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Tuesday, February 07, 2012

Fishman Unveils Triple Play Wireless Guitar Controller

The new Fishman Triple Play Wireless Guitar Controller combines guitar with any virtual instrument or hardware synthesizer to access a wide range of instruments, samples and sounds on stage to expand the depth and impact of live performances.

Triple Play comes with a wireless controller, hexaphonic pickup, and wireless USB receiver. The controller and included software works with industry standard DAWs and vitual instruments and installs quickly on any electric guitar. The system can be easily removed from the guitar because it doesn’t require any permanent installation.

The Triple Play system features several “hold” functions such as sustain, looping, and arpeggiators, along with string or fret splits for multiple instruments.

Also included are menu navigation controls for the included software and a guitar synthesizer volume control. A guitar, mix, synth switch is easily accessible during performances.

A low profile design (less than .5-inch) allows the controller to be left on the guitar and still fit in the case. It operates with a rechargeable Lithium Ion battery (included).

Triple Play’s powered USB wireless receiver interfaces with computers or iOS devices. The system comes with a comprehensive Windows, OSX and iOS software bundle to get users started.

A Triple Play Wireless Guitar Expander option provides additional connectivity for interfacing wireless MIDI signals to computers or iOS devices. It adds a full function USB audio interface with guitar input, bypass and headphone output, MIDI hardware IN and OUT and support for footswitches to extend Triple Play’s capabilities for recording, performing or composing music.

The new Triple Play Wireless Guitar Controller is scheduled for release in June 2012.

Fishman

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Monday, February 06, 2012

RE/P Files: A Quadraphonic Microphone Development

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature provides an interesting look back at quadraphonic recording. This article dates back to September of 1970. (Volume 1, Number 3). The text is presented unaltered, along with all original graphics.

As a complete oversimplification, a microphone is an instrument which measures differences in air pressure.

It is not surprising that somebody would, in light of the interest in Quadraphonic sound, experiment and perfect an instrument which would measure and transduce the differences in air pressure around a full 360 degrees - to effectively create a quadraphonic microphone.

Such a truly Quadraphonic device, developed by engineer Carl Countryman and producer Brad Miller, is in external appearance no different than the several models of standard microphones (Figure 1).

This Quadraphonic microphone has been designed and built using the case and chassis of a Neumann SM-2, into which four independent microphone heads have been built to provide full 360-degree pick-up.

The pick-up patterns (Figure 2) are cardioid, front and back, and figure-8 at the sides.

Although the obviously complicated matrixing data are proprietary, and unavailable for publication, the discussion of pickup patterns, generally, yields an understanding of how the design provides excellent separation and naturality of sounds.

Cardioid, also sometimes called unidirectional, is a heart-shaped response. It is resultant of an omnidirectional and figure-8 pickup.

The signals are superimposed on each other; at the very rear they are anti-phase, and so cancel out.

At the front they are in phase, hence the tapering hear-shaped response toward the rear.

Figure-8, or bi-directional pickup-patterns, are the result of two directional pickup patterns, one in phase and the other anti-phase.

The output at the front and the back are equal, although opposite,.

As the input signal moves to the side, the output is gradually reduced until at 90 degrees, the two patterns have, for all intent and purpose, canceled each other out.

Figure 3 shows microphone capsules as they are arranged in the microphone head.

“Front to Back” and “Left to Right” are one above the other at 90 degrees to each other.

Three demonstrations, on very spontaneous, served to convince that development of the unit is very nearly complete.

The microphone was hung in Miller’s back yard garden, surrounded by about 200 degrees of sound source emanating from a waterfall with various small tributary streams flowing from it. It presented an excellent opportunity to “hear” the complete environment; the waterfall in stereo on the two speakers in “front,” and from behind, the beautiful ambiance of the total environment and the reflected sound.

Several minutes into the demonstration, on the Southern Pacific tracks bordering on the rear of the Miller garden, a slow-moving freight train ambled by. The completeness of the sound, the way it engulfed the listening room, is difficult to describe. It was totally complete… almost frighteningly so.

Miller completed the demonstration by playing a 4-track tape of his “Mystic Moods Orchestra” on an especially adapted Sony. The machine (Figure 4) has been adapted for 4-track, in and out, and will be able to accommodate 10-inch reels of 2-inch tape.

The machine is the forerunner of a new design from the Countryman/Miller collaboration which will weigh in the vicinity of 20 pounds.

The “Mystic Moods” piece only served to further impress that Quad or Multi is certainly on the way… with an endless spectrum of sound combinations and tonal effects.

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.

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

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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Thursday, February 02, 2012

Stagetec NEXUS Supporting Jimmy Fallon Show Remote Production In Indianapolis

Tied into the upcoming Super Bowl game in Indianapolis, late night television host Jimmy Fallon is broadcasting four shows from the Hilbert Circle Theatre in downtown Indianapolis.

