Friday, March 14, 2014

Beyma Introduces Two New 18-inch Cone Drivers At Prolight+Sound

New 18P1000FeV2 is an upgraded version of the popular 18P1000Fe model

Beyma has introduced two new 18-inch low-frequency cone drivers at Prolight+Sound/Messe in Frankfurt.

The new 18P1000FeV2 is an upgraded version of the popular 18P1000Fe model. It provides improved power handling—increased to 1,200 watts AES—while delivering accurate and controlled bass performance.

The 18P1000FeV2 has a redesigned magnetic assembly and convection circuit for enhanced efficiency and extremely low air noise, sounding again improved over the previous model.

Meanwhile, the new 18PWB1000Fe is designed to deliver a lower end and a deeper sound. It has a 4-inch voice coil design with a ferrite magnet.

The 18PWB1000Fe handles up to 2,000 watts program, has long excursion capabilities (12.5mm Xmax and 55mm peak to peak for mechanical displacement), and a low resonance point. It adds up to lend this driver a very deep “personality.”

18P1000FeV2 Specs:

Nominal diameter—18-inch
Nominal impedance—8 ohms
Minimum impedance—5.9 ohms
Power capacity—1200 W AES
Program power—2400 W
Sensitivity—98 dB 1 W @ 1 m @ ZN
Frequency range —25-2000 Hz
Voice coil diameter—4-inch
BL factor—27.6 N/A
Moving mass—0.230 kg
Winding length—21 mm
Air gap height—12 mm
Xdamage pp—52 mm

18PWB1000Fe Specs:

Nominal diameter—18-inch
Nominal impedance—8 ohms
Minimum impedance—6 ohms
Power capacity—1000 W AES
Program power—2000 W
Sensitivity—96.5 dB 1 W @ 1 m @ ZN
Frequency range —20-20000 Hz
Voice coil diameter—4 inches
BL factor—25.8 N/A
Moving mass—0.233 Kg
Winding length—30 mm
Air gap height—12 mm
Xdamage pp—55 mm


Posted by Keith Clark on 03/14 at 12:31 PM
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Focusrite Names Damian Hawley Director Of Global Marketing & Sales

Will manage the merged marketing and sales departments for the Focusrite, Novation and RedNet brands worldwide

Focusrite Audio Engineering has named Damian Hawley as director of global marketing and sales, where he will manage the merged marketing and sales departments for the Focusrite, Novation and RedNet brands worldwide, including Focusrite Novation, Inc. (the U.S.-based wholly-owned subsidiary).

Both Phil Wagner (president of Focusrite Novation, Inc) and Giles Orford (marketign director) remain in their respective positions and will continue to serve on the board of directors while reporting to Damian Hawley.

Dave Froker (managing director) states: “Following a period of strong company performance we have decided to bring marketing and sales together as a single organisation to position the business for it’s next phase of growth.”

Hawley, who joined Focusrite in 2004 and became sales director in 2009, adds, “The Focusrite and Novation brands have seen significant growth over the last five years and I now look forward to the privilege of leading our newly combined team of highly talented marketing and sales professionals.”


Posted by Keith Clark on 03/14 at 10:29 AM
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Clearwing Takes L-Acoustics KARA And KIVA To The Streets Of New York City

Biggest challenge was devising system for Super Bowl Boulevard for live performances, autograph sessions and more for a week

Clearwing Productions of Milwaukee was hired by the National Football League for three Super Bowl venues, all of which it supported with a complement of L-Acoustics loudspeakers.

The NFL’s official Super Bowl Tailgate party held at the old Meadowlands Racetrack featured American Idol’s Phillip Phillips, The Band Perry and the cast of Jersey Boys on a new K2 rig. The NFL On Location Super Bowl Club featured the Fray and Cyndi Lauper before the game on another K2 system hung at the Izod Center in East Rutherford.

However, the real test for Clearwing took place during the entire week prior to the game across the river in Manhattan.

Super Bowl Boulevard took over Broadway between 34th and 47th streets featuring a stage that by day hosted NFL player autograph sessions and by night bands like Blondie, the Bacon Brothers, Michael Cavanaugh and the Cafe Wha? House Band.

The system had to be compact because, although the stage spanned the Great White Way, emergency vehicle lanes needed to be created on both sides of the street, which kept the stage width down to a rather tight 32 feet. Further, the system had to ground stacked, which made the choice of KARA and KIVA, supported by eight SB18 and a pair of SB28 subwoofers, one that met both the logistical and audio needs.

“Stacking in the downstage corners of the stage, the width of the speakers took up minimal stage space; plus, being able to stack the SB18s directly behind the KARA stacks allowed us to get the bigger sound that a three-way enclosure would provide while still maintaining a small footprint,” says Clearwing’s Steve Harvey, who, along with system tech Trevor Powers, was in charge of the audio for the stage. “We had the SB18s running up to 100 Hz for improved low end from bass guitar and drums, and the SB28s in front of the stage added extra lows in the drums and playback elements.”

As if the logistical challenges of an outdoor gig in the middle of Manhattan were not enough, the weather in New York City that week was at times very cold.

