Article

Monday, February 06, 2012

ESS Audio Outfits Stadion Miejski With System Headlined By Harman Components For Euro 2012 Champ

Polish technology company ESS Audio is midway through an intensive development program that will see the completion of several world-class football stadia by the time Poland co-hosts the UEFA Euro 2012 Championship (along with Ukraine) next summer.

Earlier this year, ESS Audio commissioned an integrated Harman Pro audio components in the new PGE Arena in Gdansk, before turning its attention to Stadion Miejski in Wroclaw, the highest Category 4 Municipal Stadium in the country.

The ESS technical team engaged in a similar fast-fit installation for the rebuilding of the 42,000-capacity stadium, again sourcing most of the equipment from the Harman Pro portfolio.

ESS Audio worked within an innovative architectural concept—devised by JSK Architekci and built by German company, Max Boegl—in which the building is covered by glass fiber mesh coated with a Teflon fiber net façade. As with the successful PGE Arena design, the company used EASE predictions for system optimization, dividing the stadium stands into 14 separate zones, and assigning a zone each to the ancillary UEFA, VIP and Incentive boxes.

ESS Audio has equipped the main bowl and stands with JBL PD Series loudspeakers, with rotated horns, specifying a total of 59 PD5200/95-WRX (90 x 50 degrees) and 28 x PD5200/43-WRX (40 x 30 degrees) weatherized speakers, along with 56 PD5125-WRX weatherized subwoofers. These have mainly been mounted in clusters of five, comprising two PD5200/43-WRX, one PD5200/95-WRX and two PD5125-WRX weatherized speakers.

A further three single PD5200/95’s fire onto the field, with two single clusters playing in front of the LED screens, aiming 90 degrees down to the lower seating. Two additional clusters, comprising a pair of PD5200/95-WRX and PD5125-WRX, are set behind the LED screens for mid and high seat coverage.

With the wind factor a major area for consideration, ESS prepared custom brackets for the PD enclosures, and as a further precaution, secured each PD loudspeaker in the cluster with a steel line covered in polymer.

Powering the rig are 47 Crown CTs3000 amplifiers, fitted with PIP USP4 processor modules, and interfaced with a BSS Soundweb London DSP environment—allowing distribution over CobraNet, and for the system to be remotely controlled and monitored via Harman HiQnet System Architect.

The stadium is divided into a number of dedicated zones, including 20 VIP boxes,10 further Incentive Boxes (and UEFA boxes), a Business Club, a general esplanade / concourse, a fan shop, team zones (including indoor swimming pool) and media zone. In the UEFA, Incentive and VIP areas, a further 90 JBL Control Contractor 8128 ceiling speakers have been specified—powered by Crown CTs600 amplification.

Amp racks are stationed in the four corners of the stadium (each containing a BSS Soundweb London BLU-80 DSP), while an additional BLU-800 processor is located in the Skybox. All five processors are equipped with Input/Output cards.

Each rack also contains an automatic amp changeover (in the unlikely event of amplifier failure), an Edimax switch along with several Moxa optical-Ethernet converters/switches, making the system fully redundant. This is easily interfaced with the stadium’s voice evacuation and fire alarm system via Soundweb London.

Up in the Skybox is the technical control room where a Soundcraft GB4-16 console, a pair of JBL LSR2325P studio monitors, a rack with AKG DMS 700 wireless mic systems, various line devices (players and recorders) and Soundcraft redundant console power supply are located.

From the Skybox music is broadcast and live announcements made, with an adjacent room for evacuation procedures, complete with fireman’s switch.

The installation meets all required standards (including an STI of 0.5). ESS provided full training including use of the Skybox equipment, System Architect software and general problem solving. They will be able to log on and carry out regular health checks under strict safety rules.

According to ESS Audio project manager Witold Karalow, this stadium project has been one of the most challenging the company has undertaken.

Aside from Karalow, the ESS Audio project team in Wroclaw comprised system designer Wojciech Zielinski and BSS Soundweb programmer Konrad Fengler, with Wojciech Kopytek handling system setup and Dariusz Kuta serving as project coordinator.

Harman Pro

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Church Sound: How To EQ Speech For Maximum Intelligibility

The problem is unclear words are a distraction from the message

 
Haven’t we all had stories of misheard words? It could have been a song lyric or you misheard your spouse?  Maybe they mumbled a word or it just wasn’t clear what was said. This has been the cause for a few hilarious moments at our dinner table. 