It presents a challenge in recreating the studio setup of the “Late Night with Jimmy Fallon Show,” which is normally broadcast from NBC Studios at 30 Rockefeller Center. 

Nathaniel Hare, front of house mixer for the show, was tapped as the sound designer for the four dates (three live-to-disc and then a live show after the big game). Wireless First, a Clair Global company, was selected due to their experience in both TV and touring sound to supply all of the necessary equipment.

Hare decided on Stagetec NEXUS to provide the backbone of the audio routing, with all audio signals from all stage sources to front of house and monitor— and to the Music Mix Mobile and Game Creek Trucks parked outside—are being transported through the NEXUS system.

The NEXUS network comprises of some 1728 inputs and 1984 outputs. Central to the system is the NEXUS Star, which connects 12 remote base devices throughout the theatre and the OB trucks. 

Further, several consoles connect via MADI, including front of house, music and main production, and the rest have direct connections to the NEXUS via AES/EBU.

“As the show after the Super Bowl is going to be live, I need to be 150 percent sure that there will be no equipment failures,” states Hare, “and the NEXUS not only has complete redundancy built in but it’s track record on reliability speaks for itself.”

Stagetec

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In The Studio: The Attack Of The Lopsided Stereo Monster

Nowadays I’ll either go with a single mic or use two mics in an XY configuration. Why?

 
A reader is puzzled by stereo (2-mic) acoustic guitar recording:

I recently got into mixing acoustic guitar with 2 mics. The problem is that I do not know how to create as much ‘space’ as some tracks I know of. I’ve tried XY, ORTF, and spaced pair.

XY and ORTF are too narrow. Spaced pair seems reasonable (following the 3:1 Rule), but the mic pointed closest to the body becomes overly ‘bassy.’

How can I balance the stereo image? EQ can control the problem but not by much. How would you go about on fixing this problem?

I know mic position has to do with it but I don’t know where to start. Just wondering if you had to overcome this type of problem before.

As much acoustic guitar recording and mixing as I do, I’ve dealt with problems like this a LOT.

(And this applies to ANY instrument, not just acoustic guitar.)

First things first…

Mic Placement Is Everything

I’ve played the “Hey, I’m Just Going to Throw a Couple of Mics in Front of the Guitar and Hope it Works” game.

It’s not a very fun game, trust me. You always end up losing.

Whenever you’re recording an acoustic instrument, always plan to give yourself at least a few minutes to try a few different mic techniques. I

s one mic (mono) appropriate? Does it need the wider sound of a stereo (2-mic) technique? If so, which technique is best?

There are a lot of options, and it would behoove you to try at least a couple of them before committing the recording to tape. (Tape…who says tape anymore?)

I feel like I’ve come full circle when it comes to stereo recording. I used to love a nice, wide acoustic guitar sound. But the last year or so I’ve simplified a lot.

Nowadays I’ll either go with a single mic or use two mics in an XY configuration. Why? Because having a really “wide” recording isn’t all it’s cracked up to be.

Even though XY doesn’t give the widest stereo image, it doesn’t lend itself to phase issues and lopsided recordings.

Speaking of lopsided, let’s talk about that stereo recording that has too much bass in one mic versus the other.

Even if you do your job on the front end with mic placement, sometimes one mic (the one pointed at the sound hole) picks up more bass than the other.

Here’s how I deal with it:

—Place an EQ plug-in on the bass-heavy track ONLY.
—Use the EQ to remove some of the excess low end, until the two tracks are more balanced.
—Bus the two tracks to a stereo aux track.
—Put any additional EQ and compression plugins on the stereo aux track.

The first EQ is simply corrective. It lets you balance out the sound. (No more lopsided guitar.)

The second EQ (on the stereo aux) is what you’ll use to carve the entire sound of the entire stereo recording to fit it in the mix.

As you may have guessed, mic placement and technique play a HUGE roll in how awesome your acoustic guitar recordings (and mixes) are going to sound.

Make that a priority on your next session.

If you’re interested in diving in deeper, I created a 4-week class on getting consistently awesome acoustic guitar recordings. You can join any time here.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

 

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Tuesday, January 31, 2012

Extron Introduces New Eight-Port Fiber Optic Audio Extractor

Extron Electronics has introduced the FOX AEX 108, an eight-port fiber optic audio extractor for independent processing and routing of audio signals in a fiber optic AV distribution system.

Each port accepts signals from a FOX Series transmitter to extract a two-channel analog audio signal for processing, and then re-transmits the original signal to a FOX Series receiver.

To simplify integration with mixers, DSP devices, and audio amplifiers, the FOX AEX 108 provides both balanced and unbalanced stereo.

