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Tuesday, October 27, 2009

d&b audiotechnik Releases New Version (v1.8) Of R1 Remote Control Software

Incorporates several new features; available for download from company website

d&b audiotechnik has released the latest version of its R1 Remote control software, v1.8, allowing a system design to be created, its performance predicted with d&b ArrayCalc, and the final system configuration applied either offline or onsite within R1 and then manipulated remotely.

The whole menu of DSP control can be applied down to an individual d&b amplifier channel, groups of loudspeakers or the complete system. At the touch of a button, the System Check device diagnostics enable detailed monitoring as and when required, before, during, or after a show.

Within installation projects, system integrators can configure R1 Remote control software to offer different levels of control, simplified functionality for daily use and more complex functionality when multiple applications of a room are required.

In low light indoor environments a darker screen design can be used or this can be switched to a brighter design for outdoor situations in daylight.

Further, the new version incorporates templates, which consist of a group of pre-configured elements that can be dragged and dropped into the workspace.

Templates provided by d&b are included in the R1 file while the user can also pre-configure a group of elements to create their own templates. For example, they can include the descriptions and range settings that can be saved and made available across all R1 project files and shared with other users.

After dragging and dropping a template into the workspace, the complete template can be assigned to a group of amplifier channels. This was one of the biggest user requests for enhancements to R1 as it enables significantly quicker creation of R1 project files.

R1 Remote control software v1.8 is available for download from www.dbaudio.com

d&b audiotechnik Website

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Wednesday, October 14, 2009

Eliminating Troublesome Hum & Buzz Created By Electric Guitars

A wide range of solutions to this annoying, frequent problem that happens in the studio and on the stage. Plus, a discussion of power issues, including a sidebar by noted audio consultant Jim Brown.

You’re recording an electric guitar, or amplifying it through a P.A., and there it is: hum! This annoying sound is a common occurrence.

Acoustic guitars fitted with pickups can have the same problem.

Hum is an unwanted 60 Hz tone—50 Hz outside the U.S.—plus harmonics. If the harmonics are especially strong, the hum becomes an edgy buzz.

Let’s take a look at what’s going on and how to fix it. First we need to review how an electric guitar works.

Inside The Electric Guitar
Built into the guitar, under the strings, is a magnetic pickup: a transducer that converts the strings’ vibration into an electrical signal. The pickup is a bar magnet wrapped with thousands of turns of wire, forming a coil.

When the player plucks the steel strings, they vibrate next to the magnet, producing a similar vibration in the magnet’s magnetic field, which in turn causes a varying current in the coil.

Another type of pickup uses a separate magnet under each string. Some pickups have a screw on each magnet’s polepiece to adjust the distance between the polepiece and string, allowing level control of each string.

A humbucking pickup uses two coils wired in series but with opposite polarity so that they cancel common hum fields. One coil is mounted far from the strings.

The high-impedance signal from the pickup coil goes through a simple circuit (Figure 1) and comes out the unbalanced guitar jack.

Components in the circuit are usually connected by single wires. The sleeve (ground) terminal on the jack is connected to the pickup coil, the strings, and the shield around the circuit.

From the guitar jack, the signal travels through a guitar cord: an unbalanced shielded cable.

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At the end of the cable, the signal can go to several destinations: a direct box, a guitar amp, a mixer’s high-Z input, or guitar stomp boxes/processors.

Most acoustic-guitar pickups are piezoelectric types installed under the bridge or saddle. Vibrations of the guitar body flex the pickup, which generates an electrical signal. It’s very high impedance, and often is run through a preamp built into the guitar which reduces the impedance.

Whether the guitar is electric or acoustic, any component in the signal chain is susceptible to picking up hum and buzz, especially because the entire circuit is high-impedance unbalanced.

Hum Sources
Alternating current in a building’s power wiring generates electric and magnetic fields that oscillate at 60 Hz and its harmonics. Hum fields also radiate from lighting circuits and fluorescent lights.

The magnetic fields couple inductively to the guitar wiring. When the magnetic lines of force cut the conductors in the guitar and its pickup, the conductors generate a 60 Hz signal, which is amplified by the mixer or guitar amp.

Also, the power wiring and pickup act as two plates of a capacitor. The varying electric fields from the power wiring couple capacitively to the pickup and guitar wiring.

Another hum source is radio-frequency fields from computers, motors, and TV transmitters (vertical sync, blanking and vertical component video).

This RFI can be detected by the guitar or audio equipment.

A major cause of hum is the ground loop. It is the circuit loop that is formed when two pieces of audio gear are connected to each other through a cable shield and also through the AC safety ground.

If the two chassis are at different ground potential, a 60 Hz current can flow on the cable shield connecting them, causing audible hum.

Figure 2 shows a ground loop. Two equipment chassis (guitar amp and mixer) are connected to two separate safety grounds by their AC cords.

Also, the equipment chassis are connected together by the shield of the audio cable coming from the direct box. The shield and safety-ground wires form a ground loop.

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Causes & Cures
Let’s give several examples of causes and cures of hum related to electric guitars.

Magnetic Hum Fields
As we said, AC in a room’s power wiring generates electric and magnetic fields that oscillate at 60 Hz and its harmonics. When the magnetic lines of force cut the conductors in the guitar and its pickup, the conductors generate a 60 Hz signal, which is amplified by the mixer or guitar amp.

Cure: The amount of hum generated depends on the angle between the pickup coil wires and the magnetic hum field. At certain angles, a lot of the hum goes away. So the player can rotate or move around to find a spot with minimum hum.

Electric Hum Fields
The power wiring and pickup act as two plates of a capacitor. The varying electric fields from the power wiring couple capacitively to the pickup and guitar wiring.

Cure: A grounded Faraday shield. In the guitar body are cutouts that house the electronics, wiring and pickups. These cutouts should be lined with conductive foil (such as copper foil) that is soldered together to form one continuous piece. This shield must be connected to the guitar jack’s ground terminal so that the hum fields are bypassed to ground (ideally, to the mixer chassis).

The guitar cord should also be well shielded. Use only high-quality cords with plugs having a metal jacket, which acts as a shield to the wires inside it.

Some guitar amps are painted on the inside with conductive paint that acts as a shield, but this paint coating can crack when the amp cabinet is jostled, breaking the shield connection. It’s best to run a ground strap between all panels of the amp head cabinet.

RFI (Radio Frequency Interference)
A strong TV signal can be rectified and demodulated by some electronic components or a bad solder joint.

Cure: This is a subject in itself (see the references at the end of this article). But for a quick fix, install ferrite beads and .001 microfarad capacitors on mic inputs. Install RFI chokes on guitar cords or mic cables. Check solder joints.

Ground Loop
Suppose you’re recording a guitar direct, and the guitar is plugged into a guitar amp. The amp and your mixer have 3-prong power cords that connect to the safety ground. The amp is plugged into an AC outlet across the room, and your mixer is plugged into a nearby outlet. When you connect the amp ground to your mixer ground through the mic-cable shield, and monitor the signal, you hear hum.