“Our semi trucks had to be unloaded at midnight and the show started at noon,” says Harvey. Outdoor temperatures got down to zero Fahrenheit at times. “The amps were turned on at the beginning of the week and stayed on all week. The L-Acoustics gear was solid throughout, however; not a single blip from an LA-RAK—even in these extreme conditions—and the speakers were solid.”

When the bands played, the audio had to cover an area that extended a full city block. That may sound more like a gig for a full-sized K1 rig rather than the relatively diminutive KARA, but it was about coverage—not volume—which is where the KIVA boxes came into play.

“We deployed a pair of KIVAs every 40 feet on each side of the street,” adds Harvey. “There were three sets of delays total, and it was all about coverage and intelligibility. Those delays meant we could hit the very back of the block without thrashing the nine KARA boxes stacked on either side of the stage. It was a tough gig, but the L-Acoustics stuff worked great.”

Clearwing Productions 

Posted by Keith Clark on 03/14 at 10:20 AM
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Mackie Debuts Upgraded SRM450/350 Powered Loudspeakers

Four application-specific modes available to re-voice the loudspeaker for the application at hand

Mackie has announced upgraded versions of the SRM450 and SRM350 portable powered loudspeakers, including a new 1000-watt amplifier and several digital sound-shaping tools. (See them at Prolight+Sound/Messe in Frankfurt in Hall 8.0, Stand L56.)

“The original SRM450 was the first active portable loudspeaker and, today, is still the most recognized speaker out there. SRM has always meant pro performance and sound quality that’s easy to use at an incredible price,” says Greg Young, Mackie product manager, “The new SRM450 and SRM350 continue this legacy, offering improved power, sound quality and flexible tools perfect for any gig.”

Users now have a choice between four application-specific modes, each re-voicing the loudspeaker to be optimized for the application at hand. Also built-in is an intelligent feedback destroyer that simplifies sound check. At the push of a button, SRM instantly identifies and eliminates feedback using up to four narrow 1/16th octave filters.

Both models now benefit from Mackie’s HD Audio Processing, which combines patented acoustic correction DSP with optimization features like a precision crossover and driver time alignment and phase correction.

Application flexible, the SRM450 and SRM350 also incorporate an integrated 2-channel mixer with input-friendly Wide-Z inputs, and are housed in rugged, lightweight polypropylene cabinets with a range of mounting options avalable.

The new SRM450 and SRM350 will be available globally late Quarter 2, 2014. The SRM450 will have a U.S. MSRP of $629.99, and the SRM350 will have a U.S. MSRP of $519.99.

LOUD Technologies

Posted by Keith Clark on 03/14 at 09:46 AM
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Biamp Systems Appoints Vibhav Singh As North India Area Manager

Will oversee all sales and installations in the country's northern and eastern areas

Biamp Systems has announced the addition of Vibhav Singh to its growing sales team in India.

Appointed to the post of North India area manager, Singh will oversee all sales and installations in the country’s northern and eastern areas including Delhi/NCR, Punjab, Rajasthan, West Bengal, and the surrounding areas.

In his new position, Singh will support Biamp in strengthening existing relationships with integrators and end users, as well as continue the growth of Biamp audio solutions within the region.

Prior to joining Biamp, Singh was a regional sales manager for installed sound with Harman International and brings five years of professional audio experience in sales, project management, and customer service. As a new member of Biamp’s expanding Indian sales team, he is located in New Delhi and reports to Robin Ghose, regional manager — SAARC.

“Vibhav comes to us at a time of rapid expansion for Biamp in India,” says Ghose. “As we continue to grow our market presence, his experience in sales, installed sound, and project management will be instrumental to our success. We are thrilled to have him onboard and look forward to his involvement during such an exciting time.”

“Biamp understands that creating a genuine presence within a marketplace means putting customers first,” adds Singh. “In addition to being an industry leader in commercial audio solutions, Biamp has taken tangible actions for creating a culture where partnerships thrive. I am delighted to be a part of such a dynamic team.”

In addition to supervising sales and installations, Singh has held positions in areas ranging from product promotion, to AV systems integration, to national sales management. He holds a Bachelor of Technology degree from the Punjab Technical University and several industry certifications.

Biamp Systems

Posted by Keith Clark on 03/14 at 08:24 AM
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Celestion Introduces CDX14-3030 1.4-Inch-Exit Ferrite Compression Driver

High-performance titanium diaphragm HF device

Celestion announced the launch of the CDX14-3030; a new 1.4-inch exit, ferrite magnet compression driver at the Prolight+Sound/Messe in Frankfurt. (Hall 8.0, Stand J78)

With a 3-inch copper clad aluminum voice coil driving a titanium diaphragm, the device delivers 75 watts (RMS) power handling and 106.5 dB sensitivity across a frequency range of 500 Hz to 18 kHz. Recommended minimum crossover frequency (at 12 dB/octave) is 1 kHz.

According to Celestion head of engineering Paul Cork, “This compression drivers bring together the best elements of technology used in the most successful products of our CDX range. The design extracts the maximum performance from a unit that is surprisingly compact for a 1.4-inch-exit device.”