Buffered loop-throughs feature output reclocking and full transmitter power levels to ensure signal integrity.

Available in multimode and singlemode models, the FOX AEX 108 is great for use in FOX Matrix system applications that require extraction of audio signals for local processing and independent distribution.

“System designers and integrators that rely on Extron FOX Series products for fiber optic AV signal distribution now have a new option to route audio within the equipment room,” says Casey Hall, vice president of sales and marketing for Extron. “The FOX AEX 108 provides an easy way to extract audio signals for independent processing while maintaining the integrity and convenience of the fiber optic link.”

The FOX AEX 108 is part of the larger, expansive FOX Series of fiber optic products from Extron. It is compatible with FOX Series matrix switchers, switchers, distribution amplifiers, and HDMI, DVI, VGA, VGA/YUV, and AV transmitters and receivers.

Housed in a compact 1U, half-rack width metal enclosure, the FOX AEX 108 is designed to provide convenient access for audio signal processing and routing from an equipment room.

Extron Electronics

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The Differences Between Vintage And Current Instruments

 
During a long discussion about vintage instruments in the studio this week, it prompted me to think about this excerpt from The Ultimate Guitar Tone Handbook (written with writer, composer and good buddy Rich Tozzoli) that describes some of the intangible factors that went into manufacturing Gibson humbucking pickups in the 50’s and 60’s.

As you’ll see, there are a lot of external factors that went into making a pickup back then, and those factors can pretty much be applied to all instruments in one way or another.

————————————-

“As if the known factors in building a pickup weren’t enough, consider the many intangible factors as well. For instance, most pickups loose their magnetic strength over time because of environment and electrical interference. Pickups can become weakened or demagnetized completely by leaning your guitar against an amplifier with large transformers, or even from taking your guitar too close to the train motor of a subway (as happened with Andy Summers of The Police).

Another intangible is the fact that tolerances of just about every component were much looser until the 90’s. While the difference was indeed subtle, add enough components at the edge of their tolerances together and you suddenly get a pickup that sounds different even though it’s made the same.

Manufacturing intangibles are a whole other story and for that we’re going to go a bit into the history of the Gibson humbucker.

The Changes In The Humbucker
The first humbucking pickups on the 1957 models of Gibson guitars had a sticker on them saying “Patent Applied For” as the design was in the review cue before being granted a patent (see Figure 3.27). These became known as PAF pickups (“Patent Applied For”) and have become highly sought after today for their great sound.

The problem is that most PAFs sound different from one another due to manufacturing process of the time.

Until 1961 when Gibson standardized the selection process, they randomly used different strength magnets (grade 2 through 5) in their pickups, which accounts for some of the reasons for the different sounds. To make matters worse, sometimes a shorter magnet was selected (mostly seen in gold-plated guitars for some reason), which decreased the power of the magnet as well.

In July of 1961, Gibson consistently began to use all short Alnico 5 magnets, although occasionally a few Alnico 2’s showed up. In 1965, Alnico 5’s became standard in all pickups, which finally brought about a bit of consistency to the process and the sound.

If that weren’t enough, the number of windings on the pickup varied enormously as well, especially in PAFs. The early coil winding machines didn’t have an auto shut-off so the workers would shut off the machine when the bobbin looked full, which was at about 5000 turns. As a result, no two pickups were ever the same.

Even when Gibson bought a winder with an auto-stop, there continued to be problems even though the pickups became more consistent. The stop mechanism was controlled by a fiber wheel which would wear out and break, at which point the workers would approximate the number of winds by timing the wind, which resulted in more inaccuracies.

Since the humbucker is made up of two coils, sometimes the windings of each coil were different even though the total number of turns were correct. This would cause certain mid-range frequencies to stand out and give it more bite.

By mid-1962, the patent for the humbucker was granted and Gibson changed the sticker to read “PATENT NO 2,737,842” which was actually the patent number for Les Paul’s trapeze tailpiece. No one knows for sure if printing the wrong number was merely a mistake or a way to throw off the competition.

From 1963 to 1975, these “Patent number” pickups are very consistent, as are the ones thereafter when new, more precise winding machines were used (see Figure 3.28).

In the 1990’s, Gibson further refined their manufacturing and began to manufacture pickups based on the original PAF design. Thanks to precision modern manufacturing techniques, these pickups are remarkably consistent, which also means that a “magic” pickup made as a result of loose tolerances is no longer possible to get.

That being said, most experts agree that you can now get 90 percent of the way there sound-wise for 10 percent of the cost of a vintage PAF.”

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

 

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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

The Old Soundman: Dealing With Indoor & Outdoor Venue Issues

Think it’s a picnic running sound inside a club? Think it’s nothin’ but a party running sound outside? The OSM has news for you!

Old Soundman,

Yes, Stip!