Chances are that the outlets are fed from different circuit breakers, so the outlets are at different ground voltages. When you plug your amp and mixer into these separated outlets, and connect the equipment together with a mic cable from a direct box, the difference in ground voltages can make a 60-Hz hum current flow between the guitar amp and mixer. That’s a ground loop.

Cure: Flip the ground-lift switch on the direct box to break the loop. Also, it’s a good idea to power the mixer and guitar amp off the same outlet strip. That way, the ground voltage for all the equipment is about the same, so little or no hum current can flow between their chassis. Run a thick extension cord from the mixer’s outlet strip to the guitar amp, and plug the amp into the extension cord.

There still may be a slight voltage difference between components because their power supplies reflect different voltages onto their chassis. A balanced AC power supply can eliminate this problem.

Before you plug in all those power cords, make sure that the sum of the equipment fuse ratings does not exceed the amperage rating for that circuit. In most cases, a single 20-amp breaker will handle a small studio.

Guitar Not Grounded
Suppose you’re recording a guitar with a direct box, and the guitar is NOT plugged into a guitar amp. If the ground is lifted on the direct box, the guitar is not grounded, so you hear a loud buzz.

Or if the shield connection is broken in the guitar cord or mic cable, the guitar is not grounded.

Cure: Flip the ground-lift switch to the grounded position when not using a guitar amp. Check inside the cable connectors to make sure the shield is soldered at both ends. Replace or repair guitar cords that have broken shields.

Player’s Body Not Grounded
When the guitar player touches the strings, does the hum stop? This indicates that the player’s body is acting as one plate of a capacitor.

The capacitance between the body and power wiring adds to the capacitance between the guitar and power wiring, increasing the level of the hum transmitted from the power wiring to the guitar. (Incidentally, the same thing happens if you replace the player’s body with a sheet of aluminum foil).

Cure: Run a wire between a ground point on the guitar and the player’s skin. Figure 3 shows a ground wire (highlighted in yellow) between the guitar-jack ground and the player’s big toe!

This grounds the player’s body, so that it acts as a partial shield for the guitar, rather than a capacitor.

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A body close to the guitar increases hum, and connecting the body to the guitar ground stops the hum. The body is not a ground for the guitar. Rather, the guitar ground is a ground for the body.

So now we know why some heavy-metal guitarists play without a shirt. They’re removing the dielectric between their skin and the guitar.  (Thanks to Crown’s Chris Vice for that insight).

Caution: Do NOT use this ground wire in a concert situation if the guitar is plugged into a guitar amp. There might be a shock hazard if the player touches a mic. That can happen if the mic, which is grounded to the FOH mixer, is at a different ground potential than the guitar amp onstage.

To reduce the potential between mixer and guitar amp, power the mixer through a thick extension cord plugged into the AC distro outlet that the guitar amp is plugged into.

In other words, do NOT make a permanent connection between the player’s skin and the guitar ground in this situation. You might ask the player to keep their hands on the strings whenever possible.

For a technical discussion of body grounding, please see the sidebar that follows this article.

Shock Hazard
This is not about hum, but is an important related issue. In concerts, electric-guitar players can receive a shock when they touch their guitar strings and a mic simultaneously. This occurs when the guitar amp is plugged into an electrical outlet on stage, and the mixing console (to which the mics are grounded) is plugged into a separate outlet across the room.

As stated before, these two power points may be at widely different ground voltages, so a current can flow between the grounded mic housing and the player touching the grounded guitar strings.

Electric guitar shock is especially dangerous when the guitar amp and the console are on different phases of the AC mains.

The cure is to power all instrument amps and audio gear from the same AC distribution outlets. That is, run a heavy extension cord from a stage outlet back to the mixing console (or vice versa).

Plug all the power-cord ground pins into grounded outlets. That way, you prevent shocks and hum at the same time.
 
Using a neon tester or voltmeter, measure the voltage between the electric-guitar strings and the metal grille of the microphones.

If there is a voltage, flip the polarity switch on the amp. Use foam windscreens for additional protection against shocks.

Quick Tips
When you hear hum or buzz from an electric guitar, try these solutions:

• Turn up the guitar’s volume and treble controls so that the guitar signal overrides hum and noise picked up by the guitar cable and guitar amp.
• Ask the guitarist to move around, or rotate, to find a spot in the room where hum disappears.
• Flip the polarity switch on the guitar amp to the lowest-hum position.
• To remove buzzes between guitar notes, try a noise gate.
• If the hum stops when the player touches the guitar strings, ask the player to keep his or her hands on the strings, or run a wire between the player’s skin and a ground point on the guitar (such as the strings or the jack ground.)
• Set the direct-box ground lift switch to the position where you monitor the least hum.
• Replace or repair guitar cords that have broken shields. Use only high-quality cords with metal-jacket plugs.
• Power the guitar amp off the mixer’s outlet strip.
• Use guitars with humbucking pickups, or install modern humbuckers in older guitars.
• Line cutouts in the guitar body with copper foil wired to the guitar-jack ground.
• If you suspect RFI, install ferrite beads, capacitors and chokes. Also see the references below.
• Replace any defective tubes in the guitar amp. If the power-supply filter capacitors in the guitar amp are corroded, replace them. This replacement should be done by an authorized technician.
• Use a quieter amplifier.
• Don’t use a noisy amp. Instead, record the guitar direct, then process its track with a guitar-amp modeling plug-in or processor.
• Don’t use SCR lighting dimmers because they add noise and hash to the AC power. Instead, use multiway incandescent bulbs to vary the studio lighting levels. If you must use a SCR dimmer, rotate its knob to find a position with the least hum (maybe the “off” position!).
• Run the studio off its own breaker, not shared with noisy loads such as air conditioning, power tools, etc. Don’t ground the neutral at more than one point (have an electrician check this). Use an AC line isolation transformer between the AC power and the studio equipment.

If you follow these suggestions, the only buzz you get should be from the guitar player’s solo! Good luck.

Acknowledgement: Many thanks to these Syn Aud Con members for their helpful discussions: Jim Brown, Rick Kamlet, Bob Hagenbach,  Mike Miles, Pat Brown, Steve Roth, and Peter Patrick.

Bruce Bartlett is a microphone engineer (http://www.bartlettmics.com),  an audio journalist, and a recording enginee (http://www.bartlettrecording.com). He is the author of “Practical Recording Techniques 5th Edition” and “Recording Music On Location”.

Go to NEXT PAGE for a related sidebar article about body grounding by Jim Brown.

Sidebar

Technical Discussion About Body Grounding
by Jim Brown, Audio Systems Group
http://audiosystemsgroup.com

The human body is a conductor with relatively high resistivity, and it is a fairly large conductor. This means that when it makes contact with an electrical circuit it can act as an antenna, and it can also act as one “plate” of a capacitor.

The other “plate” of that capacitor might be a noise source like a power line, a noisy electric light, or computer wiring. The noise might be base band (that is, audio frequency), or it might be modulated RF, or it might be both.