Like other CDX compression drivers in the range, CDX14-3030 makes extensive use of Celestion’s experience in Finite Element Analysis (FEA) to optimize magnetic and acoustic design, helping the device to achieve top performance at a highly competitive price point.

This new driver also use Celestion’s proprietary Sound Castle diaphragm clamping assembly to ensure even clamping pressure on the diaphragm surround, while actively reducing stress on the diaphragm during operation. This contributes to lower distortion while enabling the full internal volume of the rear cover to act as a loading chamber for the diaphragm, resulting in superior mid-band response.

With exceptional heat management, plus an edge-wound coil copper clad aluminium voice coil the CDX14-3030 is a high performance HF solution for a wide variety of larger format sound reinforcement applications.


Posted by Keith Clark on 03/14 at 08:13 AM
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DPA Microphones Appoints GerrAudio To Lead Canadian Distribution

Ontario-based company represents leading brands in mixing consoles, signal processing wired and wireless intercoms and more

DPA Microphones has announced the appointment of GerrAudio as its Canadian distributor.

GerrAudio is an established distributor of professional audio equipment for the live performance and broadcast industries across Canada, representing some of the industry’s leading wired and wireless intercom, digital mixing console, signal processing and audio test and measurement equipment brands.

“We’re very fortunate to be chosen to represent DPA Microphones in Canada,” says Bob Snelgrove, president of GerrAudio. “The quality and performance of the company’s mics are consistent with GerrAudio’s long-standing philosophy of bringing unique audio solutions to the technical teams that support highly creative artists. DPA’s array of products perfectly complements our current collection of state-of-the-art audio equipment.

“Additionally, he continues, “while the company’s reputation precedes GerrAudio, we know that many professional users are not fully aware of the flexibility and depth that is available across the product line. To that end, our primary goal will be to provide greatly expanded accessibility to the company through such means as product introductions, education and demonstrations, along with quick accessibility to products and technical support.”

The partnership with GerrAudio signals DPA Microphones’ continued commitment toward growth and customer service throughout North America. To ensure greater product availability for customers in Canada, GerrAudio will carry a full stock of DPA products at its head office in Brockville, Ontario. 

“GerrAudio has an impressive reputation as one of the leading pro audio distributors in Canada,” says Niels Jørgen Øhrgaard, executive vice president of sales, DPA Microphones. “With a local Canadian distributor, we ensure better and more attentive service and support for our new and existing customers in the region. We’re excited about this new partnership and have high expectations for the opportunities it will bring.”

DPA Microphones

Posted by Keith Clark on 03/14 at 08:00 AM
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Yamaha Commercial Audio Announces Version 2.0 Of CL Series Digital Consoles

Yamaha CL V2.0 Available in May

Yamaha Commercial Audio has announced Version 2.0 of the CL Series of digital consoles at Prolight+Sound/Messe in Frankfurt.

The update includes enhancements for sound reinforcement applications as well as mix-minus capabilities for the broadcast market, broadening the range of applications where CL Series performance can be advantageous.

Another significant enhancement in v2.0 is discovery and head amp control for the compact QL Series consoles, also launched at PL+S, that inherits CL Series features and performance. A QL console can function as both monitor mixer and I/O rack, for example, while a front-of-house CL console can remotely control the head amp gain of the QL console’s I/O.

New features in CL v2.0 include DCA Roll-Put: Channels assigned to DCA groups can be instantly called up to the console faders for enhanced operational flexibility; output DCA enables the stereo/mono bus masters, mix bus masters, and matrix bus masters to be assigned to DCA groups. Mix minus, an important feature in broadcast applications, is now provided; with one simple operation the signal from a particular channel can be removed from a specified bus.

“CL v2.0 not only adds features that are also included in new the QL Series launched today, but provides additional support for our core sound reinforcement customers with features that were only previously available on PM digital mixers,” states Marc Lopez, marketing manager, Yamaha Commercial Audio.  “Future plans include built-in Dugan automatic mixing similar to the new QL Series and additional support for broadcast applications in the CL Series. We will continue to keep a close eye on the market in order to provide features and performance that will maximize our customers’ investment in CL.”

With Read Only Scene Memory, it is now possible to create read-only scene memories. A new daisy chain insert feature allows two devices to be inserted into one channel or bus for enhanced processing freedom. A GR meter (dynamics meter) option within the channel name display will show the dynamics 1 and dynamics 2 gain reduction meters in the channel name display.

The CL v2.0 update will be available for download from the Yamaha Commercial Audio web site in May and is free of charge. Those attending the upcoming NAB show in Las Vegas can also check out v2.0 at booth C2143.

