, the harmonics are more easily and quickly heard by the ear, and often contain the bulk of the melodic and rhythmic information.

What we’ve found for most popular music, the most important part of the musical range is from about 60 Hz up.

The range from 40 to 60 Hz is good for enhancing the experience, but only IF you have the 60-100 Hz range working well, and if you know how to manage the energy in your mix to avoid reverberation problems in big rooms.

Useful musical content below 40 Hz is rare in concert work. It occasionally is needed for acts that feature heavy synthesizer bass. These frequencies don’t work very well with popular music in typical large concert venues.

In their endless quest for impressive specs, loudspeaker manufacturers always seem to be building woofers that put out tons of 35 to 45 Hz at the expense of fidelity and output from 60 to 100 Hz. 

Such woofers move a lot of air, but sound ugly.

The Bass Challenge
In big sound systems, it’s not easy to get good bass. Lowering woofer distortion to an acceptable level is really difficult.

There are two main reasons for this:

1. Air is very thin stuff, and bass vibrations are relatively slow. For a slow-moving bass loudspeaker cone to have much effect on the air, it has to be large and it has to travel back and forth a long way. It may seem as though a high-power woofer is a powerful, rugged, macho device, but in reality it’s a piece of paper frantically flapping in the breeze, trying to make a difference. It’s not easy to make that kind of thing linear.

By way of comparison: under water you can get loud bass down to 20 Hz from a 2-inch diameter woofer.

2. It’s surprising how low a level of distortion is actually required. There are two reasons for this: (a) the ear is 10-20 dB more sensitive to bass distortion harmonics than to the actual fundamental frequency, and (b) many people seem to like a lot of bass boost in PA systems.

My general rule of thumb is that 1 percent distortion in the subbass is equivalent to 7-10 percent distortion in most other parts of the frequency range.

You’ll remember from my old Jason Sound story above that when our woofers changed from 4 percent to 1 percent distortion, the effect was very obvious.

Most modern woofers are rated for maximum output at 5-10 percent distortion. That’s a big part of the problem.

There’s another kind of challenge facing good concert bass: Because many, if not most, touring sound professionals haven’t heard good bass, they aren’t asking manufacturers for it. 

As a result, there are very few low-distortion concert woofer products available.

In studio monitoring and serious home stereo, the situation is somewhat different—many people know what good bass sounds like. In those markets, many manufacturers are serious about low bass distortion, and there are a number of successful products.

For really good concert bass in the 40 to 100 Hz range, I think horn-loaded woofers are the best solution. I’ll talk about horns and other kinds of bass loudspeakers in the upcoming article about woofers.

Getting It Where You Want It
If you do have a system capable of producing good bass, you’re still faced with the task of designing the woofer arrays for best coverage. 

Bass coverage is very predictable, although sometimes the predictions dictate difficult loudspeaker arrangements for production. 

This has lead to a lot of denial in the business, in which sound people pretend to have good bass coverage in order to avoid conflicts with lighting, video, and staging colleagues.

Jeff Berryman served as the director of Jasonaudio, a touring sound company based in Canada, and is a senior scientist with Electro-Voice.

Related Article:
Discussion & Analysis Of A Variety Of Bass Coverage Patterns

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Equalizing the Room - What it Really Means

You can't change the room. You can, however, equalize the response of the loudspeaker system. More than semantics, equalization has very real practical consequences.

“I am going to equalize the room.” We’ve all heard that statement so many times that we scarcely think about what it literally means. We know that in practical terms it means adjusting an equalizer to suit your taste. It may be done with the latest high-technology analysis equipment, voodoo magic or simply tweaking away “until it sounds right.”

In any case, are we really “equalizing the room”? What exactly are we doing? There are lots of disagreements on this topic but all agree on one thing: You cannot change the architecture of the room with an equalizer.

You can, however, equalize the response of the speaker system. Where the room fits into all this is a matter of debate; It is much more than semantics and has very real practical consequences on our approach to sound system alignment.

What do equalizers “equalize” anyway?
Let’s assume that we have a speaker system with a flat (or otherwise desirable) free field frequency response. That is to say, it requires no further equalization. There are three categories of interaction that will cause the frequency response to change, to become, for lack of a better word, “unequalized.”

The first of these interactions are between speakers. When a second speaker is added the combination results in a modified frequency response at virtually all locations. This is true of all speaker models and all array configurations, regardless of any claims to the contrary.