The body will react differently to those noise sources depending on what they are—their frequency content, their internal impedance, their orientation with respect to the body, etc. And the body will interact with the circuit of the audio equipment and its wiring.

The various effects of the body in any given circuit will add algebraically—that is, they may be varying degrees of in phase, and they may be in or out of polarity, and they will be at various relative levels with respect to each other, so in any given field condition they will be different.

Some examples of the guitar problem. Let’s say that the body touches the “hot” conductor of a guitar cord plugged into an amplifier. The body can act as both a capacitor, coupling both audio and RF into the equipment, and it can act as an antenna. What’s the difference? 

The word “antenna” implies reception or transmission of an electromagnetic field—that is, the simultaneous existence in space of an electric field and magnetic field at right angles to each other, and in which energy is traded back and forth between electic and magnetic fields.

An antenna has both current flow along it and a potential difference along it that either is caused by the field (reception) or generates the field (transmission). 

So when the body is acting capacitively, it is NOT acting as an antenna, it is not “receiving noise” and coupling it to the equipment. It has become part of the equipment’s wiring and is an element in the equivalent circuit. Now, the body may be acting as a capacitor to one noise source (or in one frequency range) and as an antenna to another, and may be doing so simultaneously! 

Another way that the body can get into the act is by causing current flow on the shield of an unbalanced cable. That current can couple noise in at least two ways. First is the IR (or IZ) drop in the shield, which is added to the signal. Second is via a pin 1 problem.

When the orientation of the guitar is important, there are three mechanisms I can think of that can be at play. First and most obvious is the null that occurs when the circuit that is inductively coupled to a magnetic field is at right angles to that field.

Second is the movement of the body and the guitar so that it is physically closer to the noise source, and thus has a higher capacitance to the noise source.

Third is the directivity of the antenna that it is part of. 

Suggested References

Radio Frequency Susceptibility of Capacitor Microphones (Brown and Josephson). AES Preprint #5720

Common Mode to Differential Mode Conversion in Shielded Twisted Pair Cables (Brown and Whitlock).  AES Preprint #5747

Testing for Radio-Frequency Common Impedance Coupling (the “Pin 1 Problem”) in Microphones and Other Audio Equipment (Brown). AES Preprint #5897

A Novel Method of Testing for Susceptibility of Audio Equipment to Interference from Medium and High Freqeuency Radio Transmitters (Brown).  AES Preprint #5898

Noise Susceptibility in Analog and Digital Signal Processing Systems (Muncy). JAES June 1995

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Wednesday, October 07, 2009

Syn-Aud-Con Presenting Sound Reinforcement For Designers Seminar In Louisville Later This Month

In-depth course designed to emphasize the codependency of room acoustics and sound system design

Syn-Aud-Con will be presenting its Sound Reinforcement for Designers seminar in Louisville, Kentucky, on October 26-29, 2009. 

This course is designed to emphasize the codependency of room acoustics and sound system design, with mantra being “Listen - Measure - Predict”.

View this video for more information.

Lead instructor Pat Brown uses a multimedia approach with interactive exercises which enables him to teach most concepts through demonstrations making learning quicker and much more enjoyable. (Note that attendees need a personal computer to do the exercises.)

Days one and two of the seminar cover the measurement of the acoustic characteristics of enclosed spaces. The course is built around the acquisition and interpretation of the room impulse or RIR, using it to derive all the room metrics of interest to sound system designers. Topics like convolution demonstrate how to integrate listening into the evaluation process.

Days three and four outline and demonstrate an efficient, accurate sound system design process that includes listening, measurement and prediction.

Attendee comments from the previous Sound Reinforcement for Designers seminar:

  “I came here to reinforce and backfill the knowledge I have gained from my many years in the industry. The only disappointing thing is that, for a modest amount of money, anyone can learn in a few days what it has taken me years to figure out!”

  “Syn-Aud-Con has given me the knowledge to do my job even better. I credit the growth of my company to the information that I have learned.”

  “I am amazed how much useable information Pat packs into a 4-day seminar.  I’ll be back and I will bring others from my company.”

  “Syn-Aud-Con seminars changes as the technology changes. With 40 years of audio experience, 10 time Syn-Aud-Con grad, I still added to my knowledge tremendously in these last 4 days.”

Enrollment can be done online (click here), or be calling 800-796-2831. Entry fee is $1,100.

Also upcoming is the Syn-Aud-Con Core Principles of Audio and SR for Technicians, to be held in conjunction with the LDI show in Orlando on November 17-20, 2009. Register for this seminar at www.ldishow.com

Syn-Aud-Con Website

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Thursday, October 01, 2009

Outline Unveils New OpenArray 3D Predictive Software

OpenArray is based on a GL platform and features very fast rendering times from input of data to final design, and it also has the ability to import DXF files

Outline has launched its OpenArray 3D software that helps predict coverage and other esults from both live and installation applications of the company’s products.

Training seminars for the new software were held during the recent PLASA 2009, with a company statement noting the intention of releasing an “alpha” version of software is to gain the experience of system engineers consistently working in the field.

Building on the success of the Outline V.I.P. (Vector Implementation Protocol), the goal is to have engineers provide a “wish list” of additions to the software that would assist them in the future. The software can also be used by sales designers to show clients what can be expected from their purchases.

OpenArray applies to three line array products from Outline -  Butterfly, Mini-COM.P.A.S.S. and Mantas - and by the end of this year, all the company’s subwoofer products will be included. Early next year, all Outline products, including all point-source products, will be covered.

OpenArray is based on a GL platform and features very fast rendering times from input of data to final design, and it also has the ability to import DXF files, thus giving engineers a head start to final deployment of the intended system.

The program went into its first event one week following PLASA at the Guinness 250 celebrations, where an Outline Butterfl’ system was used in HOP HOUSE 13 for performances by Tom Jones and Kasabian.

Britannia Row System Engineer Sergiy Zhtytnikov comments, “The program is really easy to use and accurately predicts the eventual performance of the Outline Butterfly. As a building platform for further enhancements that will appear soon and with the input from actual users the software has no competitors.”

Outline Website

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Monday, September 28, 2009

Electro-Voice Announces Release Of LAPS V2.2A Modeling & Design Program

New workbook pages, overall and specific product support, modeling enhancements, interface improvements and more

Electro-Voice has announced the release of new LAPS V2.2A, the company’s line array modeling and design program.

LAPS is an essential tool for sound system designers who use EV line array loudspeakers.

Using LAPS, a designer can quickly enter an array design and assess its performance. LAPS can model a full sound system, including main arrays, secondary arrays such as delays and front-fills, and subwoofers.

After the final system design is chosen, LAPS provides full rigging specifications. LAPS includes comprehensive checking of rigging working loads, and will not predict performance of rigs which it considers to be unsafe.

The new version includes a special page for predicting coverage of subwoofer arrays. Proper use of this page will in many cases result in significant bass coverage and quality improvement at little or no additional cost.

LAPS supports all EV line array products and related subwoofer models.