Yamaha Commercial Audio

Posted by Keith Clark on 03/14 at 07:08 AM
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SM Pro Audio Announces uMiX Series Of Wi-Fi Remote Controllable Digital Mixers

Models provide 56-bit double precision audio signal path with detailed parametric EQ, dynamics, and digital FX available on all channels

At this week’s Prolight+Sound/Messe in Frankfurt, SM Pro Audio has announced the uMiX family of Wi-Fi remote controllable digital mixers, including the uMiX mini (8 channels), uMiX 12 (12 channels), uMiX 16 (16 channels) and uMiX 24 (24 channels). (SM Pro Audio is in hall 5.1, stand B86)

uMiX models provide a 56-bit double precision audio signal path with detailed parametric EQ, dynamics, and digital FX available on all channels. All also include an on-board secure Wi-Fi and Ethernet router and remote microphone preamps, providing a truly out-of-the-box remote mixing solution.

Built for both desktop and touch devices, uMiX brings remote access, multi-touch, key commands, and an intuitive GUI with the look and feel of a hardware mixer. All this is needed is a desktop PC, laptop, tablet or smart phone with a modern web browser (no apps or software downloads).

uMiX mixer functions include sub groups, view groups, mute groups, mute FX, mute all, permissions, channel safes and more. uMiX stands for “You Mix,” which means everyone in the band can control their own monitor mix. A copy main mix to aux button makes it simple to set up personal mixes. 

Gigs can be captured with built-in “One Touch” stereo recording to any USB stick/drive (excludes uMiX mini). Simultaneously play back digital media using the integrated digital media player for remote controlled background music and/or backing tracks—no docking required. 

The uMiX16 and uMiX24 offer additional processing on the master outputs from Waves Audio.  Waves MaxxBass and MaxxVolume adds drive, punch and polish to mixes while protecting the PA.

uMiX Series Overview:
—Built in router (Wi-Fi and Ethernet)
—Remote microphone preamps
—Wide device support
—Scalable - Link 2 x uMiX24 units with one Cat 5 cable for a total of 48 inputs
—Optional multi-track recording expansion card is available for the uMiX 16 and 24
—One Touch stereo recording. No Dock required
—Media Ppayer for background music or backing tracks
—HDMI display output
Danny Olesh, SM Pro Audio CEO, states, “uMiX is a really exciting product, not just for SM Pro Audio, but for the industry. uMiX technology breaks new ground in relation to affordability especially considering the on-board remote preamps and built in Wi-Fi router.

“We are also very proud that the uMiX software works on a range of devices and operating systems maximizing user choice. No downloads, no apps. uMiX is fully remote controllable with all the benefits of digital mixing and delivers a range of large console features at small console prices.”
Projected European street pricing including 19% VAT:
uMiX Mini - €199.99
uMiX 12 - €399.99
uMiX 16 - €699.99
uMiX 24 - €999.99
Projected U.S. MAP pricing:
uMiX Mini - $249
uMiX 12 - $499
uMiX 16 - $699
uMiX 24 - $999

Expected availability in-store Q3, 2014.


SM Pro Audio

Posted by Keith Clark on 03/14 at 05:15 AM
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Thursday, March 13, 2014

Avid Announces Expanded Live Sound Plug-In Platform For S3L Mixing System

Release of free VENUE 4.1.3 software update provides support for the latest qualified plug-ins

At Prolight+Sound/Messe in Frankfurt, Avid has announced an expanded AAX live sound plug-in platform for the S3L live mixing system, including new plug-ins from Avid as well as Crane Song, McDSP, and Sonnox. (Avid is in hall 8.0, stand G40A)

“With its unique capabilities and growing plug-in platform, Avid S3L is quickly becoming the console of choice for users requiring a compact, no-compromise system that can meet the toughest challenges,” says Sheldon Radford, director of product management, live systems and consoles at Avid.

“Avid S3L gives live sound professionals the greatest flexibility to choose the right plug-in for the job,” he adds. “The plug-ins appear within the VENUE software and are fully integrated within the session file and snapshots. This provides the best possible user experience without the need for additional hardware or management.”

“Partnering with Avid helps ensure that we stay at the forefront of industry innovation when it comes to delivering high-performance plug-ins for the live sound industry,” said Colin McDowell, CEO/CTO at McDSP. “Together, AAX and VENUE offer live sound professionals the perfect integrated processing solution, so we’re delighted to be able to bring our range of plug-ins, including our 6030 Ultimate Compressor, to the Avid S3L system.”

The release of the free VENUE 4.1.3 software update provides support for the latest qualified plug-ins, with Avid continually qualifying additional AAX live sound plug-ins. As part of the update, current and new S3L customers can also download additional free AAX plug-ins for their system through their Avid customer account.


Posted by Keith Clark on 03/13 at 02:54 PM
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Why Measured System Response Doesn’t Match What’s Heard

Many audio field technicians are now in possession of measurement systems that can be used to assist the listening process in equalizing sound reinforcement systems. But, they’re often surprised to find that the measured system response correlates poorly with subjective impression of how the system sounds.

In other words, the system can sound good when it looks bad on the analyzer, and it can sound bad when it looks good on the analyzer. As a result, some users have become frustrated and distrustful of analysis systems in general.

Let’s look at why the eye and ear do not always agree on what is best regarding the response of the sound system. First, consider the most popular methods of measuring the response of the sound system.

By “response,” I am referring to the magnitude of the frequency response as displayed on a dB (vertical) vs. logarithmic (horizontal) scale. The goal of technical system equalization is to produce a “flat” horizontal line on this display.