I occasionally run sound for a band that tends to play local hole-in-the-wall venues.

Okay, we feel sorry for you, now move on!

The “stage” for the band is always in one of 2 places: in a nice boomy corner, or, better yet, right in front of that brick or paneled wall.

These are the times that try men’s souls!

I guess you might be a female, so no offense intended. I don’t know what “Stip” is short for. I am pretty sure that Jacquie (below) is…

One of many problems I run into (including the lead guitarist who insists he hears better with his knees)...

I know that guy! and I think half our readers at home do, too. He must have cloned himself a dozen times in each and every state of the union!

...is cymbal bleed-thru on the vocal mic’s. If I try to spare the audience the shrill ring of these upper frequencies by pulling back the highs on the board, I seem to lose clarity in the vocal.

That is not an illusion, Stip. That is, indeed what is happening, you are perceiving it correctly.

This problem gets worse when the guys are playing at a particularly loud stage volume, and I need to crank a little more vocal, which of course starts to feed back when the ring of the cymbals hit the mic’s, then come thru the monitors and hit the mic’s again…

You know the sad, sad story.

Help!!!

Stip

I do indeed know the sad story. And even sadder is the fact that the list of remedies is a very short one. I’m a straight shooter, Stip.

Move back the drum riser. Can’t. You’re stuck in this little club with a stage the size of a saltine.

Now that you mention it, some cheese and crackers would really hit the spot right about now! Wait a minute, you were saying something about cymbals …

The drummer can be asked to use lighter cymbals with a shorter decay time. But since he is a club guy, getting paid very little beyond the endless chain of longnecks he consumes, he probably only has his local music store’s finest, thickest bang-a-langa models.

Don’t tell me he wears those warm-up things on his wrists? You do have it rough, Stip.

I would be fired if I mentioned a brand name here, but it is kosher for me to tell you that you want a hypercardioid mic for your singer, and he needs to stay right on top of it.

The most radical thing you could do would be to ask the band to buy an infrared gate device to put on the mic, so that when his head moves away, it mutes the mic.

However, this has the undesired effect of really changing your mix, since that is the loudest mic on stage.

When that cymbal noise becomes the evil frosting on the cake of a monitor mix, isn’t that just the worst? You can try to identify as narrow a band as possible to reduce, on the graphics for the affected mixes.

I’m not gonna lie to ya, Stip, everything I have said boils down to band-aids. I am pretty much doctor dan the bandage man here. Stip, it is hellish there where you are. But the bigger gigs are hellish in different ways.

Okay, I’m just trying to cheer you up! on the big stages, it is really fun, sonically, when the drum riser is a mile behind the singer.

Would it make you feel better to hear how Jacquie gets treated? Sure it would!

Just had an outdoor gig. Singer was freaking out, saying “the sound sucks” when in actuality it didn’t suck at all. Tried to tell him (from my limited experience) that running sound outdoors is quite a bit different from running sound indoors.

Since I’m a rank amateur at this, is there anything specific I can tell him to shut him up? He’s a great singer, but like most musicians, he has high end hearing loss.

Thanks mucho. Dig your site. You crack me up.

Jacquie

Thank you, Jacquie! My, what excellent taste you have in humor. I am a much funnier man than others, am I not?

What you are going through reflects the agony of having a limited number of clients. If I read between the lines correctly, you don’t want to just tell this guy to take a hike.

Most of the self-righteous hornblowers over on the live audio board would be real quick to say that you should proudly tell this character off, and then march off into the sunset, with your pride intact, and your wallet quite empty.

Well, I guess some of the more sensible ones who read a lot of self-help books would advise you to talk to the guy when he is calmer (since right after a gig is a notorious time for musicians to make ludicrous remarks, usually due to their lack of confidence in their own abilities.)

In the past, I believe that the lads and lasses of the L.A.B. have recommended gently informing your yodeler that there is no “suck” knob on your console. And, that the way for him to win in life is to express himself as clearly as he can, to the limits of his ability.

He may continue to say “wull, I dunno, Jacquie, it just sucked, y’know?” most of us would shake your hand if you just hauled off and slugged him then. But we live in a very litigious society, so it is best not to.

What you are digging for is him saying something like “there was too much low end” or “it was too trebly.” Precise technical terms like that. Is he criticizing the monitor sound or the house?

Hey, you know what? You sound like you have your head on straight. I think you’re gonna go far, with or without this dullard! You rule, Jacquie!

Luv
The Old Soundman

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

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Tuesday, January 24, 2012

Radial Engineering Announces The New Firefly Tube Direct Box

Radial Engineering has introduced the Firefly tube direct box, a fully discrete class-A unity gain amplifier designed for both studio and live performance.