The new version is designated LAPS 2.2A, and it is available for free download at http://www.electrovoice.com/documents/ev/LAPS%202.2A%20Installer.zip

LAPS 2.2A includes the following specific changes:

New Workbook Pages
· Arrays Bass page, for predicting horizontal coverage of subwoofer arrays.
· Air Loss page, for predicting sound attenuation in air over large distances.
· Cable Loss page, for modeling the resistive and capacitive effects of loudspeaker cable.
· Supported Products page, which lists all products supported by LAPS 2.2A.

New Product Support
XLC Family
· XLC-907DVX Full-range loudspeaker, 90-degree horizontal coverage.
· CBEAM (new beam) in reversed orientation.
· XLCbeamF (old XLC tilt angle extender) extending to the front.
· XLCbeamR (old XLC tilt angle extender) extending to the rear.
· dB Sound XLC grid and extender bar.

XLVC Family
· XLD-291 Full-range loudspeaker, 90-degree horizontal coverage.
· XLE-291 Full-range loudspeaker, 90-degree horizontal coverage.
· XCS312 3x12-degree cardioid subwoofer.
· CBEAM (new beam) in reversed orientation.
· AGSE XS-212 to XLE adapter grid.

X-Line Family
· XVLSred Full-range loudspeaker, 90-degree horizontal coverage, high rig load version.
· dB Sound X-line grid and extender bar.

EVA Family
· EVA loudspeaker line (four models and three grids).

Modeling Enhancements
· Maximum main-to-delay distance is increased to 412 ft (125m).
· Coverage prediction curves are now more accurate above 6000 Hz.
· Array equalization predictions now include realistic target frequency response curves.
· The polar response display is now more usable for large array tilt angles.

User Interface Improvements
· In the array specifications, it is now possible insert boxes without re-entering existing ones.
· General appearance has been upgraded.
· Venue and array printouts now deal correctly with hidden data.
· There are now pop-up in-progress message boxes for saving files and generating reports.
· Optional dropdown lists have been implemented for entry of box drive level.

Rigging Enhancements
· Rig report formatting and pagination has been improved.
· Trim height and rig bottom height now appear on rig report.
· Rig report display of diagnostic messages is improved.

Other Enhancements and Fixes
· Pullback points for X-Line arrays are now handled better.
· For uptilted X-Line arrays, LAPS now warns if front chain links will collapse.
· The rig top reference point has been changed to top front corner of topmost element.

Installations & Configuration Improvements
· Changed internal relationships to allow multiple versions of LAPS II to coexist on same machine more easily.
· Windows Vista is officially supported.

Help System Upgrades
· Updated and enhanced help information generally.

LAPS File Load/Save Enhancements
· Made LAPS able to load .EVA files written by EV’s EVADA modeling program.
· Improved Load/Save handling of international date formats. Previously, LAPS sometimes failed to load .laps files from foreign countries.

LAPS is a Microsoft Excel-based application that runs in Windows environments. It requires Excel 2000 or later and Microsoft Windows 2000 or later.

Electro-Voice Website

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Russ Berger To Provide Surround Sound Presentation At Upcoming AES Show In New York

Focus will be monitoring surround in acoustically small spaces

Russ Berger, president of Russ Berger Design Group (RBDG), will be sharing his expertise in small room acoustics as part of the Surround Sound Live Seven symposium taking place on October 8, 2009, at the Manhattan Center in New York, one of several special events to be held during the Audio Engineering Society (AES) 2009 convention.

Surround Sound Live Seven brings together the authorities in surround sound for a candid and informative discussion with today’s audio professionals about issues relating to the capture, broadcast and monitoring of surround.

Berger will be helping to kick off the full-day symposium with his talk entitled “Surround Monitoring Challenges in Acoustically Small Spaces,” where he will highlight the practical criteria and solutions to help overcome the limitations of monitoring surround sound in acoustically small spaces not originally intended for that purpose. 

“For live events, finding a critical listening space for use in monitoring and producing surround sound product is nearly impossible,” explains Berger. “Decisions are made not only about source material content and veracity, but also about how best to process and present the multichannel surround event in a manner that translates the aural experience of a live event to a wider audience. 

“From our work on various surround spaces for broadcast and music recording, we have found there are methods that optimize an oftentimes less-than-ideal acoustical environment.” 

Berger has been involved with the acoustical design development of broadcast and recording studios, specifically the behavior of sound in acoustically small spaces and the properties of acoustically coupled spaces.

Projects include Sony Music Entertainment, ABC-TV, NBC-TV, CBS-TV, more than 80 public radio facilities (including National Public Radio in D.C., WBUR-FM/Car Talk, WPLN/Nashville Public Radio and Southern California Public Radio), NFL Films, Pro Football Hall of Fame, Sweetwater, recording studios for Michael Bolton, Mariah Carey, Steve Miller and Whitney Houston, Paragon Studios, BiCoastal Music, MasterMix, TNN, Lakewood Church and recording/broadcast facilities for RadioShack and JCPenney.

For more information about Surround Sound Live Seven go to www.aes.org.

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Tuesday, September 08, 2009

Smaart Training Class Confirmed For Houston In Late October

Smaart Training Class is open to all interested persons - no prior measurement/system alignment experience required

Rational Acoustics has confirmed a new Smaart Training Class in Houston, Texas, slated for October 27 - 29 at LD Systems.

The class will follow the standard three-day format and will be taught by Smaart Guru Jamie Anderson of Rational Acoustics. 

Days 1 and 2 cover:
Fundamentals
Applications

Day 3 will be the working practicum/advanced class. 

Smaart Training Class is open to all interested persons - no prior measurement/system alignment experience required. Note, however, that the course does assume a working knowledge of professional sound system engineering practices and basic audio fundamentals.

An informational sheet on the class is available on the Rational Acoustics class schedule page. 

Registration for this class is now open on the Rational Acoustics training registration page, or by calling Barb Mattson at 860-928-7828.

Rational Acoustics Website

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Tuesday, August 25, 2009

Rational Acoustics Appoints SonoTribe As Official Smaart Instructors

Mexico City audio consultants to offer Spanish language Smaart training classes in Mexico and Latin America

Rational Acoustics has appointed SonoTribe Audio Consultants as the official providers of Spanish language Smaart Training Classes for Mexico and Latin America.

Founded in 2007 by Oscar Gamas and Salvador Castaneda, SonoTribe provides system design and analysis services for a wide variety of clients across Latin America in addition to teaching classes in audio theory, system design and implementation. 

Prior to founding SonoTribe, Gamas and Castaneda were employed by Meyer Sound in the design and education departments respectively. Together they possess over 25 years of experience in professional audio design, development and instruction in Mexico and worldwide.

“Our goal has always been to provide the highest-quality technical and educational support for the Smaart user-base.” notes Jamie Anderson, Rational Acoustics CEO and Smaart Instructor.  “Providing regional, local-language training classes is an excellent way to accomplish this goal and I can’t think of anyone better equipped to lead this effort in Latin America than SonoTribe.  Oscar and Salvador are incredibly talented instructors and we have every confidence that they will provide a phenomenal experience for those attending their classes.”