Working In Real Time
The real-time analyzer (RTA) is essentially a bank of meters, each driven with a 1/n-octave constant percentage bandwidth filter so that only the level of a limited range of frequencies is displayed by each meter.

The original RTAs used analog meters, but current versions use a vertical row of LEDs for each 1/n-octave band. One-third octave resolution is the most popular, and correlates well with the response of the human auditory system. 

The measured vs. “ideal” response for the direct field of a loudspeaker.

The RTA input is fed from an omnidirectional test microphone located at a listener position. Omnis are used because they typically have a very flat, “benign” frequency response over most of their band pass.

RTAs can also be software-based, utilizing the sound card on a personal computer to provide the A/D conversion of the microphone output voltage. A mathematical algorithm (the FFT) is used to produce the previously described dB vs. frequency display.

These “digital” analyzers emulate their analog counterparts in how the information is displayed, but differ in that the filters and display is the product of a computer algorithm rather than analog filters. This type of RTA is more versatile, as the octave-fractions, colors, etc. are under software control.

Regardless of which type is used, the standard method-of-use is to drive the sound system with pink noise (equal energy per 1/n octave) and adjust the system equalizer for a “flat” magnitude response on the analyzer display.

RTAs are powerful tools when certain guidelines are followed, but indoors they can indicate a system response with poor correlation to what the listener is hearing. The major consideration is the placement of the measurement microphone.

The effect of increasing distance outdoors (top) versus indoors.

If the mic is placed in the near field of the loudspeaker (typically less than 10 feet), the correlation with human hearing is pretty good. At this position, the direct energy from the loudspeaker dominates what is being observed on the analyzer and very little of the reflected energy from the room is included in the displayed response. Adjustment of the equalizer for a flat direct sound field on the analyzer produces a desirable result.

The down side to the near-field placement is that the measured response is very sensitive to small vertical movements of the microphone when the loudspeaker has offset vertical components (as most do). This sensitivity can be reduced if the microphone is moved to a greater distance from the loudspeaker (into the far field) since the path-length difference back to the individual components becomes more equal.

But, as the microphone is moved further away, the reflected energy from the room begins to dominate the displayed response.

Giving Equal Weight
Microphones have no “perceptual” abilities. They do not localize sound or discriminate early sound energy from late energy like humans do.

A listener at a distance remote from the loudspeaker will pay more attention to the direct field of the loudspeaker than sound that is building-up in the room. A microphone gives equal weight to all energy without regard to where it is coming from.

A simple experiment to verify this is to stand at the microphone position and listen to the loudspeaker and then route the mic through a headphone amplifier and listen to it through headphones - not the same thing at all.

Low frequency sounds tend to linger in rooms longer than high frequency sounds, because most rooms have more high frequency absorption than low frequency absorption.

As such, the room becomes “bass heavy” when the total sound field is considered. This extra low frequency information will dominate what is observed on the RTA, and the knee-jerk reaction is to attempt to “flatten” the response by boosting the high frequency bands on the equalizer.

The result is a system with excessive high frequency output and a resultant “harsh” sound quality.

When RTAs are used in this manner, it is important to equalize to a “target curve” rather than for a flat frequency response. The popular “X” curve for theaters is flat to 2 kHz, where it starts rolling off the high frequency response at about 3 dB per octave. It is -10 dB at 10 kHz relative to 2 kHz.

A target curve can be used with the RTA to compensate for the low-frequency build-up that occurs in many rooms.

This represents 1/10th power at 10 kHz relative to flat response. The one-third-octave analyzer and the target curve have served sound practitioners well for years, and remains a viable approach to system calibration.

Recent Methodologies
Technology has yielded some new methods for acquiring the system response at a listener position. A complex comparison (both time and frequency information) of the input and output of a system is called the transfer function. It includes both the magnitude and phase response of the loudspeaker/room at the microphone position.

This has become a popular method of analysis, as it allows any input stimulus to be used to test the system, since the displayed response is just the difference between “what you put in” and “what you got out.”

Transfer function analysis has the added advantage of the ability to use a “time window” to exclude late arriving energy from consideration in the response. This can prevent the low-frequency build-up problem that plagues traditional real-time analysis. With proper implementation of a time window, the system response can be adjusted without the need for frequency weighting via a target curve.

A full-bandwidth transfer function measurement (with Smaart) using variable time windows. This measurement was made indoors at about 50 feet from the loudspeaker.

A major difference between transfer function analysis and 1/n-octave real-time analysis is that the former requires the removal of the signal delay between the two signals being compared. The stimulus (the reference signal) always has a much shorter path back to analyzer input than the output of the measurement microphone. Sources of delay include the travel time through the air and the latency of digital processors.

Failure to properly synchronize the reference signal and the microphone’s signal will result in an erroneous display of the system’s response. The length of the time window must also be selected - in other words, “how much of the room decay do I want to include in the response?”

Unfortunately, there is not an optimum size for the entire spectrum. A short time window excludes much of the room decay at the expense of low-frequency resolution. A long time window improves frequency resolution at the expense of gathering too much of the room’s decay. A compromise is required.