Radial company president Peter Janis explains: “We have always wanted to launch a tube DI to round out our product range. But over the past two years, we have been sidetracked as we developed the Workhorse and our many 500 series modules. We finally got back on this project and are pleased to bring the Firefly DI to market.”

The Firefly begins with two inputs, each of which features a separate level control to enable the artist to set each instrument with optimal gain.

Switching between inputs can be can be done using the front panel switch or via the optional JR2 remote footswitch. The instrument signal is immediately routed to a tuner output that is always on.

When used with the JR2 footswitch, the Firefly may also be muted remotely for quiet on-stage tuning. Both the footswitch and front panel are equipped with LED indicators for status monitoring.

Following the Radial JDV, the Firefly’s front end circuit is 100 percent discrete class-A and is void of any circuit stabilizing negative feedback. This produces a more open, less constricted sound. 

The Firefly is also equipped with Drag Control load correction that enables the artist to adjust the load on the magnetic pickup for a much more natural rendering. When bypassed, the load jumps to 4 meg-ohms enabling the Firefly to be used with piezo pickups such as common with upright bass and other acoustic instruments.

The exceptional warmth and detail is achieved by combining Radial’s unique front end with an all new12AX7 tube drive circuit. Contrasting the input sensitivity with the output drive enables the artist to fine tune the grit or edge to give the sound more character. 

Firefly comes shipps with two 12AX7 tubes, a select premium tube for audiophile performance and a low-fi version for added growl.

A fully variable high-pass filter enables the engineer to set the bass cut-off frequency for optimal layering. This ‘Nashville trick’ lets you set the cut-off to better match the size of the instrument whereby a lower cut-off would be used on contrabass, slightly higher on acoustic and higher again on fiddle or mandolin. By setting a different cut-off for each instrument one can eliminate resonance while still retaining the character.

Connectivity is extensive: The rear panel begins with two stacked 1/4-inch instrument inputs. A second set of stacked 1/4-inch jacks presents the user with a buffered thru-put that delivers either the original instrument’s tone or the output from the tube circuit.

Below, an insert jack enables one to add in effects in series with the tube drive circuit and apply the effects to the overall sound.

The third set of stacked jacks feature a tuner output and a TRS jack for the JR2 remote control. The Radial transformer coupled XLR output is outfitted with a ground lift switch and a 180-degree polarity reverse. This can be helpful when controlling feedback or interfacing with older vintage gear.

Power is supplied via an exterior switching supply for 100- to 240-volt operation and delivers a variable output that ranges from a typical unity gain DI level to a full +4 dB line level for direct recording.

The Firefly is road ready with 14 gauge steel construction plus a protective zone around the controls and switches. This is augmented with steel cased switches and potentiometers plus a double sided military grade PCB for added life.

The Firefly comes with a carry handle that may be removed should rack mounting be needed for touring using optional rack-mount kits.

Radial Engineering

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Switchcraft Announces New AudioStix Line Of Adapters

Switchcraft has introduced the new AudioStix line of adapters, the newest additions to the company’s expanding line of premium pro audio products.

AudioStix can be used as stand-alone adapters, rear mounted into modular Switchcraft QGPK Series rack panels, or permanently installed into any number of custom racks and wall plates.

Four versions are available now:

—#318 Mini AudioStix is an 1/8-inch mini to XLR adapter with volume control and ground lift
—#319 is an 1/8-inch mini to terminal block adapter with ground lift and pad switch
—#366R is an XLR female to BNC (110 to 75 Ohm) adapter
—#367R is a BNC to XLR male (75 to 110 Ohm) adapter

Note that all AudioStix versions are outfitted with transformer isolation.

AudioStix are made in Chicago, USA.

Switchcraft

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Monday, January 16, 2012

Reamping For Live? A Method Of Improving Electric Guitar Performance

Focusing on making the guitar wireless system sound smoother and more natural

The majority of electric guitar players will tell you that they much prefer the sound of “hard-wired” guitars versus going wireless.

When you “radio” a signal, there is not only a sense of disconnect, but the tone never seems quite right.

I noticed this years ago when testing a guitar splitter. One of our engineers sent me a prototype, and after a few minutes of testing, I called him up and said that it worked well but was not quite right.

He said, “what do you mean? It is class-A, 100 percent discrete, and has Jensen transformers. It’s perfect!” I replied that while it might be technically perfect, there was still something wrong.

Eventually, we figured out that it had to do with how the pickup was loaded, as well as how tube amps differ from solid state inputs. (This problem is not only common with wireless systems but all types of guitar signal buffers.)

Applying The Load
To solve the problem, we added a control that would enable the guitar tech to adjust the load so that the guitar would sound right. For this to work, the load needs to be applied directly onto the pickup.

In other words, if you connect the guitar to a buffer and then try to adjust the load, it will not work. This also means that it has no effect on active pickups.