The summation of the two responses varies the frequency response at each position, depending upon the relative time arrival and level between the two speakers. As additional speakers are added the variations in response increase proportionally.

The second category is the interaction of the speaker(s) with the room. These are generally termed coupling, reflections or echoes. The mechanism is similar to the speaker interaction above. The response varies from position to position, depending upon the relative time arrival and level between the direct and reflected sound.

Both of the above effects are the result of a summation in the acoustical space of multiple sources, either speaker and speaker, or speaker and reflection. Therefore the solutions for these interactions are very closely related.

The third interaction is caused by the effects of dynamic conditions of temperature, humidity and changing absorption coefficient. However, the effects of these interactions are small by comparison with the other two, so we will not touch on them further here.

Are any of these problems solvable with an equalizer? The answer is a qualified “Yes”. The magnitude of the above problems can be reduced by equalization, and substantial progress can be made toward restoring the original desirable frequency response.

If equalizers were totally ineffective, then why have we been loading these things into our racks for the last 35 years? However, in a practical sense the equalizer can only provide complete success in equalizing the response when applied in conjunction with other techniques such as architectural modification, precise speaker positioning, delay and level setting.

To what extent is the speaker/room interaction equalizable? This has been a matter of debate for more than 15 years. In particular the advocates of various acoustical measurement systems have come down hard on these issues.

What we are doing is equalizing, among other things, the effects of the room on the speaker system. Why is this controversial? It stems from the historical relationship of equalizers and analyzers. Let’s turn on the way-back machine and take a look.

Early analysis
In ancient times (the 1970s), the alignment of sound systems centered around a crude tool known as the Real-Time Analyzer (RTA) and a companion solution device, the graphic equalizer. The analyzer displayed the amplitude response over frequency in 1/3 octave resolution and the equalizer could be adjusted until an inverse response was created, yielding a flat combined response.

It takes a negligible skill level to learn to fiddle with the graphic EQ knobs until all the LEDs line up on the RTA. It is so simple that a monkey could do it, and the result often sounded like it.

Although these tools were the standard of the day, they have severe limitations, and these very limitations can lead to gross misunderstanding of the interaction of the speakers to each other and the room, resulting in poor alignment choices.

One such limitation is the fact that the RTA lacks information regarding the temporal aspects of the system response. There is no phase information nor any indication as to the arrival order of energy at the mic.

The RTA cannot discern direct from reverberant sound, nor does it indicate whether the response variations are due to loudspeaker interaction alone and loudspeaker/room interaction. Therefore the RTA provides no help in terms of critical speaker positioning, delay setting or architectural acoustics.

Second, the RTA gives no indication as to whether the response at the mic is in any way related to the signal entering the loudspeakers. The RTA gives a status report of the acoustical energy at the microphone, with no frame of reference as to the probable causes of response peaks and dips.

These peaks and dips could be due to early room reflections or speaker interactions, which can respond favorably to equalization. However, the irregularities in response could be from late reflections, noise from a forklift engine or reflections from a steel beam in front of the loudspeaker.

The equalizer will be ineffective as a forklift or steel beam remover, but the RTA will give you no reason to suspect these problems. A system that is completely unintelligible could look the same as one that is clear as a bell.

Third is the fact 1/3-octave frequency resolution is totally insufficient for critical alignment decisions. In addition, there is the misconception that a matched analyzer/filter set system is desired. It is not. The analyzer should be three times the resolution of the filter set in order to be able to provide the visible data needed to detect center frequency, bandwidth and magnitude of the response aberrations.

A 1/3 octave RTA is only able to reliably determine bandwidths of an octave or more. What appears as a 1/3 octave peak may be much narrower. What appears as a broad 2/3 octave peak, may actually be a high narrow peak placed between the 1/3 octave points. What will your graphic equalizer do with this?

Unfortunately the absence of this critical information lulled many users into a sense of complacency predicated on the belief that equalization was the only critical parameter for system alignment. In countless cases, equalizers were employed to correct problems they had no possibility of solving, and could only make worse.

Graphic equalizers have no possibility of creating the inverse of the interactive response of the speakers with the room. Simply put: “You can’t get there from here.”

The audible results of all this tended to create a generally negative view of audio analyzers. Many engineers concluded that their ears, coupled with common sense could provide better results than the blindly followed analyzer.

As a result, though RTAs were often required on riders, they only received cursory attention on show day.