Adds Gamas, Sonotribe CEO and Consultant, “Our objective is to expand professional audio education services in México and Latin America and to provide our students with the best information and tools required for their success. Smaart is one of the most powerful solutions in the industry and now under the management of a remarkable company like Rational Acoustics the further development of Smaart software and educational programs is guaranteed.

“SonoTribe and Rational share very similar business philosophies and we are sure that this will be a successful and strategic business partnership to support regional requirements of the Latino market.”

After an inaugural Mexico City Smaart Class in July, 2009, SonoTribe is now actively scheduling classes in other locations in Latin America throughout the fall and winter. 

The next Spanish language classes will be in Bogota, Columbia on September 7-9 and in Caracas, Venezuela on September 14-16.  Interested attendees can contact SonoTribe directly at .(JavaScript must be enabled to view this email address) for information on attending a class.

Rational Acoustics Website

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Friday, August 21, 2009

Meyer Sound To Host Audio Seminars In Mexico, U.S., Canada, Germany, Spain, Italy, & Netherlands

Seminars will cover core audio principles encompassing sound system design and alignment, line array theory, equalization, delay, and acoustical prediction

Meyer Sound has announced its international education schedule for August to October 2009, with seminars to be held in Mexico City; Toronto; Los Angeles; Montabaur, Germany; Seville and Madrid, Spain; Duivendrecht, Netherlands; and Osimo Stazione, Italy.

Some seminars will be taught in the local languages, including Spanish and German, and others will be interpreted in real time by qualified translators. 

The upcoming seminars will cover core audio principles encompassing sound system design and alignment, line array theory, equalization, delay, and acoustical prediction with MAPP Online Pro.

Other courses explore system measurement and optimization, FFT analysis, filters and phase relationships at an advanced level with practical reference to the use of Meyer Sound’s SIM 3 audio analyzer.

Led by an instructor force of active sound reinforcement specialists, similar sessions were hosted in recent months in Nigeria, Lithuania, France, Spain, Chile, Australia, Hong Kong, and the U.S..

Large groups of industry professionals and students took advantage of the opportunity to further their understanding of audio theory and engage in an open forum discussing the challenges and solutions in touring and other live event applications.

“Meyer Sound seminars can bring even the inexperienced users up to speed on the most important subjects in audio,” says Arturas Krasauskas, an attendee of the Line Array Design and Application seminar in Lithuania.

“My staff is now doing a better job at system design and alignment. In the long run, I think these seminars put us in a better position to achieve customer satisfaction and improve our business. It is also a great way to familiarize with Meyer systems and really hear how good properly tuned systems can sound.”

Since its first seminar in 1984, Meyer Sound has expanded the scale of its education program, including a growing international instructor base, and actively ventures new ways to meet the increasing demand for high-level training in sound reinforcement.

In addition to the mentioned curriculum, the program includes courses on audio show control with Matrix3 and the popular Mixing Workshop by veteran audio engineer Buford Jones, as well as the recently debuted Practical Training: Methods and Applications, a new seminar on the techniques to set up systems quickly and efficiently.

Meyer Sound also partners with universities and industry organizations in offering training and scholarship opportunities in conjunction with institutions such as Spain’s ESAMA.

“The international success of our program tells us that members of the audio community appreciate the importance in continuing to hone their skills and knowledge of the precision tools available to them,” says Gavin Canaan, education programs manager at Meyer Sound.

“And since the emphasis of our seminars is on science, not on marketing, and on the real world outside the classroom, we believe we are providing the industry with valuable professional development that our participants can apply in their day-to-day work.”

To find out more about Meyer Sound’s education program and register for an upcoming seminar, go to http://www.meyersound.com/education/

Meyer Sound Website

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Thursday, July 30, 2009

White Spaces To Green Touring: AES Announces Live Sound Events For 127th Convention In New York

Wireless advice, loudspeaker design, microphone dressing, and much more

Reflecting the exponential growth of live sound as an increasingly significant component in the health of the pro audio industry, 127th AES Convention Live Sound Co-Chairs Henry Cohen and Mac Kerr have devised a broad program of critical events designed to keep the “Road Warriors” fully on top of their game. 

Presentations run the gamut from an assessment of the White Spaces issue to innovations for meeting the increasingly urgent need for compliance with the Green Movement. 

Live sound programs include:

The Greening of the Band - Green Touring Solutions for the Live Engineer: AES Education Committee Vice Chair John Krivit – The Dave Matthews Band, John Mayer, Guster, Jack Johnson, Barenaked Ladies, Sheryl Crow and many other top artists are incorporating practices to reduce the environmental impact of touring.

White Spaces & TVBD Update - Chris Lyons, Joe Ciaudelli, Edger Reihl. While the DTV conversion is basically complete, the impact of this and related FCC decisions remains of great concern to wireless microphone users. Exactly what is the status of the 700MHz spectrum and equipment? What types of interference will the proposed Television Band Devices create and what are the proposed methods of mitigation? Will there be an FCC crack-down on unlicensed microphone use?

Practical Advice for Wireless System Users - James Stoffo. From houses of worship to wedding bands and community theater productions, hundreds of thousands of small to medium-sized wireless microphone systems and IEMs are in daily use around the country. Unlike the Super Bowl or Grammy Awards, these smaller systems rarely have benefit of dedicated technicians, sophisticated frequency coordination, or even basic attention to system setup. This panel will address the elements of successful component selection, designing systems, and setting them up to minimize potential interference and maximize performance.

Microphone Dressing - Mary McGregor. Fitting actors with wireless microphone elements and wireless transmitters is a detail-oriented art form. Challenges range from ensuring the actor is comfortable and the electronics safe, to providing optimal sound with minimal muddle while maintaining the visual illusion.  One of the most widely recognized artisans in this field will provide hands-on demonstrations of basic techniques and share some time tested “tricks of the trade.” 
                                                                                 
State of the Art Loudspeaker Design for Live Sound – Tom Young. The loudspeakers we employ today are vastly improved over what existed back in the ‘60s when rock-n-roll pushed the limits of existing technology. Along with advances in design and fabrication, system engineering, ergonomics and rigging, have come various methods to improve overall performance. Many of these advances are directly related to the use of computers as a design tool. This presentation will clarify the capabilities of modern day loudspeakers/systems and consider where they need to go.

Microphone Selection and Techniques For Live Sound - Dean Giavaras. While countless factors contribute to a good sounding live event, microphone selection, placement, and use can make the difference between a pleasant event and a sonic nightmare. A panel of experts from mic manufacturers and sound reinforcement providers will discuss tips and tricks for getting the job done at the start of the signal path. Conventional and unconventional techniques will be discussed, and cautionary stories from the trenches will be shared.