When using a wireless system, the guitar is connected directly to the wireless transmitter, which then buffers the signal and sends it to the receiver.

Then, that output is either routed to the guitar amp, or a fridge full of pedals, or to the front of the stage so that it can go to the pedalboard and then back to the amp (Figure 1).

Because the wireless system is a buffer, the load must be placed in between the guitar output and the transmitter.

My company makes a device to do this called the Dragster. It’s designed to be attached to the guitar strap and then simply wired in series.

Even though this approach works very well, the last thing a guitarist wants is another widget on his strap.

A number of artists have also been implementing an old recording trick known as Reamping on the live stage.

When doing this in the studio, you basically take a dry track from the recording system, send it out line level to a Reamp device (“Reamper”), which then convert the balanced signal to an unbalanced one that is better suited for a guitar amp.

This enables the studio engineer to capture the performance and worry about getting the “ultimate” guitar tone later.

It works much the same for live. You take the output from the wireless receiver and send it through the Reamper to get the same effect (Figure 2).

By converting the signal, the wireless system sounds smoother and more natural. And when artists are happy, they perform better.

In The Field
Mix engineer Brad Baisley recently talked to me about his work with Reamping and related facets for noted country artist Clint Black, and he provided me with this overview:

“I started formulating my approach after a show where Clint expressed concern that his guitar tone was dull.

“The guitar tech, Kenny Barnwell, and I were also tired of battling noise emanating from the wiring to and from the pedal board. I knew that the 100 feet (each way) run of 1/4-inch cable was primarily to blame for both problems - not the modern RF equipment he was using.

“We added a Radial Headbone amplifier switching device that allows two different guitar amp heads to be used with a single speaker cabinet, and then also decided to try a Radial ProRMP Reamp box as well as an SGI interface to boost the signal.

“We were immediately happy with the result, both in terms of sonic quality and noise level. Clint noted that his guitar sounds much more natural, with smoother, more extended highs and fuller low end. 

“Another bonus is these devices have XLR interconnects. If the 100-foot cable loom we built is ever too short, I can dig into our audio spares and help the techs extend the wiring with no loss. And, locking XLR connectors add a considerable amount of security.

“The output on the Shure UR4D wireless receiver we use is 200 ohms, while pedals and amps are designed to see much higher impedance. The level of the output is also far higher than that of a guitar. This leaves you having to turn down the output on the receiver compromising on gain structure and signal-to-noise ratio. The Reamp solves this elegantly.

“We also often find that some local PA providers run cross-stage feeder on the downstage lip of the stage, so running long lengths of 1/4-inch cable parallel to it is just asking for noise issues. But with an all-balanced signal flow, that is now barely a concern.

“When setting up Reamping, first we make sure the levels on the wireless receiver are correct. The UR4D has a digitally controlled level trim (we have it set it at unity), and a Mic or Line level switch on the back for the XLR output (we set it to Line to send as hot of a a signal possible to the Reamp box). The companion UR1 beltpack transmitter has coarse and fine gain controls, and we also set these both at unity. 

“With the pedalboard connected to the amp through the SGI, we then plug directly into the pedalboard input using a short 1/4-inch cable. We do a sound check of the guitar, using our ears (and maybe an SPL meter or VU meter on the console), evaluating the loudness of the guitar.

“After that, we plug the guitar into the wireless beltpack, and connect the pedalboard to the wireless receiver through the Reamp. The level on the Reamp is turned way down, and we slowly bring it up until it matches the level noted earlier. 

“It’s a good idea to A-B back and forth several times to further dial in the level on the Reamp box. It puts us into a unity gain situation where the wireless (and cabling from it) will have the least effect on the guitar tone. We also have a Radial BigShot ABY bypass switcher on the pedalboard with a 1/4-inch plugged in and hidden under the pedalboard for a backup in case there are any issues with the wireless. It’s easily accessible and can instantly be activated using the switch.

“I’ve recently transitioned to a position doing monitors for Blake Shelton. We’re embarking on an eight-truck, full-production tour in 2012.  During a recent production rehearsal, Blake’s tech and I revamped the instrument wireless setup, moving all of the Sennheiser inbound RF into racks and networking it for easy coordination.

“In order to move the amplifiers off the deck for a clean look, we implemented four sets of Radial SGI and ProRMP (as described above) for acoustic guitar, electric guitars, and bass. So far, so good!”

Peter Janis is president of Radial Engineering, which last year purchased the Reamp brand and patents from inventor John Cuniberti.

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Thursday, January 12, 2012

Build A Thumper: A Simple Way To Test Loudspeaker Polarity

A cheap and easy way to test for proper wiring inside your loudspeakers

 
If you’ve attended a HOW-TO Sound Workshop, you’ve heard me play examples of loudspeakers out of polarity.