The human auditory system perceives pitch on a proportional (logarithmic) frequency scale. This is one reason that we use constant-percentage bandwidth filters for tuning audio systems - the bandwidth grows with increasing frequency.

Frequency-dependent bandwidth suggests that the length of the windowing function used in transfer function analysis should be varied in the same manner—a decreasing length with increasing frequency. This produces a somewhat “anechoic” response at high frequencies with increasing frequency resolution as frequency decreases.

The time window length is a function of frequency, with even the longest window (highest frequency resolution) excluding much of the late energy from the room.

Another caveat of this type of analysis is that much greater frequency detail is possible than with the typical 1/3-octave banded display. Phase interference effects from reflections or multiple drivers are clearly visible on the analyzer.

Such anomalies are almost always position-dependent, so careful “corrections” at one seating position will be inappropriate for another.

Both the loudspeaker and the measurement microphone should be carefully positioned to avoid the creation of very early high-level reflections.

Special Effects?
The “floor bounce” effect is a common example of a very early reflection (typically within a few milliseconds of the first sound arrival) that produces a unique acoustic response for each listener seat for all but the lowest octaves of the spectrum. This is an example of “less is better” when measuring the response, as a 1/3-octave display lacks the resolution to observe the effect in detail and produces less of a temptation to “fix” it.

Placing the test microphone on a stand makes it impossible to observe the loudspeaker’s response without interference.

The floor bounce effect can be minimized by use of an appropriate frequency-dependent time window or by simply laying the measurement microphone on the floor, or on a board placed across the listener seats. The effect usually disappears with the presence of an audience, so we do not wish to consider it when tuning the sound system.

The use of variable-length time windows and the synchronous transfer function allow the system to be tuned in a manner similar to the near-field RTA method (flat response on a log frequency display), even at remote positions in the room.

It is superior to the RTA method in that the effects of air absorption are readily apparent and can be compensated for via equalization. Near-field techniques do not include air absorption for the simple fact that the sound has not traveled very far before it is picked up by the microphone, so it hasn’t passed though enough air to be significantly attenuated.

By far, the biggest problem with tuning sound systems is failure on the part of the technician to recognize anomalies that cannot be corrected with equalization.

The test microphone was placed on a stand for this measurement. Note the comb filtering due to the floor bounce effect.

The equalizer is a “global” device, meaning that its response curve will be impressed on all of the sound radiated from the loudspeaker, regardless of the direction in which it is radiated.

Many, if not most, of the anomalies observed on the analyzer are unique to each listener position. The technician must learn to recognize and ignore such events. They include:

—Floor-bounce effect

—Interference between multiple drivers

—Reflections from objects near the mic or loudspeaker

Events that produce a more global effect, and can therefore be addressed with equalization include:

—Boundary-loading of loudspeakers

—Coupling between multiple low-frequency drivers

—The direct-field loudspeaker response

With training and experience, the system technician can implement methods that reveal system imperfections that are correctable, and hide those that are not - regardless of the analysis method used. Better yet, system designers can design systems with fewer “un-equalizable” problems.

Bad Is Always Bad

The old adage “An ounce of prevention…” could never be more true. System equalization then becomes meaningful and fast, providing the “icing on the cake” of the performance of a sound system.

It makes a good loudspeaker sound better, and brings the system to its fullest potential given the acoustic environment into which it is placed. A bad room is a bad room, regardless of how we process the electrical signal that drives the loudspeakers.

When used properly, the traditional 1/n-octave real-time analyzer is a useful tool outdoors at any distance. Indoors, the effects of reflected sound and non-frequency-uniform room absorption produce some problems for this method at measurement distances remote from the loudspeaker.

One solution is to utilize a weighting curve that reduces the target level of the high-frequency portion of the spectrum. Attempts to achieve a flat system response at remote listener positions without the use of a weighting curve can result in harsh-sounding systems and even component damage.

Transfer function analysis addresses some of the shortcomings of the 1/n-octave RTA, but it requires greater expertise on the part of the user. Failure to properly compensate for the time differential between the reference and measured signal can produce wildly erroneous results.

The test microphone was placed on a stand for this measurement. Note the comb filtering due to the floor bounce effect.

The time window length must also be selected by the user, and different lengths will produce different displayed responses. A frequency-dependent time window produces a display that correlates well with human perception.

The most important feature of either measurement method is a knowledgeable operator - one who understands the caveats of each approach along with the basic characteristics of the human auditory system.

None of the questions raised here have a single, correct answer. This means that experience, good judgment, and common sense rooted in Newtonian physics are still a part of the measurement process.

The microphone was placed on the floor for this measurement. Anomalies inherent to the loudspeaker are now visible on the analyzer.

Sound is a relatively easy quantity to measure, but measurements that correlate with human perception are much more difficult. Analyzers driven by omni directional microphones do a poor job of emulating the human listener. At this point one could ask, “So why measure at all? Why not just listen?”

Next time, we’ll have a look at this provocative question.

