Article

Monday, May 14, 2012

Fill-osophy 101: Using Fill Loudspeakers To Optimize Coverage, Localization & More

A well-designed main/primary loudspeaker system is expected to provide clear and intelligible sound to the entire audience.

Over the years our industry has benefitted from a steadily growing selection of types and configurations of primary loudspeakers to do this; they possess the specific characteristics (performance and physical) that we need in our base inventory, or for a given project, or to solve a specific problem.

Many of these varied types of products are also available over a wide range of size, quality and price options.

Yet more often than not when we deploy primary loudspeaker devices - singly or in all manner of arrays - there still may be holes in the coverage. This occurs for a number of valid reasons:

• Not enough primary loudspeakers for the specific venue;
• The need to avoid aiming primary loudspeakers to cover all of the front seats (at high frequencies) due to the low and mid frequency energy that ends up on stage or in the orchestra pit;
• Box seats that are located close to the stage (proscenium) that can’t be covered by the primary loudspeakers due to their close proximity;
• Architectural elements, including balconies and columns, that may shadow the sound projected from the primary system;
• The depth of seating (indoors and outdoors) can require additional devices, located further into the seating area, to “kick it up” a bit; 
• The need to provide sound to contained areas to the sides of the stage.

And the list goes on.

Coverage is an absolutely critical performance characteristic, at the very heart of what we must provide. The challenge to be met is supplying the correct number of fill loudspeakers that individually, or collectively, insure the coverage needed to fill the existing hole(s) and to ensure minimal overlap into the coverage that is already provided.

Few low-profile fill loudspeakers exhibit narrow coverage patterns across a wide frequency range and are also inherently restricted in acoustic output. Therefore their throw is restricted, although this actually may be of benefit in some cases.

Further, in many cases, clear and intelligible coverage is not our only concern. There often is an equally important need to provide or maintain localization to the source, and for as many seats as possible.

Localization is defined as the aural perception that the sound is coming from the apparent point of origin (where we locate the source visually, usually on the stage or platform) or, in larger venues and specifically for those seated further away, from the stage itself. 

The need for localization is no more prevalent than in sound reinforcement for musical theater, but in many other venues and types of productions there may be just as much desire to provide realistic imaging. This can be accomplished (or improved) using fill loudspeakers plus various processing functions, measurement and our ears.

We see these challenges often, usually when someone has misinformed us, or forgotten to mention key information, or has “changed their mind” on seating location/configuration.

So we grab what we can (assuming that we can) and set it/them up to fill any holes. It might be sloppy but it usually solves or reduces the immediate problem.

Similar Behavior
When faced with advanced knowledge of the need to fill holes in coverage or improve imaging, etc,. the type of loudspeaker we use requires just as much thought as we (hopefully) apply to the primary system devices.

Ideally, we’re able to utilize loudspeaker(s) made by the same manufacturer as the primary loudspeakers so that there is sonic consistency throughout the system.

Most reputable loudspeaker manufacturers go to great length to ensure that the various devices within a model line, if not their entire catalog, sound alike and exhibit similar behavior across their pass bands.

This can help to save time during measurement and optimization and may be more appealing and/or visually reassuring.

We must also provide appropriately sized fill loudspeakers. Too big may look bad (and may block sight lines) and too small may run out of gas (including potential damage to drivers) in addition to not providing the required pattern control.

The word “throw” relates to the distance a loudspeaker is able to project sound to a specific (targeted) seating area, and at the required sound pressure level (SPL).

Let’s look at a hypothetical example that’s quite common in the real world of permanently installed (and some touring) point-source array systems.

At the top of the array, there are “long throw” loudspeakers.

These 2-way, horn-loaded loudspeakers provide 40-degree (horizontal) by 20-degrees(vertical) dispersion, projecting sound to the most rear seating areas at sufficient SPL, without scattering sound on to the walls or ceiling.

Below that are “medium throw” loudspeakers supplying 65-degree (h) by 40-degree (v) coverage to the “heart” of the coverage area, and below that are 80-degree (h) by 50-degree (v) devices employed as “short throw” to cover seats in the nearer field.

When aimed and combined correctly, arrays of this type can provide complete, fairly seamless (including even levels) vertical and horizontal coverage.

A Necessary Part
Note that only a few of the multiband line array systems offer optional narrow horizontal coverage devices intended to sit at the top of the array for what we would consider to be long-throw duty.

But in the far more common line arrays with duplicate/like devices, the elements are aimed and gain adjusted for the desired characteristic of providing even SPL from front to back, and we would therefore categorize the collective elements at the top as “long throw.”

A common scenario in performing arts venues, where the array provides needed throw, backed by fill loudspeakers serving several locations.

Properly designed (shaped) and configured J-arrays consist of elements or sections that we would categorize as long, medium and short throw.

But be it line arrays, point-source clusters or any other approach to mains, fill loudspeakers have remained a necessary part of what we do for decades.

We may need just one, a few, or many, but the majority of successful sound system designs rely on at least one type of fill loudspeaker to achieve complete coverage and/or for localization.

Over the decades we’ve assigned names to categorize fill applications: front, in, out, under balcony, over balcony, delay, box seat, lawn, down, overflow, stage, and so on.

Most of the time a device to meet a specific need is available, but once in a while, a new, specialized “custom” loudspeaker may need to be developed.

While there are numerous passing references to fill loudspeakers in articles and on manufacturer websites, and a book that discusses the alignment and equalization of them within larger systems, my research indicates that there’s nothing readily available that thoroughly details applications faced by live sound practitioners on a regular basis.

So next time out, I’ll begin a two-part series looking at specific fill design approaches and practices, as well as the basics of fill system optimization. 

Tom Young is a senior engineer at Altel Systems, Inc., a metro-NYC contracting firm. Recent projects include The Juilliard School and WaMu Theater at Madison Square Garden. He is also principal consultant at Electroacoustic Design Services in Connecticut and recently measured/optimized loudspeaker systems at a recital hall, a swimming pool complex and several churches. In addition, he is the moderator of the ProSoundWeb Church Sound Forum.

Symetrix Announces Release Of SymNet Composer DSP Configuration Software

Symetrix has announced the immediate release of SymNet Composer installed sound DSP configuration software. It will be on display at the upcoming 2012 InfoComm show in Las Vegas, booth C10331.

When performing basic Edge hardware configuration, Composer automatically identifies the type of audio I/O module installed in each of the Edge frames’ four I/O card slots (up to 16 channels of local audio).

When connecting multiple Edge frames via Dante gigabit network audio (up to 128 channels total), quick and easy network management is accomplished entirely within the SymNet Composer programming environment.

A fully open architecture application, Composer’s fast and fluid navigation accelerates the audio path design process. System designers exercise complete creative control selecting from a library of over 600 proven DSP processing, routing, mixing, and special purpose modules.

A designer’s proprietary Super-modules (complex blocks of multiple DSP modules and routings) can be exported from or imported into Composer and re-purposed in future projects.

SymNet Composer’s Event Scheduler automatically changes presets at pre-determined dates and times using a familiar Outlook-style calendar. If desired, hardware clocks can be set to sync to network time protocol (NTP).

Composer also configures the Symetrix ARC series of wall panel remotes along with the zero-cost embedded ARC-WEB for wireless control of Edge hardware using Apple and Android mobile devices.