Other live sound events programs include:
• Sound System Design and Installation Considerations for Churches and HOWs: Bill Thrasher
• Exploring the Low End – Measuring and Aligning Subwoofers: Sam Berkow
• Ten Things to Get Right with Sound Reinforcement Systems – Peter Mapp
• Automixing for Live Sound – Michael “Bink” Knowles
• Networking Digital Audio In Live Sound – Jim Risgin
• AC Power, Grounding & Shielding - Bruce Olson with Bill Whitlock
• Innovations in Live Sound – Ken Lopez

“We are extremely pleased with this year’s Live Sound Events schedule,” remarked AES Executive Director Roger Furness. “It is comprehensive, authoritative and boasts a particularly strong line up of high-level presenters. The recommendations made by Co-Chairs Henry Cohen and Mac Kerr echo the invaluable insights they have acquired during their own years on the road. The program reflects an enormous amount of labor intensive work for which we are most appreciative.”

The 127th AES Convention will be held at the Javits Center in New York City, October 9-12, 2009.

AES Website

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Thursday, July 23, 2009

On The Road With The Boss; The System & Crew For Bruce Springsteen’s Latest Tour

Appearing in Europe and North America, the system approach is virtually identical on both sides of the Atlantic

Touring in support of the new album, “Working On A Dream,”  Bruce Springsteen and the venerable E Street Band are once again utilizing the services and systems provided by Audio Analysts.

With North American dates throughout the spring and late summer and European dates in between, the approach is virtually identical on both sides of the Atlantic.

Front of House Engineer John Cooper worked with Albert Lecesse, Co-Owner and President of Engineering for Audio Analysts, in putting together the main system for the tour.

“Albert and I are constantly in discussions about how best to proceed with the PA touring plan,” explained Cooper. “We opted for JBL VerTec because it’s versatile – a customizable PA solution to tour with.”

Cooper mixes on a Digidesign Profile digital console, which he chooses over anything he’s tried. “I’m fortunate to be able to use whatever I want to use. At this point, the Profile has proven to be very reliable.”

He explains that he likes the ease of use, paired with the audio quality of the console. “I’ve become attached to using plug-ins for insert processing. Currently I am using a number of McDSP plug-ins, as well as the Waves Live bundle. Both product lines work great.”

For vocal microphones, Cooper made a change over previous tours, selecting Shure SM58 on all vocals. “I’m a bit of an old-school guy when it comes to mic selection. I have run the gauntlet of boutique mics, and I always come back to the industry standard. They just work!”

For background vocals, he time offsets them as a stereo group to give the vocals a stereo image – “A great place to sit around Bruce’s vocals. As far as Bruce is concerned, it’s pretty straight-ahead, with the Waves Renaissance compressor plug-in paired with a McDSP MC 2000 plug-in to put him on top.” Cooper processes background vocals with the Profile onbaord facilities.

Front of House System Engineer Brett Dicus details that the VerTec Series line arrays are powered by Crown I-Tech and I-Tech HD Series amplifiers under the control and monitoring of Harman HiQnet System Architect technology.

“In any given show,” he elaborates, “We have 10 hangs of PA, primarily using the VT4889 and VT4880 for the main hangs and then VT4888 for the additional outfill/delay and rear hangs, with VT4887 for frontfill surrounding the stage.”

The VerTec line arrays are easy for the crew to set up, taking approximately six hours to get fully into place. “Our setup in the field is also made very easy by the system design by Albert and Audio Analysts. The packaging, the cabling, and the prep is a very important part of the system,” Dicus adds.

By having John “Boo” Bruey also on the tour to serve as System Engineer, Dicus is free to handle the archival/recording of each show as well as more IT-related tasks. With the tight schedule, having two system techs doubles what can be accomplished within the same timespan.

Monitor world is a complex affair, with stage left and stage right setups helmed by Monitor Engineers Monty Carlo and Troy Milner,  respectively. Both positions are outfitted with a Yamaha PM1D digital mixing console, a mix of wedges, and in-ear systems.

Monitor wedges are Audio Analysts SLP (super low profile) 1 x 12-inch, 2 x 12 -inch, and 1 x 15-inch models, along with JBL VT4888 (four per side) for stage fill. On the IEM side, there are Sennheiser 300G2 systems with Westone ES2 and Ultimate Ears UE7 earphones. Aphex HeadPods with headphone amplifiers are also deployed for two musicians.

About 60 RF channels are in use during each show, with Sennheiser G2 systems and Shure UR Series for instruments (G1, J5, L3 ranges). The crew reports no RF difficulties to this point, with additional setup and coordination assistance available with the Shure Wireless Workbench and Professional Wireless IAS software with a TTIRF scanner.

Jeff MacKay is Managing Editor of Live Sound International.

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Be sure to take PSW’s Photo Gallery Tour of the crew and system out in support of Springsteen & Company.

Front of House PA
JBL VerTec VT4889 Line Array Modules (18 deep front, 14 deep sides)
JBL VerTec VT4880 Subwoofers (12 flown per side, 4 stacked per side)
JBL Vertec 4888 Line Array Modules (Rear fill - 4 arrays, 12 deep)

Front of House Amplification
Crown I-Tech IT-5000HD
Crown I-Tech IT-12000HD

Front of House Console
Digidesign Profile 96-channel with Pro Tools recording rack

Front of House Outboard
Dolby Lake Processors
Tascam CD burners

Stage Monitor Consoles
2 x Yamaha PM1D v2 96 in/64 out

Stage Monitoring
JBL VerTec VT4888 (4 per side, stage fill)
Audio Analysts 1 x 12-inch
Audio Analysts 1 x 15-inch
Audio Analysts 2 x 12-inch

Monitor Amplification
Crown I-Tech IT-400

Personal Monitor Systems
Westone ES2 earphones
Ultimate Ears UE7 earphones
Sennheiser G2 IEM systems

Wired Microphones
Shure KSM32
Shure KSM137
Shure Beta 98
Shure Beta 91
Shure SM57
Shure VP88

Wireless Microphones
Shure UR
Sennheiser G2

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Wednesday, July 22, 2009

Yamaha Commercial Audio To Present Digital Sound Reinforcement Class In Kentucky

The class provides complete audio system design and networking solutions and will focus on set up techniques ranging from microphones to large-scale loudspeakers

The Yamaha Commercial Audio Training Seminars (YCATS) group will hold a two-day Digital Sound Reinforcement (DSR) 101 class on Tuesday and Wednesday, August 11 and 12 in Covington, Kentucky.

Sponsored in part by Shure, the class provides complete audio system design and networking solutions and will focus on set up techniques ranging from microphones to large-scale loudspeakers. Topics include: gain construction, attenuator settings, and SPL estimation.

Yamaha DSR101 course components include fundamental concepts: dBSPL, dBu. dBV, calculations, and level management; cables for various signal formats, word clock distribution; microphone techniques: microphone characteristics, how to select a microphone, digital technology; wireless technology: how to maximize the benefits of wireless systems and managing frequencies; digital vs. analog systems; speakers and processors: designing output systems and level management for system protection.

The two-day course is designed to assist audio engineers, system designers, system techs, audio operators in houses of worship, live sound venue/tour engineers, obtain a better understanding of available solutions.

In order to maximize the Digital SR System 101 training seminar and demo experience, attendance for the seminar is limited. There is no cost for attending and breakfast and lunch will provided. Travel and accommodations are the responsibility of the attendee.