The problem is unclear words are a distraction from the message. 

In the church environment, the pastor’s words must be clear. We can ensure this maximum intelligibility through proper speech EQ.

There are four topics to consider when it comes to the EQ’ing needs for the spoken word. 

1. Microphone location. We are fortunate in that most pastors now use wireless microphones. This means that the distance between the mic and their mouth is pretty consistent.  In the case of the headpiece, this is especially true.

In the case of the lapel mic, remember they should drop their chin to their chest and put the mic directly below that point. Long ago, I was taught “a fist away from the chin.” The point here is that we want the best sound isolation we can possibly get while having a good gain structure in place. 

Remember, the closer to the source, the more the proximity effect comes into the equation and you’ll need to EQ out some of that added bassiness.

2. The speaker’s natural voice. Just as every guitar has a unique sound, so does every person. You want to bring out the best qualities of their voice. You don’t want them to sound like a different person. Their vocal characteristics are also “what you have to work with.” 

This means you’ll need to know how to deal with quiet speakers, bassy talkers, and nasally preachers, just to list a few.  Not everyone has a great radio voice.

3. Presence of background music. Depending on your church, your pastor might talk with a running soundtrack. There is definitely an art to being able to play the right music for this. 

However, any type of music bed means you now have to make a space for the voice amidst the instrumentals. Instrumentals can easily blur the spoken word so you’ll have to plan on tweaking the EQ for the musicians as well. 

4. The environment. Just because a vocal boost at 400 Hz sounds good in one room doesn’t mean it will sound good in another room. One of myreaders runs audio outside…in Egypt. Any EQ work must take the environment into account. The settings for a “quiet room” won’t be the same for an echo-y room or an outdoor venue.

Now that we’ve got those out of the way, let’s turn to…

The Frequency Make-Up Of Speech

Our speaking voice has three frequency ranges that need to be understood:
1. Fundamentals. The fundamental frequencies of speech occur roughly between 85 Hz and 250 Hz.
2. Vowels. Vowels sounds contain the maximum energy and power of the voice, occurring between 350 Hz and 2 kHz.
3. Consonants. Consonants occur between 1.5 kHz and 4 kHz. They contain little energy but are essential to intelligibility.

In short, this means that the “power” of the voice does not equate to the intelligibility of the voice. Think of it like this…just because a person has a booming voice doesn’t mean they are easy to understand. 

Now that you understand the audio dynamics (fundamentals, etc) in a voice and the environmental concerns (background music), let’s turn to…

What You Can Do To Provide The Maximum Speech Intelligibility For Your Pastor

There are three things you can do for tackling the EQ’ing process for the spoken word:

1. Make room for the voice. As I mentioned above, the environment makes a difference in how you EQ the spoken word. We can only control what is coming into the mixing board, so wind and rain aside, let’s talk about music. 

Mixing a large band means making space in the sonic spectrum where each instrument/vocal can sit and sound unique; and of course then blending these sounds together into a tight mix.

The spoken word needs the same treatment when music is played underneath it. This can happen in two ways—

—A. Adjust volume. This can be done using compression or simple volume adjustments. The general rule-of-thumb is the music is there to support the spoken word – to sit underneath it.  Therefore, look to cut volume levels of instruments before you boost the volume of the speaker. You can also use compression to bring volume levels up and down as you wish.
—B. Adjust the mix. Cut the frequencies of the instruments where they are the same as that of the speaker. Boost the spoken word EQ in those areas a little if needed to present the music and the voice as two distinct sounds.

2. Know sibilance and how to avoid it. Sssssssibilance in vocals is when the sound of the letter “S” sounds more like a hissing snake. You can accentuate vowel sounds/add presence by increasing the EQ in the 4.5 kHz to 6 kHz. 

However, the “S” sound lives between 5 kHz and 7 kHz. Therefore, be careful when adding presence because you can easily go from a great sound to a hissy sound.

3. Focus on vocal quality. There is no simple 1-2-3 process to EQ’ing the spoken word. Therefore, take these points into consideration:

—Roll off the low frequencies if the proximity effect is causing unusual bassiness.
—Don’t roll off so much low end as the voice loses some of its umph. Yes, I’m using “umph” as a technical word.
—Boost in the 1 kHz to 5 kHz range for improving intelligibility and clarity.
—Boost in the 3 kHz to 6 kHz range to add brightness. This can help with speakers with poor intonation.
—Boost in the 4.5 kHz to 6 kHz range to add presence. Note that too much boosting in this area can produce a thin lifeless sound.
—Boost in the 100 Hz to 250 Hz for a boomy effect.