With Composer, any chosen set of controls (i.e. faders, mutes, selectors) can be consolidated and exported to create custom technician or end user virtual control panels from a program called SymVue. There’s no scripting language to learn, no time-consuming graphic design.

Composer supports third-party RS-232 or Ethernet control devices with Symetrix’ human-readable external control protocol.

“DSP is deep in Symetrix’ DNA. Edge hardware, paired with Composer software, is our most exciting commercial audio product yet,” says Paul Roberts, Symetrix vice president of sales and marketing. “Announcements regarding additional audio input/output modules for Edge, as well as on-line training and certification for SymNet Composer, are slated for the very near future.”

The Symetrix technical support staff is available for remote design assistance and in-person training at .(JavaScript must be enabled to view this email address)

Symetrix

Powersoft Welcomes Matrix Sales As New Distributor For Denmark

The company Matrix Sales has been appointed as the new Powersoft distributor for Denmark.

Matrix Sales is part of the newly formed Matrix Group, made up of several well established companies working together to provide complete solutions for the touring, permanent installation and MI markets.

The Matrix product portfolio covers everything from high end professional audio, concert lighting, staging, cables and accessories and features brands of the calibre of Midas, L’Acoustics, Shure, DB Technologies, Coemar, Proel and Aviom.

The group is also related to some of the most established rental companies in the region.
“We are happy to welcome Matrix Sales into our ever growing team of international distributors.”  - says Steve Smith, Powersoft’s Touring Account Manager – “Matrix is an interesting proposal for us, having the drive and enthusiasm of fresh, growing company while featuring some highly experienced and well established elements.

“We are considering this to be a re-boot for us in the Danish market and through this powerful partnership expect 2012 to bring some excellent results.”

Kenneth Bremer, CEO at the Matrix Group states, “One of our key issues at Matrix is quality, both in terms of sales, service and the products we work with - to offer customers the right solutions and ensure a continuously high standard is the backbone of our company.

“Powersoft has precisely these properties and offer highly sophisticated products enhanced with proprietary technology. Powersoft is a strong brand and fits perfectly in our portfolio. It really is Italian engineering at its best and we look forward to offering our customers spicy Italian amplifier power!”

Morten Uldbaek, Pro-Audio Sales Manager at Matrix Sales added “It is a pleasure to be able to add Powersoft to our product portfolio. Saving space, lowering power consumption and the possibility to have complete software control and protection are some of the key features that will enable us to meet the high demands of the touring and install markets.”
The official launch of the new partnership will be at the Monitor Expo show in Copenhagen from the 30th of May to the 1st of June. Both Powersoft and Matrix will be present at the show to showcase the complete range of Powersoft amplifiers together with the Armonía software.

Powersoft
Matrix Group

Baker Audio Installs JBL VLA Line Arrays At UNC’s Kenan Memorial Stadium

Baker Audio of Norcross, Georgia, recently upgraded the audio system at the University of North Carolina’s (UNC) Kenan Memorial Stadium, home of Tar Heels football since 1927, with JBL VLA line arrays. Baker Audio shares strong ties to UNC and installed an innovative sound system to complement the stadium’s new “horseshoe” design.

Keith Hicks, President of Baker Audio, took great pride in revamping the sound in his alma mater’s stadium, where he played four decades ago as a scholarship UNC football player.

The 64,000-capacity stadium was renovated from its original shape to the horseshoe pattern, which required a completely upgraded sound system, designed by Jack McCallum of Wrightson, Johnson, Haddon & Williams, Inc.

Hanging from the new scoreboard, provided by CBS Sports, Baker Audio supplied 22 JBL VLA601 3-way full-range loudspeakers on each side of the scoreboard, while 23 AM5212 2-way loudspeakers were installed in the “bowl” under the balcony areas, which provided delayed sound coverage.

The system is powered by Crown amplifiers-43 MA-5000i’s, 14 MA-9000i’s and 14 MA-12000i’s accompanied by four CTS2000LITE amplifiers-and is processed through BSS London DSP technology with 11 BLU 800’s as well as 36 BLUCARD-OUT and four BLUCARD-IN analog cards which route all of the audio for the PA, play-by-play and music to achieve that powerful game day sound.

“The JBL system was peaking at 112dB-we’re rocking,” Hicks said. “The VLA line arrays are outstanding. The low-end response is spectacular and we continue to receive compliments on the clarity and coverage of the sound.  The UNC athletic staff is pleased with all aspects of the installation and we’ve even received calls from businesses around the city saying they can hear the music during games.”

Working under tight time constraints, Hicks noted one challenge associated with fitting Kenan Memorial Stadium with the new speaker system. While preparing to hoist the VLA speakers, Joe Kimsey, Project Manager for Baker Audio, noticed that the stacks, which each measure more than 25 feet tall, would have less than one foot of clearance within the housing.  Baker Audio was able to overcome this challenge by utilizing a 300-foot crane to carefully hang the speakers. Greg Coddington, project engineer for WJHW, was instrumental in the success of the system installation and coordination, says Hicks.

David Rotman of Rotman Architecture added, “In many aspects, it was a very complex construction project; however, due to the dedicated team at Baker, the Kenan Stadium at the University of North Carolina can deliver that ‘wow’ factor with an innovative sound system that, not only works in conjunction with the new, state-of-the-art scoreboard, but provides a state-of-the-art sound system experience at every seating location throughout the building.”

“The VLA line arrays give the stadium much more horsepower than it’s ever had,” Hicks continued. “Every aspect of the installation, from coordinating with the construction, to securing the arrays on the video board was excellent. The entire system is extremely impressive.”

“It was a great opportunity for me, personally, to fit the stadium with sound after years of attending games and even playing football there 100 years ago,” Hicks joked.

JBL

Total Events Supplies Harman-Based System For Davis Cup In Australia

Total Events, based in Geelong Victoria, supplied a Harman-based audio system for background music, MC and umpires at the Geelong Lawn Tennis Club stands for the recent Davis Cup tennis tournament in Australia.

The main challenge for Total Events was keeping a minimal layout without obscuring sightlines and the company’s stock of JBL Professional VRX Constant Curvature loudspeakers was well-suited for the application due to their low visual impact.

Six JBL VRX918S powered subwoofers and 12 JBL VRX932LA loudspeakers were deployed, powered by six Crown Audio IT8000 amplifiers all networked along with a Soundcraft Si Compact 32 digital console.

“The VRX cabinets are discreet so you can point and shoot without obscuring the view of the tennis action,” notes Bill Busbridge, managing director of Total Events. “They have a nice high power output and it’s good to be able to network the amplifiers so I can control remotely and time align.”

“I absolutely love the Soundcraft Si Compact console,” Busbridge adds. “I have two of them in hire and I have also sold a couple. I really enjoy using them and we’re using them a lot on our shows and events. I really like that you can move channels to any layer is great. The small footprint is a bonus and it’s just so easy to grab it and use. We find a lot of walk-in engineers who are familiar with the console.”

The system was sold through Jands, JBL and Soundcraft’s Australian distributor.

Harman
Crown Audio
JBL Professional
Soundcraft
Total Events

More Than 500 EAW Loudspeakers For Miami Marlins New Stadium

In April, the Miami Marlins inaugurated a new season of baseball in brand-new Marlins Park, located in the Little Havana neighborhood of Miami.