Pat and Brenda Brown own and operate SynAudCon, the leading independent professional audio education source, with training sessions held around the world and online. For more info go to www.synaudcon.com.


Posted by Keith Clark on 03/13 at 02:42 PM
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NSCA Reports Successful, Sold-Out 16th Annual Business & Leadership Conference

Conference revealed innovative ways to increase profits, promote engaged workplaces, create irrefutable market offers, and approach converging technology.

More than 300 professionals from the electronic commercial systems industry recently gathered in Dallas for the 16th annual NSCA Business & Leadership Conference to share ideas, discover new ways to encourage business growth, and build relationships with industry peers.

NSCA’s Business & Leadership Conference reached sell-out levels, with registration closing after 300 attendees. This conference -– developed exclusively for leaders within systems integration firms –- revealed innovative ways to increase profits, promote engaged workplaces, create irrefutable market offers, and approach converging technology.

“As always, the BLC was a highlight to our year,” says John Bangs, president of AV/COM Integrators in Severn, MD. “Excellent information, a relaxing atmosphere, and great insight to help us expand our firm’s capability into the future.”

Employee engagement was a recurring theme at this year’s Business & Leadership Conference. Keynote Adrian Gostick, New York Times best-selling author of “The Carrot Principle” and “All In,” shared insights about the factors affecting business performance – including employee engagement – based on two years of research from 300,000+ global business leaders.

Keynote Daniel Pink, New York Times best-selling author of “Drive” and “To Sell Is Human,” offered new sales approaches to help firms stand out in a world where information is abundant. Informed buyers – surrounded by information, choices, and ways to talk back to brands – now control the sales conversation, says Pink. It’s no longer an economy of “buyer beware,” but “seller beware.”

“We were amazed by the outpouring of positive responses from our attendees, sponsors, and speakers,” says NSCA executive director Chuck Wilson. “This sell-out event was full of industry professionals who were anxious to hear from our speakers about concepts and theories to improve their employees and their businesses.”

At the Business & Leadership Conference, NSCA recognized Bob Coffeen as the annual Per Haugen Lifetime Achievement Award winner for his industry contributions. Five Excellence in Business Award winners were also recognized for implementing new business, growth, marketing, and professional development strategies.

And for the first time, NSCA welcomed AV Nation to its Business & Leadership Conference. For attendees who want to relive the event – or for industry leaders who had to miss the conference – AV Nation produced two podcasts featuring event takeaways and insights on the state of the industry, how the economy will affect integrators, and upcoming trends.

For spurring conference dialog and sharing lessons, NSCA also recognized Brock McGinnis Westbury National and Kourtney Govro of Sphere3 and All Systems as BLC’s two “Most Prolific Tweeters.” They each received a pair of Shure noise-isolating earbuds. The #NSCABLC hashtag lit up across Twitter as attendees shared tidbits from presenters and panel discussions, and engaged fellow audience members with questions and comments.

“Once again, this conference stands out,” says John Graham, executive vice president at Solutions360. “We get great value from our sponsorship of the event, but it’s just as much for our own professional development and great insight into what’s important to our customers and prospective customers’ businesses.”

NSCA thanks the 2014 sponsors:
• Host Sponsor: Atlas Sound/IED
• Keynote Sponsor: Synnex
• Media Sponsor: Commercial Integrator
• Platinum Sponsors: AMX; BIAMP; Chief; Shure; West Penn Wire
• Gold Sponsors: Almo; Cisco; FSR; Gepco; LG; Liberty AV; MediaVision; Panasonic; Rauland-Borg; Sharp; Solutions360; SurgeX; Tannoy; ViewSonic
• Session Sponsors: Bose; Listen; Middle Atlantic
• Item Sponsors: Herman; Solutions 360
• Integration Sponsors: Bosch; BTX; Eaton; Kramer; Sennheiser; Stealth Acoustics
• Conference Endorsers: PSA Security Network and USAV Group

NSCA announced the 2015 Business & Leadership Conference details at the conclusion of the 2014 conference, receiving nearly 60 on-site registrations. The 2015 Business & Leadership Conference will be held on Feb.26-28 in Tampa,.


Posted by Keith Clark on 03/13 at 01:54 PM
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Mackie Thump Series Loudspeakers Get Power Increase, Joined By New Subwoofer

Both full-range models now driven by up to 1,000 watts on onboard power

At Prolight+Sound/Messe in Frankfurt, Mackie has launched updated Thump Series powered loudspeakers, now driven by up to 1,000 watts on onboard power. (Mackie is in hall 8.0, stand L56)

The series includes the full-range Thump12 (12-inch) and Thump15 (15-inch) joined by the new companion Thump18S (18-inch) subwoofer.

Thump Series loudspeakers include onboard optimization that includes precision crossovers, transducer time alignment and a unique 3-band EQ with a sweepable mid. Models also offer combo mic/line inputs, flexible mounting options, and a stage wedge angle, 

The new Thump18S sub can be driven by up to 1,200 watts of onboard power, housed in a band-pass design. System protection is in place, as well as a range of I/O that makes integration easy.