Modern Analysis
Technological progress led to the development and acceptance of two analysis techniques in the early 80s: Time Delay Spectrometry (TDS) and dual-channel FFT analysis. Both of these systems brought to the table whole new capabilities, such as phase response measurement, the ability to identify echoes and high-resolution frequency response.

No longer could an unintelligible pile of junk look the same as the real McCoy on an analyzer. The complexity of these analyzers required a well-trained, highly skilled practitioner in order to realize the true benefits.

Advocates of both systems stressed the need for engineers to utilize all tools in their system, not equalizers alone, to remedy the response anomalies. Delay lines, speaker positioning, crossover optimization and architectural solutions were to be employed whenever possible. And now we had tools capable of identifying the different interactions.

But on the issue of “equalizing the room” a division arose. All parties agreed that speaker/speaker interaction was somewhat equalizable. The critical disagreement was over the extent the loudspeaker/room interaction could be compensated by equalization.

The TDS camp advocated that speaker/room interaction was not at all equalizable and therefore, the measurement system should screen out the speaker/room interaction, leaving only the equalizable portion of the loudspeaker system on the analyzer screen. Then the inverse of the response is applied via the equalizer and that was as far as one should go.

The TDS system was designed to screen out the frequency response effects of reflections from its measurements via a sine frequency sweep and delayed tracking filter mechanism, thereby displaying a simulated anechoic response. The measurements are able to clearly show the speaker/speaker interaction of a cluster and provide useful data for optimization.

Such an approach can be effective in the mid and upper frequency ranges where the frequency resolution can remain high even with fast sweeps but it is less effective at low frequencies. Low frequencies have such long periods that it is impossible to get high-resolution data without taking long time records, thereby allowing the room into the measurement.

For example, to achieve 1/12th octave resolution, the equivalent to the Western Tempered Scale, one must have a time record 12x longer than the period of the frequency in question. For 30 Hz you will need a 360ms (12x30ms). If fast sweeps are made to remove echoes from the measurement, the low frequency data has insufficient resolution to be of practical use.

Dual-channel FFT analyzers utilize varying time record lengths. In the HF range, where the period is short, the time record is short. As the frequency decreases, the time record length increases, creating an approximately constant frequency resolution.

The measurements reveal a constant proportion of direct sound and early reflections, the most critical area in terms of perceived tonal quality of a speaker system.

The most popular FFT systems utilize 1/24th octave resolution, which means that the measurements are confined to the direct sound and the reflections inside a 24 wavelengths time period across the board. This is a good practical level of resolution, allowing us to accurately equalize at around the 1/8 octave level.

With the FFT approach, more and more of the room enters the response as frequency decreases. This is appropriate because at low frequencies the room/speaker interaction is still inside the practical equalizability window.

For example, the arena scoreboard reflection is 150 ms later than the direct signal. At 10 kHz, the peaks and dips from this reflection are spaced 1/1500 of an octave apart. At 30 Hz, they will be only 1/3 octave apart. Thus the scoreboard is in the distant field relative to the tweeters, and applying equalization to counter its effects will be totally impractical.

An architectural solution such as a curtain would be effective. But for the subwoofers, the scoreboard is a near-field boundary and will yield to filters much more practically than the 50 tons of absorptive material required to suppress it acoustically.

Many years ago, the FFT camp boldly stated that the echoes in the room could be suppressed through equalization. Unfortunately, these statements were made in absolute terms without qualifying parameters, leaving the impression that the FFT advocates thought it was desirable or practical to remove all of the effects of reverberation in a space through equalization.

While it can be proven from a theoretical standpoint that the frequency response effects of a single echo can be fully compensated for, that does not mean it is practical or desirable. The suppression can only be accomplished if the relative level of the echo does not equal or exceed that of the direct and that no other special circumstances arise that cause excess delay. (Excess delay causes a “non-minimum phase” aberration and is outside the scope of this article.)

If the direct level and echo level are equal the cancellation dip becomes infinitely deep and the corresponding filter required to equalize it is an infinite peak. As we know from sci-fi movies, bad things happen when positive and negative infinity meet up.

Compensating for the response requires adjustable bandwidth filters capable of creating an inverse to each comb filter peak and dip in the response. As the echo increases, you will need increasing numbers of ever narrowing filters.

A 1ms echo corrected to 20 kHz will require some 40 filters because there are 20 peaks and 20 dips varying in bandwidth from 1 to .025 octave. A 10 ms echo would need 400 with bandwidths down to an 1/400 octave.