The stadium’s distributed sound reinforcement system, designed by Wrightson, Johnson, Haddon & Williams (WJHW) of Dallas, incorporates 517 EAW MK, QX and MQX Series loudspeakers. The system was installed by Minneapolis-based systems integrator Parsons Electric.

Marlins Park has a retractable roof, but in order to best take advantage of the sun for its natural-grass field, the stadium had to be angled in such a way that the grass get as much exposure to the sun as possible between games.

As a result, rather than the usual half-dozen or so section templates in the seating areas for which a basic configuration of loudspeakers would make up a repeatable pattern, this system’s design required over 30 individual section drawings, with nearly every one of the upper-deck sections being unique. Thus, loudspeaker placement and aiming was different for virtually every location on the upper deck.

The loudspeakers are grouped into 23 categories, in some cases with multiple units required to deliver coverage to a single area.

In addition, the stadium’s asymmetrical design meant that focal points often needed different throw-length speakers set at different distances and patterns. For instance, the area around home plate uses the EAW MQX8343-MS-WP long-throw enclosure from one side, but is matched on the baseline sides and outfield porch by a shorter-throw QX564-WP unit.

“The system design was very complex, but the EAW speakers were exactly what we needed to make it work,” says Tim Habedank, systems specialist at Parsons Electric. “On opening day, the sound was clear and highly intelligible, even over the noise of a capacity crowd.”

Habedank adds that while the roof is likely to be closed for most events at the stadium, the long reverb time when it is closed, particularly in the lower frequencies, will remain under complete control, thanks to the precise aiming pattern that the EAW speakers are capable of.

“We were able to achieve very tight pattern control across the spectrum,” Habedank says. “Yet at the same time the longest-throw speakers maintain the coherence of the sound for distances of between 200 and 300 feet. It’s really remarkable performance and the sound of the stadium shows it.”

EAW

Renkus-Heinz Continues To Expand Engineering Department

Renkus-Heinz has announced the continued expansion of its engineering department with the addition of Alejandro Fidalgo.

Fidalgo will work within the engineering group to continue to develop advanced digital processing and transport technologies for Renkus-Heinz products.

He will report to engineering manager Tim Shuttleworth and will be based at the company’s Foothill Ranch, CA headquarters.

Fidalgo joins Renkus-Heinz after working in audio electronics firmware and hardware development at Isaac Daniel Group in Burbank, CA. He holds a degree from the University of California, Santa Cruz.

“We’re very happy to welcome Alejandro to Renkus-Heinz engineering,” says Tim Shuttleworth. “His expertise in developing complex digital audio technologies is a great fit for our team.”

“I’m thrilled to be working with a company like Renkus-Heinz,” Fidalgo adds. “This company has such a long history of making great products, and I’m really excited to be able to contribute to the next generation of Renkus-Heinz technologies.”

Renkus-Heinz

Friday, May 11, 2012

Wireless Primer: The Key Issues Of Digital Audio Transmission

Digital is a buzzword that many presume solves all the technical issues we face today.

More and more digital equipment, such as mixing consoles, audio signal processors, and the like, are used for several applications, as a digital audio signal chain offers many advantages.

A digital signal on a wire (i.e., fiber optic cable) is easier to handle than on a copper wire because 48, 64, or more audio channels can be transported on one thin fiber optic cable. If an audio signal is already in the digital domain, it makes sense to keep it in this domain as long as possible.

As for digital wireless transmission, a digital wireless system is beneficial when the sound, occupied RF spectrum, and battery lifetime is as good or even better than an analog system. On top of this, latency (time delay between input and output) is always a very important topic to keep in mind.

Let’s start with sound and the related data rate.

The best sound can be expected if there is no audio data compression used in the wireless system. This will lead to a very high data rate.

• Minimum for 20 kHz audio and approximately 110 dB dynamic range: 18-bit 48 kHz = 0.864 Mbit/s

• Necessary overhead (framing, channel coding) leads to even higher data rate (factor approx. 1.5 to 1.296 Mbit/s)

When transmitting this high amount of data, it is no longer possible to use a simple and robust digital modulation scheme like FSK (Frequency Shift Keying) ASK (Amplitude Shift Keying) or PSK (Phase Shift Keying), because these concepts will be not able to fulfill the spectrum mask, 200 kHz of occupied RF spectrum, defined by the FCC. Even if this constraint didn’t exist, greater occupied RF spectrum could inhibit large multichannel systems.

To improve this, it is necessary to use a more complex modulation scheme with narrow filtering. The amplitude and the phase of the transmitted signal must be very precise when usmg this approach.

Behind every point of the constellation diagram, a digital word is deposited, which the receiver has to pick up and transfer back into an audio signal. This requires a very linear RF amplifier. This is a current-hungry device. The unwanted effect is reduced battery life of transmitters and portable receivers. By driving the RF amplifier with a better efficiency, the occupied RF spectrum will increase in an unwanted way.

If the data rate described above can be reduced, the modulation scheme can be simplified and the amplified RF can be used in a more efficient way to conserve battery power and increase operational time.

Constellation diagram of a 16 GAM modulation. (click to enlarge)

To reduce the amount of digital data, a compression algorithm has to be defined. This algorithm will add some latency to the whole data= transmission process. low latency is especially important during a live performance on stage.

If the total latency in a PA system, including contributions from digital mixing consoles, effects, etc., is greater than 10 ms, the timing of the band will be thrown off.

Further, if streaming video is projected to accommodate a large audience the picture and sound will be out of sync.

New audio data compression algorithms show good performance with a very low latency’, However, audio compression would introduce the possibility of audible artifacts (at least with awkward signals).

As technology improves, there will be solutions to the obstacles described above and digital will become available for wireless transmission.

The key questions for a digital system at this time are:

—Is data compression used?

—What RF spectrum is necessary and how will this impact multichannel systems?

—What is the latency of the system?

—What is the battery lifetime?

Volker Schmitt is a senior engineer for Sennheiser US, and Joe Ciaudelli also works with Sennheiser US and has a history of providing frequency coordination for large multi-channel wireless microphone systems used on Broadway and by broadcast networks.

New York Designer Specifies RCF At La Bodega Negra

CP Sound has completed a full sound system installation at an upmarket Mexican restaurant in London’s Soho district.

The brainchild of Will Ricker (who heads up Ricker Restaurants), top New York nightlife designer Serge Becker was brought in to create the interior of this two-in-one operation.

With a fully featured DJ booth set downstairs in the 95-cover basement restaurant lounge, Becker specified RCF sound reinforcement throughout, based on his positive experiences with the brand in various locations in New York City.

Designing the sound for the hacienda style basement into five zones, Colin Pattenden of CP Sound wall-mounted four C3110 full range, compact 10-inch and 1.5-inch horn, wide-dispersion, low profile loudspeakers, one in each corner of the main Piano Room. Providing low frequency extension are a pair of concealed Acustica S8015 low-profile Band-pass 15-inch subwoofers.

In the basement’s secondary dining area, which forms its own independent zone, are an additional pair of RCF C3110’s and an S8015 sub, with three further C3110’s and a low profile sub distributing sound around the Basement Bar.