Attendees must reserve their spot by August 7. Seminar details can be found at http://www.yamaha.com/ycats/digisr101/description.asp.

For Registration visit: https://www.yamaha.com/ycats/digisr101/registration.asp  The location and class time will be provided upon registering on line.

Yamaha Commercial Audio Website

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Monday, July 06, 2009

Pro Media/UltraSound Chooses Meyer Sound To Support San Francisco Opera Simulcast At AT&T Ballpark

The sound reinforcement system included 28 MILO line array loudspeakers (two arrays of 14 each), along with eight legacy MSL-2A loudspeakers, eight M3D-Sub directional subwoofers, a Galileo loudspeaker management system, and a SIM 3 audio analyzer for system tuning

“Opera at the Ballpark” returned to San Francisco at AT&T Park last month with a free simulcast of Tosca, setting a new ballpark attendance record for the San Francisco Opera with an estimated 27,000 viewers.

For its fourth simulcast, Bay Area rental house Pro Media/UltraSound turned to Meyer Sound loudspeakers, subwoofers and processing to meet the unique task of presenting a live performance from the stage of the War Memorial Opera House to a crowded sports stadium.

David Bowers of Pro Media/UltraSound stated that the decision to go with Meyer Sound was a relatively simple one: “It’s just the way it is. There’s nothing that works better for opera. It’s absolutely, purely linear. It predicts well, and there’s very low distortion.”

The sound reinforcement system included 28 MILO line array loudspeakers (two arrays of 14 each), along with eight legacy MSL-2A loudspeakers, eight M3D-Sub directional subwoofers, a Galileo loudspeaker management system, and a SIM 3 audio analyzer for system tuning.

“We showed up at 1 (on show day) to load in,” said Bowers on the day of the show. “By 7 pm that night, we were doing our first pass for optimization. It was so expeditious. That’s a credit to self-powered systems—they’re very easy to rig.”

Bowers, who has a wealth of experience when it comes to mixing opera and classical music, recognizes the benefits of choosing Meyer Sound for this project. “We love the M3D subwoofer for the opera,” said Bowers. “It’s the most musical-sounding sub for opera and classical music. We want the sound system to be invisible, even if the stage is 400 feet away from the audience. M3D-Sub, MILO, and Meyer allow us to do that.”

“I am overwhelmed and gratified by the Bay Area’s enthusiastic response to our third simulcast of live opera to AT&T Park,” commented Opera General Director David Gockley.  “Our presentation of Tosca onto the scoreboard in high def transmission was fantastic and was matched by the superior sound quality under the careful guidance of Meyer Sound , ProMedia Ultra Sound and the SF Opera Sound Dept.  From the outfield and the infield to stadium seating, more than 27,000 guests enjoyed this growing San Francisco tradition of Opera at the Ballpark. It truly was a grand night for singing.”

Meyer Sound Website

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Monday, June 29, 2009

Tuning A System At The Ultra Music Festival With EASERA SysTune (And “Dr. Bassenstein”)

A look at the tuning process - for subwoofers in particular - for an annual festival featuring performances by some of the top DJs in the world (Also be sure to check out "An Overview Of EASERA SysTune" by Charlie Hughes)

The Ultra Music Festival takes place annually in the spring in Miami, concluding a week-long Winter Music Conference attended by electronic music artists, producers, and fans from around the world.

Ultra Music features performances by some of the world’s top DJs, with the festival site crammed full of PA systems from various vendors.

This year I returned at the invitation of UMF Audio Chief Terry MacNeil (“Dr. Bassenstein”) to perform alignment work on systems as a couple of the stages.

In particular, a lot of attention gets paid to the subwoofers - there’s a lot of content below 50 Hz, and the subs need to be as “right” as possible. Unfortunately, festival scheduling issues restricted my efforts to a fairly tight window..

Advance Work
The PA vendor for both the main stage and Bayfront stage was Beach Sound (www.beachsound.com). The main stage would be equipped with 32 d&b audiotechnik J8/J12, 16 d&b J-Sub subwoofers flown along the J8s, and 24 d&b B2 subs in three high CSA stacks, four stacks per side.

In addition, BASSMAXX supplied 16 beta test subwoofers currently given the model designation SP218 or the “Dub-ill 18.” These are double 18-inch direct radiating vented subs. The challenge would be integrating the centered BASSMAXX subs with the B2s flanking them.

Issues:
1. Physical separation between sources, setting us up for interference problems.
2. Different models of subwoofers, setting us up for potential phase (frequency specific delay) issues.
3. Subjective sound quality difference between the two models of subs.

BASSMAXX chief David Lee supplied some phase data for the new sub, and it appeared the phase response wrapped smoothly enough (for example, no abrupt variations in the operating band) that there would be a good chance of acceptable integration.

I contacted Neil Rosenstock, Beach Sound System Engineer, about “the plan” and we began to coordinate a rational approach to getting as much of the work done in advance as possible. The initial plan was to be able to use incremental delay taps for the center cluster. However, the stacking arrangement proved to be advantageous, allowing us to fill the center without beaming as much as if it had been an eight wide/two high system.

Neil came up with a CSA stacking plan that would steer the B2s away from the center a bit, supplying an ArrayCalc solution that did just this.

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As we shall see later, a bit less steering got it within acceptable limits out in the overlap areas.

EASERA SysTune
I decided to give SysTune a test drive for the event, so three weeks prior I downloaded the evaluation and worked my way through the tutorial.

The tutorial is very good, and anyone who understands measurement issues and has used a dual channel FFT analyzer before will be in a good starting place after completion. Particularly, I wanted to use SysTune’s ASIO multichannel capacity.

My multichannel measurement rig currently consists of:
PreSonus Firestudio Project 8 channel FireWire mixer
Small custom 2 space rack, with power strip
4 SIA RTA-420 microphones
Josephson C-535 microphone
4 Manfrotto collapsible microphone stands
7 microphone cables, of 50- and 100-foot lengths
Assortment of cables and turnarounds
WiFi router and IBM X41 tablet PC for remote access into measurement computer

This all rides in a Pelican 1650 case. Because I was arriving on tuning day and didn’t want take a chance on an airline losing it, I used FedEx to deliver it to Beach Sound.

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On With It
After some travel delays, I finally arrived at the festival site around 2 p.m. the day prior to the kick-off of the festival. Due to the schedule restrictions noted above, we weren’t able to make “big noise” until 5 p.m.

I got my multichannel rig set up in less than 10 minutes, but the amp rack for the center BASSMAXX subs had not yet arrived. Rosenstock and I walked the field briefly to see what the steered B2’s were doing, and sure enough there was somewhat of a hole carved out in the center.

In the meantime, I took a small measurement rig to the Bayfront Stage, with this rig consisting of the Dell Inspiron 4150 measurement notebook, an M-Audio MobilePre USB, and the Josephson microphone.

The PA consisted of flown d&b J8/J12 (no J-Subs), and 16 BASSMAXX X2C “Deuces” lined up across the stage.