Obviously, it would be insane to attempt to remove all of the interaction at even a single point in the hall. In the practical world, we have no intention of attacking every minuscule peak and dip, but instead will go after the biggest repeat offenders. The narrower the filters are, the less practical value they have because the response changes over position.

Practical Implications
It is indeed possible and practical to suppress some of the effects of speaker/room interaction. If this was not possible, it would be standard practice to equalize your rig in the shop, put a steel case around the EQ rack and hit the road. The practical side of this is that we must be realistic about what is attainable and what are the best means of getting there.

The variations in frequency response due to both speaker/speaker interaction and loudspeaker/room interaction will always change with position. Once you have seen high-resolution data at multiple positions, you can never go back to thinking that your equalization will solve problems globally.

A system that has the minimal amount of the above interactions will have the greatest uniformity throughout the listening environment and, therefore, stand to gain the most practical benefit from equalization. If it sounds totally different at every seat, let’s just tweak the mix position and head to catering.

To minimize the speaker/speaker interactions requires directional components, careful placement and precise arraying. In areas where the speakers overlap, time delays and level controls will minimize the damage in the shared area. To minimize loudspeaker/room interaction, the global solutions lie in architectural modification (it’s curtain time), the selection of directionally controlled elements and precise placement.

Finally you are left with equalization. For each subsystem with an equalizer, map out the response in the area by placing a microphone in as many spots as you can and see what the trends are.

In particular, measure around the central coverage area of the speaker. Stay away from areas of high interaction, where the response will vary dramatically every inch.

Examples of this include the seam between two cabinets in an array or very close to a wall. Each position will be unique, but if you place filters on the top four to six repeat offenders you will have effectively neutralized the response in that area.

Conclusion
Modern analyzers are capable of displaying a dizzying array of spectral data. But little practical benefit will come to us if we continue with the antiquated approach of the RTA era. To fully take advantage of the benefits of equalization, we must fully comprehend how to identify the mechanisms that “unequalize” the system.

With modern tools, it becomes possible to analyze the response such that the interactive factors of speaker systems can be distilled and viewed separately. This allows the alignment engineer to prepare the way for successful equalization by using other techniques that reduce interaction and maximize uniformity in the system.

“Equalizing the room” will remain in the domain of architectural acousticians, but with advanced tools and techniques, we can proceed forward to better equalize the speaker system in the room.

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Monday, December 12, 2011

Church Sound: Interfacing Microphones With Sound Systems

A look at phantom power and other output and impedance issues with microphones, in addition to a discussion of best mic cabling, connector and accessory practices

A key area of microphone use is the interface of the microphone with the sound system.

This primarily involves electrical considerations, so here are a few simple rules for proper interface based on the electrical characteristics of the microphone output and the sound system input, and on the requirements for cables and connectors to achieve maximum reliability.

All condenser type microphones require power for their operation. This is provided by an internal battery in some models, or by phantom power in others.

If a condenser is selected, care must be taken to assure that the appropriate power source (battery or phantom) is available. A battery-powered condenser is fine for applications such as portable recording, but phantom power should be used for any permanent micro phone installation.

Phantom power, sometimes called “simplex”, is provided through the microphone cable itself. It is a DC (direct current) voltage that may range from 9 to 48 volts, depending on the microphone requirement and the phantom power source rating.

This voltage is applied equally to the two conductors of a balanced microphone cable, that is pin 2 and pin 3 of an XLR-type connector. The voltage source may be either in the mixer itself or in a separate phantom power supply connected in line with the microphone cable.

Most recent mixers have phantom power built in, and the actual voltage will be stated on the mixer or in the operating manual.

The voltage requirement for a phantom-powered condenser microphone will also generally be stated on the microphone or in the manufacturer’s literature.

Some types, particularly those that are externally charged, may require a full 48-volt supply. Electret types, which have a permanent charge, will typically operate over the entire range from 12 to 48 volts.

Unless specifically stated otherwise by the manufacturer, these microphones will deliver their full performance at any voltage in this range, and further, they will not be damaged by a full 48-volt supply. Supplying less than the recommended voltage to either type may result in lower dynamic range, higher distortion, or increased noise, but this also will not damage the microphone.

Dynamic microphones, of course, do not require phantom power. However, many mixers have only a single switch that supplies phantom power to all microphone inputs, which may include some used by dynamic microphones.