You can lose bass response, cancel your vocals and cause general phase mayhem in your sound system.

Well, here’s a cheap and easy way to test for proper wiring inside your loudspeaker cabinets before you install them up on the wall and without running any audio signal through them. See below for Thumper, my trusted polarity signal injector.

Once you build Thumper, you just hook a loudspeaker cable from its phone jack to whatever cabinet you want to test.

A brief push on the momentary push button will inject a 9-volt positive signal into your speaker cones. You should then see all the woofers pop out just a bit.

If one loudspeaker pops in while the others pop out, you have big wiring problems inside the cabinet. If all of the loudspeakers pop in, then the input jack on the cabinet or the cable may be wired in reverse. (You already know how to solder from a previous column, so get fixing.)
 
You can build Thumper from junk parts or go to Radio Shack for a little plastic project box, quarter-inch phone jack, 9-volt battery clip and SPST (Single-Pole/Single-Throw) Normally-Open (N-O) momentary contact switch.

An exact part mounting plan isn’t critical — as long as everything fits inside the case it should work.

I usually put a small piece of foam under the battery to keep it from rattling around or sliding inside the box. But just make four solder connections, and you’re done.

The Thumper works great on loudspeakers both large and small, and the 9-volt output only dumps a 10-watt pulse into your speakers, so it’s safe to use. What’s not to like?

Mike Sokol works with the HOW-TO ASSIST Tour (Academy of Sound System Integration, Setup & Troubleshooting) which provides sound and electrical contractors and sound system installers with the best possible training on how to setup, integrate and troubleshoot live sound systems of any size. Find out more here.

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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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Wednesday, January 11, 2012

House Research Institute Outlines Five Ways To Protect Your Hearing

House Research Institute offers simple ways to preserve hearing during 2012 and beyond.

At the upcoming 2012 NAMM Show, House Research Institute (booth 1292) will be on hand providing hearing screenings throughout the duration of the show, while offering advice on how to protect hearing.

House Research Institute - a leading non-profit dedicated to improving the quality of life for people with hearing loss and related disorders - has outlined five simple ways to preserve hearing during 2012 and beyond: 

1) Know thyself: have your hearing tested
Often, hearing loss issues are initially detected by family and friends rather than the person experiencing it. “When a person frequently has trouble understanding conversations in places where there is significant background noise, such as at parties, crowded restaurants and clubs, it might be a good time for a hearing test and an ear examination,” observes John W. House, MD, president of House Research Institute and physician at the House Clinic.

Find out where you stand so you can understand and address the personal risks you may face — hearing exams take just minutes. Noise induced hearing loss begins in the higher frequencies and does not affect speech frequencies until it is advanced. Therefore, a screening audiogram is advised for those who are exposed to loud noise.

2) Know thy surroundings: avoid potentially dangerous environments
By ensuring you are in a safe listening environment, you mitigate the risk of noise induced hearing loss (NIHL). “If you have to raise your voice to be heard, you are likely in an environment with sound levels exceeding 85 dBA,” says Marilee Potthoff, director of community outreach and education at House Research Institute. Musicians and engineers depend on good hearing for their careers, but also are at high risk for hearing damage from prolonged sound exposure on the job. 

If you’re in the sound industry, it’s important for your hearing health to carefully monitor your sound environments that reach above 85 dBA both on and off the job, and know how much to limit your exposure. When relaxing with your personal stereo or player, we recommend keeping the volume setting at no louder than 60 percent of max. potential.

3) Use it or lose it: make the right choices in hearing protection
Educate yourself on what kind of hearing protection is truly effective. “Select hearing protection devices that provide the appropriate amount of sound reduction. Hearing protection with an NRR (noise reduction rating) of around 25 to 35 dB offers better protection for loud music environments than devices with lower NRRs.  Using devices with a much lower NRR may result in significant damage to the inner ear when exposed to high level [loud] sounds,” says Andrew Vermiglio, AuD, HRI research audiologist and California State University Northridge audiology professor. 

Some custom ear plugs — which are available through licensed audiology clinics, including the House Clinic — offer a flatter attenuation across the frequency range and may make listening to loud music more enjoyable than standard, over-the-counter earplugs, such as foam or pre-molded plugs. Standard earplugs tend to “colorize” what you hear by filtering the high frequencies more than the low frequencies.

4) Keep it clean: ears need good hygiene
Earwax may not be the most popular discussion topic in the world, but it is certainly worth knowing about. Knowing how to safely remove wax and dirt build up will help you keep your hearing on the right track in 2012 and beyond. “Never insert foreign objects into your ear canal, including cotton swabs — instead, use a warm washcloth to gently clean the outer area of your ears or an over the counter ear wax removal solution,” says Dr. House. 