In Case Your Head Is About To Explode From Information Overload, Remember:

—The above points can contradict each other. There is no hard and fast rule. Mixing is as much an art as a science. Trust your ears over everything else.
—It’s possible that once you EQ the vocal channel that it’s a little lacking in the low end. Boost it a bit give it that full sound. Again, trust your ears. Close your eyes and ask yourself if it a) sounds natural and b) sounds clear.

Finally

EQ’ing the spoken word is about improving the quality of the sound so it sounds clear, is easy to understand, and sounds natural.

So much of our mix time goes towards the band. Make sure you spend those few crucial minutes working on the pastor’s vocal as well. 

Church was about the sermon long before music, skits, and cool videos rolled onto the scene.

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

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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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Wednesday, February 01, 2012

Grundorf Tour 8 Series Cases Deployed At Resorts World Casino New York City

Housed on the grounds of New York’s famed Aqueduct Racetrack, New York City’s first casino—Resorts World Casino New York City—opened its doors last October with some 5,000 video gambling terminals and electronic table games.

Additionally, there are 18 food and beverage courts, two restaurants, four VIP lounges, and a bar officially known as Bar 360, with its A/V electronics protected by cases from Grundorf Corporation.

Rockledge, FL-based The Integration Factory, a design/build firm that specializes in A/V, control, access control, and surveillance systems, was contracted to handle the design and deployment of the large quantity of electronics that help make Bar 360 the hotspot that it is, and they’re all housed in a Grundorf Tour 8 Series Drawer Case and four of the company’s Shock Rack cases.

“Bar 360 is brimming with high energy and serves as a focal point for sports fans and others to kick back and have a good time,” explains Carlos Gonzalez, The Integration Factory’s senior manager for business development. “When a sporting event is taking place, you practically become part of the action with the huge 16-foot high by 28-foot wide high definition LED wall that’s installed—not to exclude the multitude of additional video screens that adorn this room.

“There’s a lot of electronics handling the video and audio systems that reside in the room and all that gear required proper protection. That’s precisely what led to our selection of the Grundorf cases.”

The Grundorf Tour 8 Series drawer case that resides in Bar 360 is used to house microphones, assorted connectors, and for adaptor storage. As with all Grundorf cases, the Tour 8 drawer case can be custom ordered.

Bar 360’s case is outfitted with foam inserts to securely house the microphones, providing a safe haven for these valuable instruments when not in use. Optional locking latches are available to create a fully lockable environment. The drawers slide smoothly on sturdy steel glides and are representative of the attention to detail involved in the manufacturing of these cases.

“The shock racks are used to support the house console,” Gonzalez adds. “The devices mounted in those cases include power supplies for the house console, digital snake processors, DVD players, plus an assortment of intercom and wireless receivers.”

Grundorf’s Tour 8 shock cases incorporate a foam surround shock system to provide superior protection. Each case has a 1-inch thick shock foam base with a 1-inch thick high-density foam on the top and all sides. Each Shock Rack has front and rear removable lids for complete access to the equipment.

“Grundorf’s ability to custom build to one’s specifications, the quality of their workmanship, and the company’s ability to deliver the products on time were all important considerations that led to their selection,” Gonzalez concludes. “With their 3/8-inch plywood construction with a durable high impact resistant ABS laminated finish; these cases offer solid protection for the equipment they house. The overall build quality is very impressive and the company’s customer service is exceptional.”

Reflecting on the project, Gonzalez summarized by saying, “The Resorts World A/V staff is enjoying their Grundorf cases. The cases provide secure protection and, at the end of the day, that’s what it’s all about. Everyone is very pleased.”

Grundorf

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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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Thursday, January 26, 2012

Electronic Versus Physical: An Analysis Of Shaping Array Directivity

Electronic modification of an array’s directivity is not always a substitute for good old mechanical arranging or aiming. Here's a look at the differences

Modifying the directivity characteristics of loudspeaker arrays through electronic delay has become increasingly popular.

Whereas 20 years ago the only option was expensive dedicated digital delay units, and a few years later the original BSS Omnidrive was a luxury, the advent of inexpensive digital processing has changed the game.