The presence of phantom power has no effect on a balanced, low-impedance dynamic microphone. It is not possible to damage or impair the performance of a balanced microphone correctly hooked up to any standard phantom supply.

If a balanced microphone is incorrectly wired or if an unbalanced, high-impedance microphone is used, there may be a loud “pop” or other noise produced when the microphone is plugged in or switched on. In addition, the sound of the microphone may be distorted or reduced in level.

Even in these cases, the microphone will still not be damaged and will work normally when the wiring is corrected or the phantom power is turned off. If an unbalanced microphone must be used with a phantom-powered input, an isolating transformer should be inserted.

By the same token, it is also not possible to damage any standard phantom power source by improper microphone connection.

Good phantom power practices:
• Check that phantom voltage is sufficient for the selected condenser microphone(s);
• Turn system levels down when connecting or disconnecting phantom-powered microphones, when turning phantom power on or off, or when turning certain phantom-powered microphones on or off;
• Check that microphones and cables are properly wired.

Following these practices will make condenser microphone use almost as simple as that of dynamics.

Not Necessary Or Even Desirable
For the expected sound level, microphone sensitivity should be high enough to give a sufficient signal to the mixer input. In practice, most mixers are capable of handling a very wide range of microphone signal levels.

Occasionally, for extremely high sound levels, an “attenuator” may be necessary to lower the output of the microphone. These are built into some microphones and mixers. Otherwise, accessory attenuators are available that may be inserted in line with the microphone cable.

It has already been mentioned that balanced, low-impedance microphones are recommended for the majority of worship facility sound applications. This will allow the use of long microphone cables, and result in the least pickup of electrical noise.

In any case, the microphone impedance should be similar to the rated impedance of the microphone input of the mixer or other equipment. It is not necessary or even desirable to match impedances precisely. It is only necessary that the actual input impedance be greater than the microphone output impedance.

In fact, the actual impedance of a typical microphone input is normally five to ten times higher than the actual output impedance of the microphone.

The microphone input impedance of most mixers ranges from 1000 ohms to 3000 ohms, which is suitable for microphones of 150 ohms to 600 ohms.

When it is necessary to match a balanced, low-impedance microphone to an unbalanced, high-impedance input, or vice versa, transformers with the appropriate input and output connectors are readily available.

Transformers provide an impedance matching function and can also change the configuration from balanced to unbalanced as needed.

Ideally, transformers should be connected so that the bulk of the cable run is balanced, low-impedance, for maximum allowable length and minimum noise pickup. This would normally place the transformer at the connector of the unbalanced, high-impedance device.

Professional (and most semi-professional) equipment has balanced, low-impedance microphone inputs using 3-pin XLR-type connectors.

Less sophisticated musical instruments, consumer electronic products, computers and many portable recording devices typically have unbalanced, high-impedance microphone inputs using 1/4-inch phone jacks or 1/8-inch mini-phone jacks.

A few mixers offer both types of connectors for each input channel. Simple adapters may be used to mate different types of connectors if no configuration change (high/low impedance or balanced/unbalanced signal) is necessary. Use only high-quality connectors and adapters.

Phantom Power & Bias Voltage
In a condenser microphone, one function of the circuitry is to convert the very high impedance of the condenser element to a lower impedance.

For an electret condenser (the most common type), this is done by a single transistor. Some condenser designs, such as lavalier types or miniature hanging types, have their electronics separate from the microphone element.

In these models, the impedance converting transistor is built in to the microphone element itself. The main part of the circuitry is contained in a separate module or pack usually connected to the element by a thin shielded cable.

The main electronics of such designs operate on phantom power supplied through the microphone cable or by means of a battery in the pack itself.

However, the impedance-converting transistor in the microphone element also requires power in a form known as “bias” voltage. This is a DC voltage, typically between 1.5 and 5 volts. It is carried on a single conductor in the miniature connecting cable, unlike phantom power, which is carried on two conductors in the main microphone cable.

In addition, the audio signal in the miniature cable is unbalanced while the signal in the main cable is balanced.

This distinction between phantom power and bias voltage is important for two reasons. The first concerns the use of wireless transmitters. Body-pack transmitters which operate on 9 volt (or smaller) batteries cannot provide phantom power (12-48 volts DC). This prevents their use with phantom-powered condenser microphones.

However, the body-pack transmitter can provide bias voltage (1.5-5 volts DC). This allows a condenser microphone element with an integrated impedance-converting transistor to be used directly with a body-pack transmitter.