“Thump users are amazing…and amazingly vocal. We’ve heard you loud and clear,” states Greg Young, Mackie Product Manager. “You love the unmatched bass of Thump, but want more power to reach even more people. With the new Thump, you get more power and a substantial increase in bass performance at the same extremely affordable price.”

The new Thump Series models will be available globally in quarter 2 of 2014. The Thump12 will have a U.S. MSRP of $389.99, with the Thump15 carrying a U.S. MSRP of $469.99 and the Thump18S offering a U.S. MSRP of $879.99

LOUD Technologies

Posted by Keith Clark on 03/13 at 01:39 PM
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Amadeus Releases New ML 8 Compact Subwoofer At Prolight+Sound

Built around a proprietary internal reinforcement structure designed to neutralize any standing waves

At Prolight+Sound/Messe in Frankfurt, Amadeus has announced the release of the new ML 8 compact subwoofer, designed for space-limited installation or live applications. (Amadeus is in hall 8.0 at stand G19)

The ML 8 is the smallest member of the company’s ML Series that includes dual 18-, 18-, 15-, and 12-inch models, available in passive, self-powered and EtherSound versions. It can be paired with the new PMX 4 compact loudspeaker, also just released at Prolight+Sound.

As with other ML Series models, the ML 8 is built around a proprietary internal reinforcement structure designed to neutralize any standing waves and to suppress energy loss caused by vibrations. This unique construction technique creates, among other things, a dramatic cut in the level of cabinet coloration of the sound using a combination of interlocking panels arranged in two perpendicular planes.

Michel Deluc, lead designer at Amadeus, states, “We started to develop the ML 8 from the ML 12 ultra-compact format and tried to reduce its size by 25 percent while preserving both the sonic clarity, the real tonal accuracy and the extreme precision of the lowest frequencies specific to Amadeus LF loudspeakers.

“We achieved this through a quite complex internal structure, hosting several tuned notch resonators and a terrific 8-inch (200 mm) transducer with a 3-inch (75 mm) ventilated voice coil, designed to reduce all forms of distortion and coloration to a bare minimum.”

The ML 8 draws 600 watts peak from the line under an 8-ohm impedance and produces 118 dB peak SPL.

The subwoofer is available in a wear-resistant black (water-soluble) paint finish as well as standard and custom colors. The cabinet and the acoustical fabric covering the front grill can each have a different color.


Posted by Keith Clark on 03/13 at 01:08 PM
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Martin Audio OmniLine Upgrade At Most Holy Rosary Church In Syracuse

New array keys system upgrade to meet current standards for articulation and clarity

Brown Sound Equipment Company of East Syracuse, NY, recently installed a Martin Audio OmniLine array in the historic Most Holy Rosary Church in Syracuse to upgrade the sound system and meet current standards for articulation and clarity.

Part of Most Holy Rosary Parish founded in 1913, the church dates back to the site’s original groundbreaking in 1927 through over 60 years of growth and expansion resulting in today’s remarkable construction that is central to the Catholic community in Syracuse.

“Having been in this business for over 45 years, it’s most satisfying when a customer repeats with you as in this installation at Most Holy Rosary Church, and that the technologies we now have at our disposal really do make a tremendous difference in outcomes,” states Brown Sound president Jim Messinger.

The Martin Audio OmniLine replaces a huge custom central cluster point-source system that Brown Sound designed in 1991. The reverb time was the same then as it is now, approaching 4 seconds.

“These clusters with large long throw horns were the only ‘tools’ we had at the time to minimize splashing sound all over the place and still throw the energy 135 feet to the last pew,” Messinger explains. “Unfortunately, the cluster was large, ugly, black, and a visual distraction in front of that beautiful artwork above the altar. It also was a major challenge to install given the height, 53 feet to the peak of the church. Everything had to be installed with a system of winches above the dome.”

Messinger recalls the OmniLine solution came down to “a matter of timing. I’d played with the idea of column type line arrays, but the church didn’t want the system to visually intrude on the altar and worship area. After a Martin Audio sales rep recommended OmniLIne, we did a demo at the church with a single module and it sounded very good, even in a big empty space.”

The installation is built around a 16-enclosure OmniLine array so compact that it’s barely visible in the dome above the altar. A Mackie DL1608 iPad-controlled mixer replaces an analog console for the music ministry microphones.

Messinger reports that Martin Audio’s array guide software made assembly of the modules in this array extremely easy. “The mechanics of the assembly are also well thought out and engineered,” he adds. “The coverage also allowed us to eliminate all of the original side fill delay speakers.

“The bottom line is always comes down to ‘how does it sound?’” he concludes. “I attended a service when the pews were approximately 65 percent filled. Even after 45 years of listening to results, I was honestly impressed at how good, how articulate, and how natural the Martin Audio OmniLine sounded in this space. I’ve been in the business a long time and frankly I wasn’t expecting to be this impressed.

“The director of music mentioned that his choir members are very impressed with the system, they hear everything a lot better because there’s been so much of an improvement in articulation and clarity.”



Martin Audio
Brown Sound Equipment Company

Posted by Keith Clark on 03/13 at 12:12 PM
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