The design of complex arrays using a relatively high number of processing channels, as required to electronically modify the directionality of an array, is now affordable and widely implemented.

However, virtual (electronic) modification of an array’s directivity is not always a substitute for good old mechanical arranging or aiming, as the two methods have widely differing radiation characteristics off-axis (i.e., to the back and sides).

Let’s look at the differences in the two approaches, how they differ across a number of array types, and suggest applications where each of them should be used with subwoofers.

Arrival Times
The reason why physically moving a loudspeaker backward is different from delaying it electronically may not be intuitively obvious, but is easily shown graphically.

Figure 1a shows two loudspeakers (“A” and “B”) located left and right at equal distance from both a listener positioned in front and another listener positioned behind.

Leaving aside subtleties such as the location of the time origin of the loudspeakers, since it does not influence the basic concept being discussed here, sound from loudspeakers A and B will arrive at the same time to both listeners.

If we move back loudspeaker B (Figure 1b), then loudspeaker A is closer to the front listener, so sound reaches that listener earlier. Behind the loudspeakers, of course, the opposite occurs.

If we return the loudspeakers back to their original positions, and then apply electronic delay to loudspeaker B (shown in Figure 1c as a diverted path length to the listeners), we see that the output of loudspeaker A arrives earlier than B in both cases (in front and behind).

Thus, it is graphically clear that physically moving enclosure B produces a significantly different result to electronically delaying it.

Focus On The Effect
Let’s now look at the implications within the context of a vertical array of loudspeakers, and predict the coverage of a column of omnidirectional sources.

I often prefer to display results via polar plots, because with plane mappings it’s often difficult to understand the behavior at distances other than those close to the system being modeled.

Also note that I’ll use mostly omnidirectional sources instead of “real-world” sources (with a certain degree of attenuation at the back, i.e., not perfectly omnidirectional) to focus on the effect that the arrangement is causing on the directional response of a single loudspeaker.

In Figure 2a and 2b, we have physically tilted a 12-element array that is 23 feet (7 meters) long downward by 30 degrees.

The front part of the radiation points down 30 degrees, and the back part points up 30 degrees, while left and right (i.e., 90 degrees to the sides) are pointing straight, as if the array had not been tilted at all.

Figure 2a shows a three-dimensional directivity balloon resembling some sort of “flying saucer” at an angle, while Figure 2b shows polar plots for the third octave bands between 80 and 160 Hz (the main lobe gets narrower as frequency increases).

In Figure 3a and 3b, the sources are delayed so that the main radiation is (electronically) steered 30 degrees down (by applying increasingly larger delay times from top to bottom).

The balloon looks a bit like a fat cone, showing that the 30-degree downward angle is taking place all around the array, not just in front of it.

This behavior is emphasized by manufacturers of electronically controlled (“digitally steerable”) column loudspeakers, correctly emphasizing that the use of their products yields better coverage than a single, down-tilted conventional enclosure.

Pointing Lobes
To provide another example illustrating the differences between mechanical tilting and delay steering, we modeled one of each in a room, this time using loudspeaker data with realistic nonperfect omnidirectionality.

The resulting pressure maps have been plotted onto the walls as well as the floor, and we’ve also drawn lines, at different horizontal angles, that represent the direction in which the main lobe is pointing.

In Figure 4a (mechanical), the lines follow the shape of a disk, which means that some of the lines are pointing to the walls, and the mapping indeed shows that significant SPL is being radiated towards the walls.

In Figure 4b (electronic), the lines form a cone and sound is mostly focused on the floor.

The 125 Hz octave band was used for the room predictions; while it is probably somewhat unrealistic of typical subwoofer bandwidth, the narrower coverage is helpful to exaggerate the effect for clarity.

It can also be seen that the covered area is roughly rectangular for the mechanical case and rounder for the electronic one. (Some may recognize the CADP2 graphics. What a beautifully elegant piece of software that was! RIP.)

Exploring Arcs
From the explanation earlier in this article, we can guess that an electronic arc (where input signal is increasingly delayed as one goes from the center to the edges of the array) will display identical front and rear radiation for omnidirectional sources.

A physical arc, in the far field, also provides symmetrical front and rear behavior – but - at close distances, rear levels will be higher.

This is because circular arc sources arrive simultaneously at the circle’s center, i.e. the array’s “virtual origin.” Accordingly, physical arc best practices should avoid any arc that displays an inconvenient center, particularly at center stage.