Stated Pattenden, “Serge Becker requested that we used RCF and we were happy to do so as these are extremely powerful speakers. The system has been carefully processed, and although La Bodega Negra is situated under the Z Hotel, there is massive concrete isolation which enables the system to be pushed up to around 115dB if necessary.

“The S8015’s were fantastic for installing because they fitted snuggly underneath all the seating.”

Feeds into the system vary, from CD playback from Pioneer CDJ-900’s — with a DJ operating Tuesday through Saturday — to computer music files, which provide general background music.

The sound has been carefully processed to optimize the coverage, and is distributed through a series of zones to the ten different areas of the restaurant.

With a lively café and taqueria on the ground floor complementing the downstairs restaurant (each served by separate entrances), La Bodega Negra is the latest addition to Ricker Restaurants’ portfolio, their other units including such well-known haunts as The Great Eastern Dining Room, Eight Over Eight and E&O. Due to the locality of the venue homage is also paid to the history of the area — with a faux frontage designed to simulate a sex shop, with red neon lights.

Stated general manager, Richard Seldon, “These speakers produce a clean, well-balanced sound, with good bottom end. Downstairs we need music that’s loud, but at the same time doesn’t interfere with the dining. The RCF system delivers that.”

RCF

Adam Shulman Appointed As EAW Installed Systems Support Manager

EAW announces the expansion of its Application Support Group (ASG) with the appointment of Adam Shulman to the position of Installed Systems Support Manager, effective immediately.

The announcement was made by Jeff Rocha, EAW President, and further underscores EAW’s dedication to customer support and the ongoing development of market-driven products.

In this newly-created role, Shulman will manage ASG support for all permanently installed sound systems, while longtime ASG resource Joe Fustolo will focus on mobile production customers.

Rocha states, “Expanding ASG and bringing in a world-class resource like Adam, who is dedicated exclusively to the installation markets, further demonstrates our commitment to supporting our customers and partners.”

Increasing global demand for ASG service, particularly in overseas markets, necessitated the expansion. Jerrold Stevens, Director, EAW Application Support Group, adds, “Adam’s experience in system design, project management, acoustics, education and publishing all further enhance the team’s capabilities, helping us meet the growing demand for our services.”

Prior to joining EAW, Adam has served as a Senior Consultant and Project Manager for SIA Acoustics (New York, Los Angeles and India) since 2003. He has provided acoustical and technical system design for a variety of projects including performing arts spaces, recording facilities, sports venues and houses of worship. Projects of note include a production facility and 3,500-seat arena for Healing Place Church in Baton Rouge, LA, Oriole Park at Camden Yards, and The Pearl at the Palms Concert Hall in Las Vegas.

In a production context, Adam has also designed and operated sound systems for live events, including Central Park SummerStage, the Madison Square Park Music series and various special events domestically and internationally. As an educator, Adam has authored numerous articles for various trade publications, and presented at Broadway Sound Master Classes, Palme Asia, ShowWay Italy and Audio Engineering Society conventions.

Shulman notes, “I feel that I can bring my past experience as a consultant to bear on the varied types of design projects ASG handles, applying my technical knowledge combined with a practical understanding of how installation projects actually work.”

Members of the EAW Application Support Group, including Stevens, Shulman and Fustolo, will be on hand at InfoComm 2012 at the EAW booth (C10139) where they will reprise their popular ASGenius Bar – a coffee bar where designers, contractors and integrators can ask questions and review plans.

EAW

Thursday, May 10, 2012

More Than One Way: Alternatives To High-Voltage Audio Power Distribution

In audio terms, high voltage means that the output power of the amplifier is converted to a high-voltage/low-current signal for transmission over long distances and/or small wire gauges.

The advantages of the method include low cost and rather “bulletproof” systems, and the downside is that the transformers required present yet another filter for the signal to pass through, often degrading the audio quality.

Since loudspeaker lines should always be kept as short as possible, the ultimate realization of this involves placing the amplifier right at the loudspeaker, connected to it by inches of cable.

This method is best understood by looking at the workings of a traditional power amplifier. There are many shapes and sizes, but they all have some commonalties.

First, all amplifiers take AC power (alternating current) and use it to amplify signals. This requires converting the 60 Hz sinusoidal signal from the power company into something that looks like the audio signal that we wish to amplify. Several steps are required to accomplish this.

To begin with, the voltage component of the power is transformed from 110 or 220 volts (common distribution voltages) into the voltage required by the amplifier circuitry, which is determined by the power rating of the amplifier.

Next, the new value of voltage and current is rectified into DC (direct current). In DC form, the power can be “modulated” by the audio signal voltage to form a higher-power facsimile of the input signal voltage to the amplifier. This step is accomplished by the output stages of the amplifier.

Figure 1 (below) shows the parts of a typical power amplifier. Conventional systems take the amplified output of the power amplifier and feed it to loudspeakers through a wire gauge of sufficient size to minimize the power loss to 0.5 dB.

When the required wire gauge becomes too large, the power is delivered to the loudspeakers by transforming it into a high voltage/low current signal, more suitable for traveling long distances.

Distributed amplification systems involve separating the parts of the amplifier and distributing them to remote physical locations that, for some applications, better optimize the wiring and interconnects that make a sound system work.

For instance, if the output stage of the amplifier were placed right at the loudspeaker, there would only be a need for a few inches of loudspeaker cable. In order to avoid having to run AC power to all loudspeaker locations, a central DC power supply can be used to drive many amplifier sections.

The central supply can be located near the AC power source, and the DC output coupled to the amplifier sections through appropriate cabling.

For large systems, several power supplies can be distributed to keep the distance between them and the loudspeaker/amplifier units at a minimum.

All that remains is to get the electrical signal voltage to all of the “distributed” amplifier/loudspeakers.

Since this is a line level signal, it can be run very long distances without significant degradation.

Finally, if the DC and signal are run through the same multi-conductor cable, installation of such a system is greatly facilitated.

For electronic systems, DC is an ideal way to power things, since almost every unit in a sound system must convert AC to DC in order to work.

In fact, when the AC power standards were established years ago, there were many people, including Thomas Edison, that wanted to use DC distribution. It makes a lot of sense for much of what we use power for.

Advantages:
Short Loudspeaker Lines
Higher Fidelity
Lower Operating Voltages
Conduit is not required in many locals

Disadvantages:
Increased cost over conventional systems
Upgrades are more complex (this is probably an advantage. Ask anyone who has ever had a customer hang a transformerless loudspeaker on their 70 volt line and load it down).
Very high power amplifiers not available

Author’s Note: The technology was actually developed by Richard Heyser of the Jet Propulsion Laboratories in the 1980s.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world. For more information go to synaudcon.com.

Upgraded Audio At Riverside Casino Events Center Led By Renkus-Heinz Arrays

The Riverside Casino & Golf Resort, located just south of Iowa City, is also home to a multipurpose Events Center that offers a busy schedule of top name entertainment featuring artists like Big & Rich, Peter Frampton, The Robert Cray Band, and Tony Orlando, as well as hosting corporate, community and private functions.

The resort has recently upgraded its in-house audio capabilities with the installation of four Renkus-Heinz CF101LA modular point source array loudspeakers. Marvin Smejkal, owner of Sound Concepts, a production sound installation and rental company with offices in Iowa, Missouri and Florida, notes that the project presented a number of challenges.