Miniature condenser lavalier types as well as other designs which have separate electronics can be operated with wireless systems in this way.

Picking Up Noise
The second reason concerns the wired installation of condenser microphones with separate electronic assemblies such as miniature hanging microphones for choir, congregation, or other “area” applications.

Since the audio signal in the cable between the microphone element and the electronics is unbalanced, it is more susceptible to pickup of electronic noise. This is particularly true for radio frequency noise because the cable itself can act as an antenna, especially for a nearby AM radio station.

For this reason it is strongly recommended to keep the length of this part of the cable as short as possible, preferably less than 35 feet. It is a much better practice to extend the length of the balanced cable between the electronics assembly and the mixer input.

Optimum microphone performance depends on the associated connectors and cables. In addition to quality connectors of the types described above, it is equally important to use high-quality cables.

Beyond the basic specification of balanced (two conductors plus shield) or unbalanced (one conductor plus shield), there are several other factors that go into the construction of good cables.

The conductors: carry the actual audio signal (and phantom voltage for condensers), usually stranded wire. They should be of sufficient size (gauge) to carry the signal and provide adequate strength and flexibility; use stranded conductors for most applications, solid conductors only for stationary connections.

The shield: protects the conductors from electrical noise, may be braided or spiral wrapped wire, or metal foil. It should provide good electrical coverage and be flexible enough for the intended use: braid or spiral for movable use, foil only for fixed use such as in conduit.

The outer jacket: protects the shield and conductors from physical damage, may be rubber or plastic. It should be flexible, durable, and abrasion resistant. Depending on the location it may need to be chemical or fire resistant.

Different color jackets are available and can be used to identify certain microphone channels or cables.

A large percentage of microphone problems are actually due to defective or improper microphone cables.

Microphone cables should be handled and maintained carefully for long life:
• Position them away from AC (electricity) lines and other sources of electrical interference to prevent hum;
• Allow them to lie flat when in use to avoid snagging;
• Use additional cable(s) if necessary to avoid stretching;
• Do not tie knots in cables;
• Coil loosely and store them when not in use;
• Periodically check cables visually and with a cable tester.

Individual, pre-assembled microphone cables are readily found in a wide variety of styles and quality. In addition, multiple cable assemblies, called “snakes”, are available for carrying many microphone signals from one location to another, such as from the sanctuary to the sound booth.

The use of only high-quality cables and their proper maintenance are absolute necessities in any successful worship facility sound application.

Range Of Accessory Options
Finally, the use of microphones for particular applications may be facilitated by microphone accessories. These are mechanical and electrical hardware items that are often used in mounting and connecting microphones.

Mechanical accessories include various kinds of acoustic devices such as windscreens and directionality modifiers. Windscreens, usually made of special foam or cloth, should be used whenever microphones are used outdoors or subjected to any air currents or rapid motion.

“Pop” filters are employed when the microphone is used close to the mouth, such as on lecterns or for handheld vocals.

These minimize noise caused by explosive consonants such as “p”, “b”, “t”, or “d”.

Although such filters are usually supplied with microphones designed for these applications, additional protection may be needed in some cases. Use only high-quality screens and filters to avoid degrading the sound of the microphone.

There are also directional or “polar” modifiers available for certain microphones that can change the pickup pattern form cardioid to supercardioid, for example, or from omnidirectional to semi-directional in the case of some boundary microphones.

Consult the manufacturer for proper use of these accessories.

Mounting accessories are of great importance in many worship facility sound applications. Stands, booms, and goosenecks should be sturdy enough to support the microphone in the intended location and to accommodate the desired range of motion.

Overhead hardware, to allow microphones to be suspended above a choir, for example, must often include a provision for preventing motion of the microphone due to air currents or temperature effects.

Stand adapters or “clips” may be designed for either permanent attachment or quick-release. “Shock mounts” are used to isolate the microphone from vibrations transmitted through the stand or the mounting surface, such as a lectern.

In addition, there are a variety of signal processors which may be used directly in line with a microphone. These can range from simple low- or high-frequency filters to complete preamp/equalizer/limiter units, though most of these functions are normally provided by the mixer and subsequent elements of the audio chain.

Creative use of these accessories can allow microphones to be placed almost anywhere, with good acoustic results and with acceptable aesthetic appearance.

(Copyright Shure Incorporated, used by permission.)

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