Figure 5a, 5b and 5c present polars for a physical arc of eight subwoofers spanning 120 degrees with a radius of 10 feet (3 meters).

In the near field (Figure 5a), the buildup of sound pressure at the back can be observed, with the array being an average of around 6 dB less sensitive at the front for theoretical omnidirectional sources (though this number changes widely with frequency as seen on the plots).

This translates approximately to the same level back and front for a typical real-life subwoofer (with a certain degree of directionality). Also, in the near field, the rear pattern is narrower at the back.

As we get farther from the array though (Figure 5b), the polars become symmetrical, with the same levels being radiated to the back and front. This was calculated at a distance of 98 feet (30 meters) from the center of the array.

Figure 5c shows the far-field results, made up of equidistant enclosures that would “virtually” follow the same arc as the physical arc above.

Unlike the physical arc, the electronic version shows the same levels back and front both up close and far away from the array.

In general, an electronic arc is preferred because it does not suffer from pressure build-up behind the array, and it requires less space in front of the stage.

And unlike array steering, where each element requires a different delay time, we can use an even number of elements, so that pairs can share the same delay, meaning one amplifier channel can power two boxes if needed.

Given today’s prices, an extra DSP unit dedicated to subs does not seem too much of a luxury. Mathematically, calculating required delay times for a straight line array of equally spaced boxes may be complicated.

However, a piece of string can be used to mark a circular arc on the floor as physical reference for measuring “virtual” distances for pairs of subs.

Case Study A: Flown array of subwoofers on an open-air concert. When flying a subwoofer array, if the array is mechanically tilted, the rear radiation lobe will point upward (Figure 6a) and minimize trouble.

Yet it might be tempting to go with a “clean” hang and implement electronic steering, in order to digitally down-aim low-frequency (LF) radiation.

Doing this, however, means that corresponding rear radiation will also be aimed downward, presenting potential noise problems with nearby housing, as shown in Figure 6b.

Case Study B: Opening up left-right subwoofers. Invariably, when left and right subwoofers are used, interference creates the notorious power alley, where LF system response is audibly louder.

Additionally, bass coverage is not uniform since interference patterns change with frequency.

One way to minimize left-right interference is to aim subwoofer arrays away from each other in order to reduce overlap.

If we aim the array physically (Figure 7a), the back radiation lobe will point to the stage, increasing LF spill (again, the extent of this will be reduced through the use of cardioid subs, be it off-the-shelf cardioid models or array elements made up of a cardioid arrangement).

However, if electronic steering is used (Figure 7b), the back lobe will point away from the stage.

This is actually the same as Case Study A, except for the fact that we are dealing with horizontal, not vertical, coverage.

Case Study C: 360-degree subwoofer array. Certain arena applications might call for 360-degree horizontal subwoofer coverage, as well as some degree of downward firing toward the seating.

Achieving this with mechanical aiming is just plain impossible, but it can be accomplished through the electronic realm.

The suggested design makes use of a somewhat unusual configuration. Since real subwoofers are not entirely omnidirectional (a typical 18-inch subwoofer box may show 4 to 6 dB less at the back relative to the front), to achieve the same level at both back and front, we use a “face-to-face” deployment.

And it might seem a bit counterintuitive, but a physically phase-aligned pair can also be achieved if the correct spacing is used between the two.

To avoid flying too much weight, we could alternate every other element in the array as seen in Figure 8, an arrangement that also minimizes obstructions to the expansion of the wavefront.

This two-column arrangement with electronic steering would generate the directivity balloon seen in Figure 3a (except that the sides would be slightly squashed), with the horizontal and vertical polars that can be seen in Figure 9.

As with any low-frequency array, a longer array generates a narrower radiation pattern, which means that different venues would require different lengths to suit their geometry.

From the point of view of level consistency, the arrangement in Figure 8, with real non-perfectly omnidirectional sources, would send slightly less SPL to the sides (in our case, around 3 dB less for a real single 18-inch front-loaded subwoofer), which would be desirable on a rectangular arena to compensate for the difference in distance to the closest and farthest tiers.

On the other hand, given the uniform downward profile, this configuration would be ideally suited, angle-wise, for circular venues such as a bullfighting ring or a Mexican Palenque.

Watch That Space
As we know from line array “laws” there is a maximum spacing between sources for any given frequency.