“The original in-house, distributed voice reinforcement system was insufficient for many of the venue’s functions, but it was generally cost prohibitive to bring in a larger rental rig,” says Smejkal, adding that the self-powered CF101LA loudspeakers serve a dual purpose, both as a primary voice reinforcement system for smaller events and as center in-fill for larger PA rental rigs.

“The venue has unique requirements due to an exceptionally wide stage,” he continues, “and when we bring in a large PA for concerts, the speakers have to be positioned in a wide configuration, making a center-fill necessary. It was costly and time-prohibitive to set up and tear down a flown center cluster on a show-by-show basis. Now, the CF101LA system can be used as the primary PA for small and mid-sized events, and as a center fill to augment a larger PA systems for big concerts.”

Smejkal designed and installed a basic four-loudspeaker setup requiring no additional processing or EQ. “The venue has a portable audio/visual mixer for breakout rooms, which can simply plug into an XLR in for smaller events. The CF101LA speakers are incorporated into every large show; when we come in we have a program to add them into our rental system.

“The self-powered speakers are a cost-effective option when there is limited space to permanently install additional equipment in a venue. “The CF101LAs are convenient, and are fulfilling a wide variety of needs, both as a stand-alone system for the venue’s day-to-day needs, and in the demanding role of center fills when larger line arrays are widely spread for larger events.”

The CF101LA speakers proved to be a unique problem solver, he concludes. “Those speakers deliver at that price point. We will use them again.”

Renkus-Heinz

Wednesday, May 09, 2012

How It’s Made: Inside FaitalPRO, Driving Loudspeaker Driver Development

Although a relative newcomer to the pro audio world, Faital, headquartered in San Donato, a suburb of Italy’s business capital Milan, has more than half a century of loudspeaker driver manufacturing to its credit.

In 2006, the family-run concern launched FaitalPRO, a division of the company targeting the international pro audio market, which has grown by leaps and bounds since inception, as explained by FaitalPRO overseas sales manager Flavio Naggi, grandson of the company’s founder.

“Although my father is company CEO, my uncle is the president, and my brother is in charge of the sourcing and purchasing department, Faital has outgrown the typical family-size business format and is now an international group, with manufacturing facilities in Italy and fully-owned factories in Hungary and Spain, chosen to ensure fast coverage of the whole of Europe,” Naggi explains.

“We also have sales branches in the U.S., Mexico, France and Hong Kong,” he continues. “Our pro audio division has averaged a 70 to 85 percent annual growth rate, because our range of woofers, compression drivers and horns are appreciated as providing very high quality and long-lasting performance.”

The Faital 86,000 square-footmanufacturingplant in Chieve, Italy. (click to enlarge)

FaitalPRO did not intend shifting manufacturing or R&D to lower cost countries, so decided to focus on the higher end of the market. “We have big momentum in direct product distribution, but, although this gives more rapid gratification, OEM work is a key objective,” Naggi notes. “This takes considerable time to develop, as potential customers must decide to launch a new range of products or have a problem with a current supplier, after which you need to develop the product they need, as they don’t always buy catalog products, and this can take up to two years.

“However, we offer a guarantee of quality, continuity and R&D integrity comparable to top brands on the market, if not higher, because in many areas we have an infrastructure originally specific to our automotive background, guaranteeing the quality of processes, materials used, design and the entire development process.”

Area manager Gianluca Turra adds, “FaitalPRO began with market research to understand what the pro audio industry required for a number of key applications, then the study, concept and design of speakers that could be competitive with or better than those already available and could be produced with our highly automated production methods - able to turn out a woofer every 15 to 20 seconds, or up to 25 with 18-inch models with complex assemblies.”

Left to right, key Faital team members Gianluca Turra (area manager), Mario Passarelli (senior project leader) and Flavio Naggi (overseas sales manager) with some of the company’s drivers. (click to enlarge)

This led to the build-up of the current range of products, with senior project leader Mario Passarelli noting, “In 2008, we were the first to market an extremely high-power subwoofer with a 4-inch voice coil. Prior to our XL Series, subwoofers with 4-inch voice coils couldn’t go to more than 1,000 or 1,200 watts.

“Our extremely long excursion very high-power 18-inch subwoofer in neodymium reached 1,500 watts and beyond, which was quiet an achievement and was a trend followed by many other manufacturers.”

Avant-Garde Facility

Adjacent to the Faital headquarters is the R&D department, the starting point of all the new products and the patented technology adopted by the company.

The specialized staff of over 20 full-time technicians on the R&D team have at their disposal an impressive array of cutting edge systems and software used for the design, validating and testing of components and prototypes, as well as materials used by the company’s suppliers and many of the tools actually used on production lines.

The avant-garde facility also cooperates regularly on joint projects with universities and other bodies.

“We have a series of sophisticated instruments for checking all aspects of the components when they arrive – physical, magnetic, variations due to external influences, such as temperature,” says R&D manager Romolo Toppi. “We must also make certain that materials’ characteristics remain constant, particularly important as far as neodymium magnets are concerned, as there is considerable misconception among suppliers regarding standards.”

Loudspeaker performance is evaluated via acoustic measurements in two anechoic chambers (one fully floating), laser-based assessment, performance with large signals and analysis of geometry and behavior of moving parts.

An entire in-house validation infrastructure enables to carrying out a variety of tests on components, prototypes and end products include corrosion, thermal shock, UV rays and vibration and shock testing, to see how they’ll stand up to use (and misuse) in future applications.

Of particular importance is the capability of guaranteeing that all Faital products will be corrosion-proof, waterproof and capable of withstanding very broad thermal and vibration shocks,
making them environmentally impervious to anything mother nature (or users) will throw at them.

“A great deal of attention goes into developing components that are producible in the most economic manner and able to guarantee performance, but having implemented the strict regulations in other industry sectors enables FaitalPRO to maintain very high quality standards,” Naggi says.

A sophisticated product development process includes 3-D design, extensive prototyping, and evaluation in one of the company’s anechoic chambers.

From the incoming inspection of materials, there are stringent almost “military” level quality control and tests to ensure that products work in the conditions decided upon with clients at the beginning of the program. The company also tests, controls and even purchases the material – such as plastic – used by its suppliers.

Cones are tested on arrival before being mounted on actual loudspeakers, and there’s also a 3-D measurement system to compare components with the original models ordered. End products are also labeled to enable them to be back-tracked down the entire chain.

Highly Productive

Located in extensive tree-shaded grounds in the rural town of Chieve, Faital’s 86,000 square-foot (8,000-square-meter) manufacturing plant, just a half-hour drive from the Milan metropolis, features highly automated production lines designed for extreme flexibility.

Naggi explains: “The design and automation of the lines enables a number of different models to be produced with almost no down-time between job lots, apart from a few minutes required to reset the machines via touch panels, ensuring an extremely high productivity rate.”

Highly automated assembly lines provide precision manufacturing in addition to enabling different models to be produced with almost no down-time.

The facility’s warehouse system is equally streamlined and includes climatized zones for components more sensitive to temperature and a special dedicated adhesive store-room.