Other ear cleaning methods known as ear candling or coning are dangerous, not effective, and can easily damage your ear canal.

5) Make a date: have your ears checked on a regular basis
Have your hearing checked annually. If you notice a change in the state of your hearing, seek immediate medical attention. “Annual hearing exams may help to identify potential hearing loss issues while there is still time to rectify them,” says Dr. Vermiglio.

Also, symptoms such as hearing loss, ringing in the ears, dizziness or loss of balance, may be related to a serious medical condition.

For more information, visit the House Research Institute website.

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Church Sound: Working Within Volume Limits

You’re better off dealing with the root cause of the problem rather than trying to figure out how to stay below an arbitrary number

 
The topic for this post comes from a reader who wants to know what he should do when faced with the requirement to mix no louder than 85 dB SPL peaks.

That’s right, 85 dB peaks.

Why I Hate Volume Limits
I used to own a video production company. We were often hired to do video based on length.

I always tried to talk the client out of imposing a length limit on a project saying, “The video needs to be as long as it needs to be, then it’s done.” I feel the same way about volume.

Ideally, the worship leader, front of house engineer and church leadership are all on the same page when it comes to volume.

In that ideal world, the music will be mixed as loud as it needs to be to convey the power and energy (or lack thereof) required. The band, the song and the crowd will tell you how loud it needs to be. Go over that and it’s too loud; go under and it’s too soft.

Imposing a arbitrary limit on volume to me seems a bit like telling the pastor his sermon needs to be 3,000 words, no more, no less. But we live in a less than ideal world, and we have to live within arbitrarily defined volume limits. So what’s a sound guy to do?

Investigate
The first thing I would do when faced with a limit like that is find out where the number is coming from.

Is it based in an inaccurate reading of OSHA hearing protection guidelines? If so, educate yourself and have a rational conversation with your pastor. Help him to understand that 8 hours of exposure to 85 dBA SPL in a machine shop 5 days a week is a whole different animal than 85 dBA SPL peaks for 15 minutes of worship music.

If that’s not the case, dig a little deeper and see where the number came from. Did someone wander by the booth one day and see 85 on the meter and think, “That sounds about right?” Are people complaining that it’s “too loud?” Is it really too loud or are there spectral balance issues? Or perhaps the setter of the number doesn’t like electric guitar. Or drums.

Acoustic drums will generate 85 dB peaks with the PA turned off, so you need to figure out where this is coming from.

System tuning and spectral balance are huge issues that can be addressed and give you a to more leeway in mixing at an appropriate level. 85 dBA mixes can still be excruciating, while 100 dBA can be enjoyable if done well.

Live Within Your Means
Or in this case, your leadership. In my current church, I have a different definition of “too loud” than my senior pastor does.

Since his is lower, I have to adapt my mixing style to suit him—he’s the boss after all.

The challenge for me is that his definition changes week to week.

I’ve been told it’s “awesome” one week at 92-94 while it could be “too loud” at 90-91 next week.

So I’ve spent a lot of time working on getting my mixes right, the balance correct and the system tuned to his liking.

I’ve also adapted a different way of metering my loudness. I use a software program called LAMA, which can display both a standard SPL readout (I use A, Slow) and an average (I have chosen 10-seconds).

LAMA allows me to set colors at various levels so I have my average number turn yellow at 85 dBA SPL, and red at 91, which gives me a “corner of the eye” indication as to where I am.

I keep an eye on the standard readout as well, and occasionally my peaks run into the low to mid 90s, but for the last month and a half, if I keep my 10-second average below 90, my pastor is happy.

Personally, I’d be happier if it was louder. But I’m not paid to be happy; I’m paid to make him happy. I often say, “If you can’t abide by the limitations your leadership puts on you, then you need to leave.” Same applies here.

Again, I would talk to my pastor and find out where this is coming from. As him if it would be OK to try mixing to a 85 dB 10-second average and see how that feels. Address the spectral and mix balance issues; you might be surprised.

Beware Compression
The reader asked if he should compress the inputs, and bus compress the mix to give him the power he wants, while staying under the “legal limit.” To me, that’s a little like putting your phone on speaker and holding it in front of you while you drive.

Yes, you could compress the inputs a few dB, then bus compress a few more, then compress the master another a little further, and compress it again in the DSP. That would certainly raise your average SPL while keeping your peak below 85.

However, it’s very likely that this technique will result in the perception that it’s even louder, which may cause your limit to be lowered further.

You could also hard-limit your DSP so you can’t exceed 85; but again, if you suck all the dynamics out of the music, all the life goes with it, and it will also sound louder. This would be self-defeating on two fronts.

At the end of the day, I think you’re better off dealing with the root cause of the problem rather than trying to figure out how to stay below an arbitrary number.

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

 

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