If that spacing is exceeded, the array loses the ability to control directivity, with higher frequencies showing lobes at the wrong angles and eventually losing directivity control. This is even more so for an electronically steered array, which requires a tighter element density.

Figure 10a shows a three-dimensional representation of the directivity balloon of an electronically steered array with excessive spacing (4.5 feet).

A significant top lobe can be seen that will surely create reverberation issues at that frequency in an indoor venue.

Figure 10b presents 80 to 250 Hz one-third octave polars for the same array where the three highest frequencies have gone haywire across the top part of the curve.

In contrast, a mechanically tilted array of subs (Figure 10c) with the same spacing only shows misbehavior at 250 Hz, which corresponds to a wavelength that correlates roughly to the spacing between sources, so it’s no surprise.

José (Joe) Brusi is an independent electroacoustical consultant. And thanks to Joan La Roda for the field phase measurements of the alternate face-to-face subwoofer configuration.

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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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Monday, January 23, 2012

AKG Launches WMS 40 MINI 2 Dual Wireless Microphone System

The latest addition to the successful line of AKG WMS 40 MINI series now features a dual channel receiver.  The AKG WMS 40 MINI 2 Dual wireless system, launched at Winter NAMM, offers high-definition audio performance, with efficient body pack and cardioid mic transmitters, ensuring advanced, reliable and quality sound for performances.

The WMS 40 MINI 2 Dual Package is available in three sets – the Vocal, Instrumental and Mix sets.  The Vocal set features two HT 40 MINI dynamic cardioid microphones and the SR 40 MINI DUAL receiver.  AKG’s Instrumental set includes two PT 40 MINI body packs - the smallest transmitters in its class that lasts up to 30 hours on one AA battery –the SR 40 MINI 2 and two guitar cables, while the Mix set offers one HT 40 MINI, a PT 40 MINI and the SR 40 MINI 2.

With swivelling antennae, easy-to-read LED meterings, volume control and the on-off switch on the front of the receiver, connecting and calibrating the plug-and-play system ensures engineers and musicians spend more time playing than worrying about the wireless technology itself.

“AKG’s WMS Series has proven to be very popular in the live sound industry and we are happy to continue offering reliable equipment to better serve musicians and engineers looking for their ultimate sound with minimal equipment,” stated Thomas Umbauer, product marketing manager – PPA, AKG.  “Expanding the capabilities of WMS 40 in the new Dual system exceeds the industry standards for smaller, quality wireless systems.  Our engineers continuously work to ensure our products offer the best in signal transmission and sound and WMS 40 MINI 2 Dual follows AKG tradition in every sense.”

The WMS 40 MINI 2 Dual system is available at an MSRP of $249.

AKG

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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

A 20 Percent Chance Of What? Weather And Live Production Work

Doing the extra footwork ahead of time can save a world of grief, not to mention things far worse

It’s the season when many head for indoor venues, leaving the “great outdoors” to winter hibernation.

But for some of us, especially here in the southland, outdoor festivals are still the rage.

Although the summer thunderstorms are long gone, bad weather of other types still poses a problem.

While I’m addressing this missive primarily to the newcomers in sound reinforcement, it never hurts any of us to brush away the cobwebs and remind our brains as well.

Regardless of the time of year, weather forecasting by the technical crew still plays a big role in the continuity of any outdoor show. While I generally trust the event planning staff to keep an eye on the sky (or at least they should be), I trust my crew even more.

My company recently completed a two-weekend festival that included a dose of it all: high winds, rain, and cold temperatures made for a near impossible event. Four outdoor stages full of PA and lighting, not to mention power distribution all over the place. Nerves on edge, and of course, the cry “the show must go on” echoes from the sponsors and promoters.

To which I respond “safety first!”

Most of us working have (or should have) a clause in our agreements that provides the sound/lighting crew with at least some decision-making ability with regard to shutting down when the weather gets ugly.

This is essential when the safety of stage and performance personnel – not to mention our paying patrons – is at stake. It should be a quick but calculated decision between tech crew and event coordinators whether or not to shut down. Further, a “plan B” should always be discussed at pre-event meetings.

Getting back to that recent gig, there was no heavy rain, but drizzle combined with 35 mph winds and mid-30s (Fahrenheit) temperatures. It wasn’t quite bad enough to totally shut things down, but plenty enough to make things pretty miserable.