The actual production line begins with the assembly of the magnet assemblies, some of which are extremely complex, includes curing chambers that can be adapted according to the type of adhesives (also formulated to Faital specs). Along the line there are cleaning stations to make certain assemblies are absolutely free from unwanted particles (or “crap in the gap” as Turra memorably refers to it).

“Thorough cleaning inside and out before applying dust caps is fundamental, as the air gap is where you have the least space and the most movement, so very little tolerance,” he adds.

Test stations verify aspects such as correct magnetization and component bonding, and although component positioning on the line is almost all auto mated, certain aspects, such as ensuring that for example one part mounted inside another is fully inserted, require an experienced human touch.

Cone application for example is carried out manually, as soft materials are unsuited to robotic handling.

Naggi stresses, “Some manufacturers also apply adhesive manually, but dosage is of fundamental importance, since – as well as looking messy – surplus adhesive adds weight and moving mass plays an important role in performance. Applicators are thus fully automatic, have preset programs for the various speaker models and can apply two (or more) adhesives simultaneously.”

Before packaging, finished products undergo thorough test procedures, starting with a visual inspection and including tests with signals to check physical integrity visually, then computerized tests for reproduction parameters.

An extensive testing process on products covers a wide range of factors such as climate issues, shock/vibration, and much more, to see how they’ll stand up in actual use.

Bold New Directions

Never believers in resting on their laurels, the FaitalPRO team has decided to launch an additional new range of products, based entirely on ferrite magnet technology, but before going into detail on the company’s incredible commitment to this ambitious project, Naggi expresses in no uncertain terms their ideas on neodymium.

“To cut a long story short, we don’t fully agree with the ongoing panic of some of our market’s players regarding neodymium, much of which is caused by incorrect information,” he says. “My opinion - also that of the rest of our top management and sourcing department - is that the situation will not remain stressful for a long period, as there is the opportunity for other countries and other companies to start extracting rare earth minerals from several other sources not currently being exploited, including a very serious project under way at the moment for extraction from the seabed.”

FaitalPRO has adopted a two-fold approach to the situation. One part is to mitigate whatever damage has come from the way the market is behaving, purchasing on average from 200 to 220 metric tons per year of neodymium magnets.

“In the market we’re one of the largest purchasers, which is appreciated by our suppliers,” Naggi explains. “Therefore, we have the capability of minimizing the effect of cost increases for neodymium on the final price of the speaker, thus transferring to our clients as little increase as possible.”

The other part of the approach is a much stronger statement – the creation of a big alternative to neodymium products, on which the entire catalog with the exception of a few small-format models has been based so far – with the launch of an eye-popping 31 new ferrite-based products.

Just a few of the new ferrite driver models currently being rolled out by FaitalPRO. (click to enlarge)

“Although we made our neodymium products competitive with other manufacturers’ ferrite models, there was a slice of the market that wanted ferrite speakers no matter what, so before the end of the year we enter pre-series production on the new products, designed from scratch – new baskets, new magnet assemblies, new everything,” he states.

Ready To Go

The entire development process began in March, and after just nine months of intensive work, a full product range is ready to go. Company officials stressed that FaitalPRO is not discontinuing neodymium, but rather offering an alternative – in fact, it will continue offering all current neodymium models and even add new models next year, alongside ferrite, thus creating a new flow of development, not a substitute.

Turra adds, “This project is not a trade-in at the cost of quality either. In fact, some speakers actually improve with ferrite, since the magnetic field works in a completely different way, favoring some of the features that are particularly appreciated in subwoofers.”

Concluding, Naggi notes, “After lengthy simulation work and a considerable amount of in-depth practical work in the field by our R&D team, we have also devised a very innovative method for cooling the ferrite magnets mounted, so this has been a huge undertaking, a very exciting time for us, and one of the biggest achievements since the inception of the company.”

Based in Italy, Mike Clark is a long-time writer on professional audio topics.

Tuesday, May 08, 2012

Understanding Sound System, Loudspeaker & Room Interactions

If one could listen to only the direct sound of a loudspeaker, the world would be a very different place!

Unfortunately, free field listening, where you have no reflections, room modes or ambient noise, is hard to achieve in everyday life, so we listen to loudspeakers in real rooms.

The interaction of a loudspeaker system and a room can be very complex to understand, model or measure!

One way to measure this interaction is to measure the impulse response of the loudspeaker/room system.

The impulse response of a typical sound system in a room contains lots of interesting information, including:

1) The delay between the loudspeaker and measurement microphone

2) The direct sound-to-reverberent level ratio

3) The time arrival, frequency content and level of reflections of sound

4) The early and late decay rates of the sound

5) The frequency response of the direct sound.

This last point is particularly interesting. The question is “What do we want to measure and why?”

Figure 1: The impulse response of a 1250 seat multi-purpose hall. The x-axis is time (~0.75 sec) and the y-axis is magnitude in dB. Note the direct sound, reflections, the reverberant decay and the noise floor.

One question that goes to the heart of “system” measurement and optimization issues is “If the impulse response contains the frequency response of the direct sound, can we separate the loudspeaker response from the room response?”

Also “If we can, do we want to?”

Figure 1 shows an impulse response displayed in the time domain.

The “spike” that represents the direct sound actually contains the frequency and phase information about the loudspeaker.

To see this information we must transform this portion of the impulse response into the frequency domain.

Figure 2: The impulse response of a 1250 seat multi-purpose hall. The vertical lines suggest a time window that ignores most of the effects of the room at frequencies whose periods are longer than the time window (i.e. low frequencies).

To achieve this isolation of the direct sound from the room response, we must select a time window that includes the direct sound but excludes the reflections and decay of the room.

Figure 2 displays such a time window. This measurement was made using a full range loudspeaker system with the microphone approximately 60’ from the loudspeaker.

Pink noise was used as a reference signal and the impulse response was calculated using a 512K FFT (although only the first ~0.75 seconds are shown).

We can take the “time windowed” data and transform it into the frequency domain using FFT mathematics.

This transformation yields a result that shows how much energy is present at each frequency, as shown in Figure 3.

You can see the pronounced roll-off of low frequency energy. You can also notice the lack of LF resolution in this figure.

The lack of resolution at LF is offset by a excess of HF resolution.

This uneven resolution between LF and HF energy is the result of the FFT mathematics used to transform the data from the time domain to the frequency domain.

Standard FFTs yield data that is distributed linearly in frequency (one data point every X Hertz).

Unfortunately, humans perceive frequency logarithmically.

Figure 3: The frequency response of the direct sound portion of an impulse response of a 1250 seat multi-purpose hall. The response was calculated using a 512 point FFT (which equals a 512/48000 or ~11 msec). As you can see the frequency response shows a pronounced LF roll-off.

This lack of LF resolution in Figure 3 is a direct result of the use of a short time window in our transformation from the time domain to the frequency domain.

It is interesting to note that this plot does not correlate with what we hear.

Simply listening to the full range loudspeaker system we were measuring made it clear that the system was reproducing LF energy down to at least 100 Hz!

I would suggest that a primary goal of an effective measurement system should be to provide results that correlate well with what we hear.

So the lack of correlation between what we have heard and what we measured suggests a modification to our approach.

As an alternate approach to trying to find a measurement that correlates with what we hear, we can try using a longer time window to “see” the LF response with better resolution.

A longer time window of approximately 250 msec is shown in Figure 4.