The sub line was long enough to get some serious pattern control outside the edges, and in fact this is what we wanted in order to avoid spill to the extent possible in other areas.

With this many systems, and this much sub content, any control available is gladly exercised.

The quick alignment job consisted of sub alignment via phase trace to the flown PA, some quick EQ on the PA (the d&b systems seem to never need much, particularly outdoors), and some sweetening by ear on the subs.

A 6 dB low shelf boost on the Deuces suggested by Dr. Bassenstein was applied, and after a little experimentation using Lake Contour controller, we liked what we heard.

The lack of a tech day meant that forklifts and lulls were constantly working everywhere. At one point I had my back turned to the stage, trying to figure out what was going with this crazy transfer function that could not possibly be right, noise running at a fairly high level.

I turned around and a lull had pulled up next to my measurement microphone, completely contaminating the measurement. It turned out this would be the rule, and not the exception, the rest of the day.

Back To The Main Stage
The amp rack for the center BASSMAXX array eventually arrived and we began. The first set of measurements was mostly on axis with the house right portion.

Using the multi-channel capacity of SysTune, it was quite easy to quickly switch between measurement microphones. Additionally, the easy management of overlays helped me move quickly between tasks.

Since we were short on time, our efforts were concentrated on integrating the center BASSMAXX stack with the spread CSA B2 stacks residing under each side of the PA.

The Figure 1 screen shot shows a measurement overlay taken on site, and reloaded back into SysTune after the fact. The measurements were done with a 64 kHz FFT size, yielding 1.46 Hz frequency resolution.

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This resolution is OK for low frequency work but is too fine for HF work.

Also, the delay offset and measurement levels do not affect the loaded overlays. Savvy users will notice the “zero” delay time – this is because we’re looking at reloaded overlays, not live measurements.

The previous measurement is of only the steered-out B2s, taken from the center of the audience area. The next step is to add the BASSMAXX cluster in the center as seen in Figure 2.

Keep in mind here we have steered the outside B2s away from the center a bit to allow the BASSMAXX boxes to have some of their own space. We have not adjusted the gain on the center sub array at this point.

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Listening tests validated our original notion that if they sound different enough, we should avoid overlap if possible, and these subs definitely do not sound “the same” - whatever that means.

So in our center position, the B2s began to drop off just below 40 Hz as seen in the screen shot. The BASSMAXX subs remained quite flat to 30 Hz.

The phase angles are mostly matched, if not perfectly timed at the measurement position.

However, after listening we decided to treat the BASSMAXX array separately and Rosenstock inserted an 18 dB octave low pass filter at 47 Hz (after some experimentation), and this yielded the above result.

Note the phase response in the areas of interest, not varying more than 90 degrees between the two systems. We played with some delay times but this yielded no appreciable difference, so we left it “as is.”

Walking the field revealed a few but mostly insignificant nulls, certainly far fewer than a traditional left and right sub arrangement. This is the measured response of the summed systems, in the center of the audience area, as shown in Figure 3.

A little subsequent tweaking sweetened the very bottom end, of which a few of the DJs (and particularly The Prodigy) tested to the limits.

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Overall, everyone was satisfied with the arrangement.

DJ Tiesto’s Production Manager walked the field and commented on a few minor nulls but said “it is powerful enough.” Rosenstock and I walked around a bit and decided to pull the CSA stacks back in toward the center and it did help.

The Prodigy’s Front of House Engineer Jon Burto, had a few comments regarding the implementation and offered some suggestions. He had an interesting night mixing, between artists directly in front of the PA, an artist with a somewhat weak voice that evening, and the unenviable position of not hearing the same bass response his audience was hearing.

Final notes and observations:
• The mix platform was located on an SL-100 mobile stage. In fact, the bass response “up there” was significantly different than what the audience experienced.
• The RTA-420 microphone delivered equivalent performance to an Earthworks M30 for subwoofer work. The sensitivity is different but I detected no real difference in either magnitude or phase.
• The BASSMAXX array was quite powerful. We inspected it a few times during the show and experienced blurred vision and difficulty communicating. Yes, I had my -25 dB earplugs in.
• Behind the barricade, there was complete (and I mean COMPLETE) cancellation for a small distance between the stacks.
• While mixing sub models is generally not recommended, with some careful planning and overall awareness of the issues, if you have the tools you can make it work (usually…).
• Yes, it was “powerful enough.”

Doug Fowler is Director of Audio Engineering Services for Logic Systems Sound and Lighting in St. Louis.

(Also be sure to check out “An Overview Of EASERA SysTune” by Charlie Hughes)

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BSS Audio Introduces AEC Input Card To Soundweb London Family of Digital Signal Processors

Configuration, control and monitoring of the AEC input card is provided by Harman HiQnet London Architect

At 2009 InfoComm in Orlando, BSS Audio introduced the Acoustic Echo Cancellation (AEC) card to its Soundweb London family of digital signal processors.

Designed specifically for the Soundweb London BLU-800, BLU-320, BLU-160 and BLU-120 devices, the Soundweb London AEC input card complements the analog and digital, input and output card options, extending the reach of Soundweb London into teleconferencing applications.

The Soundweb London AEC input card utilizes a proprietary algorithm developed by Wavemakers, a Harman International company and specialist in voice optimization software for automotive and communications applications. The algorithm used on the Soundweb London AEC input card has been specifically designed to meet the needs of installed sound applications and the high expectations of today’s teleconference participant.

The Soundweb London AEC input card features four microphone / line level channels with Acoustic Echo Cancellation per channel. A direct microphone feed is made available to facilitate local sound reinforcement. Configuration, control and monitoring of the AEC input card is provided by Harman HiQnet London Architect.

The Soundweb London BLU-800, BLU-320, BLU-160 and BLU-120 devices joined the existing BLU-80, BLU-32 and BLU-16 devices. The newer devices feature a low-latency, fault-tolerant digital audio bus of 256 channels.

In addition to providing a backbone for the transportation of multiple channels, this bus also facilitates the creation of large, fault-tolerant, centralized matrices containing multiple devices. The BLU-800 and BLU-160 models feature configurable DSP, boasting four times the processing capability of BLU-80 and BLU-16 devices.

The BLU-BOB cost-effective break-out box was also added to the family and represents an inexpensive solution to increasing the number of outputs in a Harman HiQnet networked system.

Since its advent, the BSS Audio Soundweb London family has drawn acclaim for its flexibility and intuitive operation. All eight models offer pristine audio quality, advanced A/D and D/A conversion, 96 kHz capability, drag-and-drop system design with CobraNet bundle assignment, Ethernet-based control, an extensive range of control options from simple to sophisticated, easy expansion or reconfiguration of system hardware in the field and an upgrade path for future enhancements.

Each of the eight different Soundweb London devices offer a different mix of signal processing, CobraNet and digital audio bus functionality, making a Soundweb London system the perfect match for any application.

BSS Audio Website
Harman Professional Website

(Be sure to visit PSW’s 2009 InfoComm New Product Gallery.)

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