At times like these, I really count on my wingman – the person who watches the radar constantly on his/her PDA, shuts down the phantom power supplies on consoles without being told, is just as handy with a welder as a soldering iron, and graduated at the top of the class in knot tying. When faced with 35 mph winds, this is the person that makes sure nothing moves!

We all need someone this handy and responsible if we’re not doing it ourselves. I used to be the wingman, but when handling larger venues and festivals as the front of house guy and technical supervisor, I simply don’t have the time to run it all. Lean on the wing man – mine always seems to be one step ahead of me and that’s a good thing.

We also position easy-to-get-to road cases that are specifically labeled “weather” and stock them full of plastic sheeting, tarps, towels, and several five-gallon buckets filled with everything from carabineers and rope to bungees and ratchet straps.

Additional steel safety cables are also a must. Secure those lighting trees and trusses. Sandbag those tents and canopies.

And regardless of what anyone else says, if it’s a heavy downpour or at the first sign of lightning, shut down and cover up. Cut off power sources – literally pull the plug if that’s the situation. Clear the stage and fall zones of personnel.

Doing the extra footwork ahead of time can save a world of grief, not to mention things far worse. Here’s to a safe and prosperous 2012.

Greg Stone has worked in live sound since 1976 and is the owner of Hill Country Ears Sound Company (www.hillcountryears.com) in south Texas.

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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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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

The Old Soundman: Studio Versus Front Of House

You’ve gotta be kidding me!

Dear Old Soundman:
Studio engineers verses front-of-house engineers - can they be one and the same?

Studio guys don’t want to let us get a shot at their jobs, because we would wipe the floor with them! Unfortunately, we would also wipe the floor with their clients!

Do I really want to spend three months listening to some rock star (who tries to pretend he’s a man of the people) reveal how completely dependent he is on the click track, and the Pro Tools, and his bloody re-amping?

The lights must be kept turned down nice and low, while his sycophantic entourage sits on the couch, smoking ganja and getting crumbs from their vegan pastries on my carpet.

Only the love and devotion I feel for my family keeps me from climbing on star boy and beating his head in with an RE-20 while shrieking, “play a damn song, why don’t you!”

If I ever did that, David, I would make sure to have the theme music from “Psycho” playing very loudly in the background.

Don’t even get me started on pretty boy from the record company, and his cowardly criticizing of my mixes behind my back. Hanging is too good for the likes of him! I’m thinking a year in solitary confinement, with the same Melvins record playing over and over again, 24/7.

By the way, you are honored and respected, OSM.

Righteous!

Signed,
A frontman/singer well cared for by his own OSM.
David P.

See, here is a smart guy who didn’t employ a younger sound buddy who would have asked for less money but doesn’t know his expletive deleted from a hole in the ground!

Our friend went and got an OSM who knows the drill, and it pays off in the long run because now David P. can exist in a superior, relaxing audio environment rather than wondering what input is about to go wildly intermittent or explode into hellish feedback.

We old guys change the batteries - you know what I’m saying? We not only check things, we double-check them! Maybe at one point in our lives, we did that mainly because we were all amped up on controlled substances deleted, so to speak, but now we do it because we know it’s the right thing to do!

And it pays off for our artists. If they want some Chippendale-looking guys, they can damn well go to Chippendale’s!

I like being the absolute ruler of my little acre. When you and pretty boy and Mr. Royal Rock Star come into my club, you do things my way or the highway!

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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Wednesday, January 04, 2012

Going Around The Corner: Setting Up Multiple Line Arrays for Wide Venues

A procedure for getting smooth coverage and minimizing comb filtering over the entire coverage angle

Often, it is necessary to cover audience angles wider than the horizontal pattern of a single line array, even an array of 120-degree loudspeakers.

In such cases, multiple line arrays are needed, as shown in the diagram below.

When multiple arrays are used to cover a single area, the loudspeakers must be positioned and aimed for best sound and minimum wave interference (aka comb filtering).

Here is a procedure for getting smooth coverage and minimizing comb filtering over the entire coverage angle.

1. Rig
A. Arrange side stack for a 5- to 10-degree overlap in the intersection zone. With 120-degree side loudspeakers, this will require the side stacks to be pointed more than 90 degrees offstage. Don’t worry - this looks strange but it works!
B. Hang the side stacks closely behind the front stacks. Get the two as close to each other as possible. This may require swinging the front stacks outward to clear the side stacks as they are hoisted up.