Figure 4: The impulse response of a 1250 seat multipurpose hall. The vertical lines suggest a time window that INCLUDES most of the effects of the room. The time window shown is approximately 0.25 seconds.

To transform this longer “slice” of the impulse response into the frequency domain, we will use an 8k FFT which represents 8k/48000 seconds, or 0.171 seconds.

Notice again that this time window includes both the direct sound and the response of the room.

In Figure 5 the low frequency information is seen in adequate resolution, however the high frequency results look confusing. The plot shows data that has 5 Hz resolution (i.e. one data point every 5 Hz).

While this resolution provides excellent LF resolution (between 31 Hz and 62.5 Hz there are 15 data points.

However at HF we have excessive resolution - between 4 kHz and 8 kHz there are approximately 800 data points.

Simply stated, the longer time window provides good LF resolution, but excessive HF resolution.

The result of studying these plots might lead you to conclude that in order to make measurements that correlate well with our listening experience, we must use very short time windows that isolate the direct sound at high frequencies, and increasingly longer time windows as we look at lower frequencies.

At first glance this idea might seem to violate the often quoted phrase, “One can only affect the direct sound with processing.”

However this is not the case. At mid-low and low frequencies, the interaction of a sound system and a room can be affected and optimized by signal processing.

In other words, at low frequencies (long wavelengths) the direct sound and reflections from nearby surfaces combine to form a composite response. It is this composite response that a listener hears.

The ability to measure several time windows simultaneously provides a measurement that both correlates well with human hearing and provides insight into how the signal being sent to the loudspeaker can be tailored (via equalizers, or other processing) to optimize the loudspeaker/room interaction.

Figure 5: The frequency response of the direct sound portion of an impulse response of a 1,250-seat multi-purpose hall. The response was calculated using a 8192 point FFT (which equals a 8192/48000 or ~107 msec). As you can see the frequency response shows low frequency energy that is much more pronounced than seen with the shorter time window.

Our last figure shows a measurement of a loudspeaker system that includes multiple time windows and displays both the magnitude and phase response of the “system.”

The use of multiple time windows allows one to isolate the direct sound of a loudspeaker in a real-world situation at high frequencies.

However, at lower frequencies, longer time windows that include the loudspeaker/room interaction have been found to correlate well with our listening experience.

Multiple time windows in a single measurement is an extremely interesting way to measure and optimize the response of a sound system in a room.

Sam Berkow has completed a wide variety of acoustical design projects including: concert halls, recording studios, broadcast facilities, production facilities, house of worship facilities, large multi-purpose venues, amphitheaters and stadiums. His educational background includes a masters degree in Engineering from the Stevens Institute of Technology, where he specialized in acoustic measurement and design. He is also the original developer of Smaart acoustic measurement & system optimization software.

Monday, May 07, 2012

Factors Defining A “Good” Sound Reinforcement System

How many sound systems have been built and are in use? Many millions, for sure, and they’re found in all types of venues and for all kinds of programs.

So one would think we’d know exactly how to do it by now. But there seems to be plenty of examples to prove that we don’t.

Why should this be? What is it we don’t yet understand? Do we even know enough to know what we don’t know?

Perhaps we should start by trying to define the characteristics of a good system. Not just “it sounds good” but - exactly - what makes the difference between “good” sound and not so good.

Then we might be able to quantify how good each characteristic needs to be and how to judge whether it’s good enough or not.

After nearly 40 years spent designing and testing sound systems, I’ve finally come up with a list of the factors that I feel make up what we could call quality in a system, and why. For purposes of my discussion here, I’m going to confine my list and discussion to systems for speech reinforcement only, and will look at factors for music systems at a later date.

Reliability. The most important quality factor has to be reliability. No matter how good the performance of a system may be, if it fails to work, it is useless.

Reliability is largely an engineering matter, involving component selection, configuration design, and assembly and installation correctness, for example, but any system can be abused to the point of failure. 

Significantly, failure may not be abrupt and catastrophic, but instead may take the form of performance decline due to damage.

One particular, and common, example of damage-induced deterioration can be found commonly-used transducer for higher audio frequencies, the horn and compression driver combination.

Drivers have a severe amplitude limit; if over driven, the driver diaphragm will impact the phasing plug, an essential part of the structure. If the diaphragm material is metallic, it can fracture and fail.

Surviving a Collision
Some diaphragms, however, are made of a resin-impregnated fabric, which is much less brittle and, therefore, more able to survive a collision with the phasing plug.

Repeated collisions, however, still cause progressive deformation (or warping) of the diaphragm, resulting in eventual failure and therefore, progressive decline of the driver’s performance characteristics.

Predicting and detecting this impending failure, however, is not easy to do.

The audible change in performance is fairly subtle and can be detected reliably only by careful comparison of the sound of a single questionable driver with that of a known good one.

In the field, such a comparison is usually impractical.

Further, a driver that has been used heavily for some time will also exhibit some performance deterioration, even though it has never been over driven into diaphragm collision.

Figure 1 (at right, click to en­large) illus­trates these per­form­ance differences.

The frequency response (amplitude versus frequency) of three drivers of the same model (with an impregnated-fabric diaphragm), one new, one well used but apparently undamaged, and one with observable damage.

It can be seen that the response at higher frequencies changes with use or abuse. The differences between the upper two measurements are slight, while the third one is significantly different.

There seems to be a good relationship between the measured and (subjectively) observed performances in cases like these, but no real study of this relationship has been performed.

So it would seem that a response measurement could be a valid substitute for a listening test. In fact, such a relationship has been established under certain circumstances, but not definitively in a sound reinforcement context. An investigation of this relationship would certainly be worthwhile. 

However, there is another measurement that is easy to make, even though it’s seldom done. The bottom three curves on Figure 1 represent the measured electrical impedance at the input terminals of each of the three drivers. 

Such a measurement is usually quite easy to make, even on a driver installed in a system. 

It’s apparent that these curves separate the characteristics of the three drivers as well as any other common measurement does, especially in the case of the damaged unit, and much more easily. In fact, automated tests of this type have been designed into integrated systems as performance and reliability checks, with good results.

Thus it appears that different types of tests on the same items can yield corresponding results. In fact, experience has shown that such relationships hold in some cases but not in others, and that it may be difficult to predict which is which.

And in many cases, no acceptable substitute for a listening test has yet been found. Worse, some widely accepted tests might prove inadequate.

Turn It Up?
Loudness. It’s obvious that any sound system must provide enough sound level at the audience locations to ensure a satisfactory listening experience. Defining what this level actually should be is less obvious, and use of a valid measurement technique is not obvious at all. Subjective opinions on appropriate sound levels often vary widely as well, depending on a host of factors. (Investigating this matter alone could become a major research project!)

In fact, the correct sound level may not be just a matter of loudness. How well speech is understood (intelligibility) is often the overriding concern, and this is the result of many factors other than just loudness. In some cases, the loudness may need to be set other than as would normally be expected, because of adverse acoustical or system functional characteristics. It may also be found that the audience prefers a sound level different from that which exists near the performer.

Other acoustical factors may also be highly significant. The level of the reinforced sound must be sufficiently higher than that of any background noise so that speech intelligibility or program enjoyment is maintained. Some guidelines in this regard have been established empirically, and they may be adequate for most situations. 