2. Tuning
A. Place a measurement microphone (or a good set of ears) in the middle of the overlap zone—where the little gold sphere is in the diagram.
B. Adjust the relative timing of the two stacks for time alignment at the test point.
C. Equalize the side array with a broad 5 - 8 dB dip at 200 Hz . This will prevent midbass beaming at 45 degrees off axis.

CONCEPT
This setup approach minimizes the amount of overlap between coverage areas of the adjacent line arrays.

In the small areas that do overlap, the procedure ensures that signals from the two arrays arrive in time synchronization, or nearly so.

Thus, there is only a small “zone of confusion” in which listeners are hearing sound from both arrays, and in that zone, time smear is minimized. EV’s experience is that when this procedure is followed correctly, overlap effects are nearly inaudible in most circumstances.

A different approach to going around the corner has been proposed by others. In their scheme, the line arrays are situated quite far apart—20-30 feet (7-9m) or more—and may be aimed so as to have considerable overlap.

As we understand it, this scheme is based on the idea that when one puts the loudspeakers sufficiently far from each other, the sound arrival times from the two arrays are very different at most listening positions.

So different, in fact, that with most program material the signals are fairly unrelated to each other. If two signals are unrelated to each other (the technical term is “decorrelated”), then they can not cancel and reinforce each other in the way that makes comb filtering.

We won’t comment on how good this approach sounds, but our opinion is that in many production environments, flying widely separated loudspeaker clusters isn’t feasible. We prefer the closely-spaced
approach, if only for that reason.

This article was provided by Electro-Voice Senior Scientist Jeff Berryman and the tech department at Electro-Voice.

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Monday, December 26, 2011

Drum Tuning Tips From The Famous Drum Doctor

These quick drum tuning tips could be lifesavers in your next session if you're new to tuning drums

 
If you’re doing a session in Los Angeles and you want your drums to instantly sound great, then your first call is to the Drum Doctors to either rent a fantastic sounding kit, or have your kit tuned.

Ross Garfield is the “Drum Doctor” and you’ve heard his drum sounds on platinum recordings from Bruce Springsteen, Rod Stewart, Metallica, Dwight Yokum,  Red Hot Chili Peppers, Foo Fighters, Lenny Kravitiz, Michael Jackson and many, many more.

Ross was kind enough to sit down for an interview when I wrote The Recording Engineer’s Handbook, but I’ve featured some of his tips in other books as well including The Drum Recording Handbook, The Touring Musician’s Handbook, and The Music Producer’s Handbook.

So, I like to think his tips are worth sharing! Here are a few of his quick drum tuning tips, which can be lifesavers if you’re new to tuning drums.

If the snares buzz when the toms are hit:
• Check that the snares are straight.
• Check to see if the snares are flat and centered on the drum.
• Loosen the bottom head.
• Retune the offending toms.

If the snare drum has too much ring:
• Tune the heads lower.
• Use a heavier head like a coated CS with the dot on the bottom or a coated Emperor.
• Use a full or partial muffling ring.
• Have the edges checked and/or recut to a flatter angle.

If the kick drum isn’t punchy and lacks power in the context of the music:
• Try increasing and decreasing the amount of muffling in the drum, or try a different blanket or pillow.
• Change to a heavier, uncoated head like a clear Emperor or PowerStroke 3.
• Change to a thinner front head or one with a larger cutout.
• Have the edges recut to create more attack.

If one or more of the toms are difficult to tune or have an unwanted “growl”:
• Check the top heads for dents and replace as necessary.
• Check the evenness of tension all around on the top and bottom heads.
• Tighten the bottom head.
• Have the bearing edges checked and recut as required.

If the floor tom has an undesirable “basketball-type” after-ring, try this:
• Loosen the bottom head.
• Check the top heads for dents and replace as necessary.
• Loosen the top head.
• Switch to a different type or weight top or bottom head like a clear Ambassador or Emperor).
• Have the bearing edges recut to emphasize the lower partials.

If the cymbals are cracking or breaking with greater frequency, try the following:
• Always transport the cymbals in a top-quality, reinforced cymbal case or bag to avoid nicks that can become cracks.
• Use the proper cymbals felts, washers and sleeves at all times.
• Avoid over-tightening the cymbal stand.
• Use larger or heavier cymbals that you won’t have to overplay to hear.

Hopefully these tips are useful to you in your next session!

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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