A common and complicating factor is that background noise level may vary significantly, rapidly and unpredictably. Further, since adequate performance in this area may be a matter of life safety, accuracy can be quite important.

It’s often the case that the desired sound level is greater than that which the system is capable of producing without difficulty. This difficulty is the result of one or more components overloading, which results in an audible distortion of the sound.

Distortion may take various forms, depending on the type of component that is overloaded, the magnitude of the overload, and the nature of the program material, among other factors.

Therefore, the audibility of the distortion may vary greatly with the situation, and each type of distortion must be evaluated individually.

Many listeners even believe that certain types of distortion are desirable, such as that typically produced by vacuum tube amplifiers. This usually applies to music playback systems in small rooms, however, so it’s unclear if such an effect is valid in a larger sound reinforcement situation. 

Some devices are available that deliberately introduce controlled distortion, specifically for pro audio applications. Many have noticed that a limited amount of distortion adds to the apparent loudness of amplified sound, and without being objectionable.  If anyone has actually studied this effect, the results remain obscure

Timbre. The overall timbre, or tonal balance, of a sound system undoubtedly has the strongest influence on the overall perceived quality. This characteristic is easy to measure, both subjectively and objectively, and there is a very good correlation between the two in a small-room configuration. 

In a large-room sound reinforcement situation, however, this correlation does not hold. If the system has an overall response that is measurably flat (has nearly the same input-to-output level ratio at all frequencies), it will sound too bright, with the high frequencies being too loud. A system which sounds subjectively flat, so that the reproduced sound is perceived as being a close duplicate of the source, will have a measured response which rolls down at high frequencies. 

Should the analysis be done with a swept filter, which yields more information, or is a stepped filter technique acceptable? What amplitude smoothing or averaging is appropriate? If measurements are taken at single, discrete frequencies, as are commonly done with contemporary techniques, how many measurement points are needed and at what spacing? This could be a major source of misleading data, especially at lower frequencies.

Whatever the technique, how many measurement locations should be taken, and where should they be located? And exactly how should the individual measurements be averaged to yield the overall system response? Also, how much variation between individual measurements is acceptable, and what should be done if the variation exceeds this tolerance?

Small vs Large
Schulein documented this discrepancy in 1975 in an elegant experiment and offered a plausible explanation. He noted that in all rooms, the listener receives sound directly from the source and also reflected from the room surfaces.

In a small room, the level of the direct sound is almost always higher than that of the reflected sound and, therefore, dominates in the perception process. Because of directional characteristics of human hearing at high frequencies, largely due to head shadowing effects, less total sound energy enters the ears at high frequencies than at lower. This imbalance is perceived as normal. 

In a large room with typical acoustics, however, the opposite is true; the level of the reflected, or reverberant, sound is significantly higher than that of the direct at most listener locations. 

Since this reverberant sound arrives at the listener from all directions rather than just one, more of it enters the ears at high frequencies. Thus the highs are perceived as being louder.

A simple experiment tends to confirm this theory. A loudspeaker is located at head level in a relatively non-reverberant environment and fed with broadband noise. A listener stands one to two meters (about three to six feet) in front of the loudspeaker and slowly turns around while listening to the tonal character of the noise. Typically, the overall tonal balance will change little, if at all, with head direction. 

However, if two identical loudspeakers are placed two or three meters apart facing each other and both are fed the same broadband noise, a listener between them, turning around as before, will hear the high frequencies more loudly when his ears are toward the loudspeakers than when he is facing one or the other loudspeaker.

The measured response (and perceived timbre) of a loudspeaker in a room deviates significantly from its performance in an anechoic environment, in ways that are complex and quite difficult to predict. Also, these deviations are different at each location in the room. Therefore, the only practical solution is to measure the actual response of the completed system and correct it as needed with additional circuitry.

This turns out to be a bit trickier than one might expect, however. If a pure tone, slowly swept in frequency, is fed over a sound system and the resulting level is measured at a point in the audience area, it will be found to consist of strong peaks and valleys, tens of decibels in amplitude, and spaced at intervals of about 1 Hz, caused by room resonances.

It’s almost impossible to get meaningful information from such readings. Besides, we don’t perceive these variations because they are averaged by our hearing process in ways that are only partly understood. The measurements must incorporate averaging which simulates the hearing process.

Making Assumptions
However, this presents us with a shopping list of unanswered questions pertaining to the measurement techniques. What frequency resolution (bandwidth) is needed? A first assumption might be to use a bandwidth similar to that of the auditory (critical bandwidth) filters, but system measurements are typically done with third-octave filters, which are considerably wider than critical over much of the spectrum. 

Should the analysis be done with a swept filter, which yields more information, or is a stepped filter technique acceptable? What amplitude smoothing or averaging is appropriate? How many measurement locations should be taken, and where should they be located? And exactly how should the individual measurements be averaged to yield the overall system response? 

Despite countless practical field experiments in this area, beginning at least 65 years ago, little critical research has been carried out. As a result, there exist only a few de facto standards, and the actual results of these procedures vary considerably in quality.

In addition to the these considerations, it might be expected that nonlinear distortion in any of the system’s components, especially the loudspeakers, would significantly affect its timbre, but such does not seem to be the case. The distortion levels of modern components, properly used, are low enough to be unnoticeable in a reinforcement situation. 

Intelligibilty. As the name suggests, intelligibility is the measure of how easy or difficult it is to understand speech over a system. It’s ultimately measured subjectively and directly, typically using rhyming words as the test signal.

The execution of this test is tedious and time-consuming with only one test subject, which is quite inadequate. Different subjects will render somewhat different results even under apparently identical conditions, and conditions vary significantly with location, program sound levels, room noise, hearing acuity, and many other factors. 

The typically broad variance of test results makes it difficult to determine whether a system is actually performing acceptably or not. It hardly seems worth the rather considerable effort required to execute such a test, but there may be little choice.

Because of these difficulties, a lot of effort has gone into devising an objective test regime, with several products resulting. All these involve dedicated gear and techniques, which, while not simple, are quite preferable to subjective tests.

These objective tests have been demonstrated to produce results comparable to those obtained subjectively in some, but not all, conditions. Unfortunately, the worst correlations tend to occur in conditions that produce low scores, exactly where accurate results are most desired. In fact, after extensive experience with all the commonly used objective techniques, Mapp has concluded that all are inadequate.

More Physical Approach
It gets worse. Low intelligibility scores, which indicate serious problems, usually provide little or no information on the nature of these problems.

Sometimes one or more physical problems are apparent in such cases, but are these really the causes of the poor performance?

Often, the only way to be sure is to correct the problems and see if that improves the scores.

Of course, this may be completely impractical, and in fact, there may be multiple problems, some masking others, so that correcting the most obvious might accomplish nothing useful.

A much more practical approach might be to identify exactly which physical factors adversely affect speech intelligibility, and how, and calibrate physical measurements to subjective effects.

If this were accomplished, then not only would meaningful test methods be available, but effective design criteria could be established to predict results and avoid problems in the design stage.

Some significant work has already been done in this area, with results pointing to the ratio of direct to reflected (or reverberant) sound being the most important factor.

Bob Thurmond is principle consultant with G. R. Thurmond and Associates of Austin, Texas.