Article

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.

Crown Audio Begins Shipping I-Tech HD DriveCore Multichannel Power Amplifiers

Crown Audio today began shipping its new flagship I-Tech HD 4x3500HD DriveCore Series 4-channel power amplifier.

The 4-channel I-Tech 4x3500HD incorporates Crown’s sixth-generation Class I engine with DriveCore technology to deliver 3,500 watts per channel burst into two ohms, and offers a host of Crown’s most advanced features including a 4.3-inch TFT LCD with capacitive touchscreen, exclusive Harman HiQnet System Architect 2.0 control functionality and V5 DSP preset support for JBL Professional’s newly-launched VTX Series V25 plus VERTEC Series line array loudspeakers.

The Crown I-Tech 4x3500HD delivers 1,900 watts per channel into eight ohms, 2,400 watts per channel into four ohms and a massive 4,800 watts into eight ohms bridged with all channels driven at full bandwidth all from an amplifier that measures just 2U rack spaces high. This remarkable power to size ratio is made possible by Crown’s exclusive DriveCore amplifier IC chip. The DriveCore chip combines the amplifier driver stage into the power output stage (along with additional audio-signal functions), to dramatically reduce overall size and power-consumption requirements and yield energy-efficient operation that conforms to Harman International’s GreenEdge environmental initiative.

“The I-Tech 4x3500HD DriveCore Series sets a new benchmark in control, connectivity, DSP and system interfacing capabilities for professional high-power multichannel amplifiers,” said Brian Pickowitz, Market Manager, Tour Sound for Crown. “We are confident customers will immediately recognize the distinct benefits of a true 4-channel solution that this amplifier offers for a wide range of live sound and fixed installation applications.”

The I-Tech 4x3500HD provides an ideal real-world solution with its four analog inputs, four AES3 digital inputs and four AES inputs over VDrive and the ability to select four CobraNet inputs. The amplifier also includes SpeakON or banana plug speaker connectors, as well as a Neutrik PowerCON AC input connector to prevent the power cord from coming loose in transit.

The Crown I-Tech 4x3500HD provides more DSP sound-tailoring capability than any other amplifier on the market. Its proprietary BSS OmniDriveHD processing engine employs 32bit/192kHz A/D and D/A converters for superb sonic clarity and the ability to precisely tailor the amplifier’s audio output. Crown’s exclusive linear phase FIR and IIR filters provide optimized loudspeaker crossover points with improved midrange clarity and off-axis loudspeaker response.

The amplifier is compatible with the Harman HiQnet System Architect and JBL HiQnet Performance Manager sound reinforcement system design software. The inclusion of JBL VerTec V5 DSP preset tunings for VerTec Series loudspeakers interface with the I-Tech 4x3500HD’s FIR filters to improve the loudspeakers’ sound quality and horizontal coverage performance. For enhanced control and monitoring capabilities the amplifier is also compatible with the Powered By Crown iPad/iPhone application.

The I-Tech 4x3500HD incorporates a host of additional useful features including Crown’s innovative LevelMax limiter technology that combines the operation of the amplifier’s Peak, Thermal and RMS limiters for more effective protection; and a front-panel USB port that enables users to load preset amplifier settings or device files and update firmware.

Crown Audio

Manhattan’s LQ Nightclub Pulses To The Sound Of D.A.S. Audio

The Latin Quarter—commonly known these days as LQ Nightclub—has a storied history that dates back to 1942 when Lou Walters, father of renowned TV journalist Barbara Walters, first founded the popular nightspot. While much has changed over the years, there remain two constants: the desire to have superior sound quality and vibrant entertainment.

Recently, the popular club upgraded its sound reinforcement capabilities and, to ensure world-class audio quality for both the performers and the clientele, a new loudspeaker system featuring D.A.S. Audio Aero Series 2 line arrays.

North Bergen, NJ – based E&A Sound LLC, a design/build firm that in addition to audio installations also handles live sound reinforcement projects for both the pro touring and special events markets, was contracted to handle LQ Nightclub’s recent sound system upgrade. Edwin Diaz, the firm’s owner, discussed the nature of the project and his decision to deploy D.A.S. Audio’s highly regarded Aero 12A powered, two-way, mid-high line array modules.

“The Latin Quarter, which is now located in the Radisson Lexington Hotel at 511 Lexington Ave at 48th in Manhattan, has multiple zones that needed to be properly covered as part of this installation,” Diaz explained. “In addition to the main dance floor, there are two VIP sections plus a bar area. In all cases, we were looking for compact loudspeaker enclosures that delivered live performance audio quality and sound pressure levels while maintaining a relatively small footprint. The club’s owner was aware of the D.A.S. installation we handled for the Luna Nightclub in the Bronx, which also used D.A.S. Aero 12A as well as Convert 12A loudspeakers. After visiting the club and hearing the system, he gave us the green light for the LQ project.”

At the LQ Nightclub, Diaz and his crew deployed a total of 16 D.A.S. Audio Aero 12A loudspeakers. Eight Aero 12A’s handle the main dance floor—with four enclosures flown per side. In addition to these loudspeakers, the adjacent bar area has two Aero 12A’s for fill purposes while the lower level VIP area also has two Aero 12A’s. There is a second VIP area located upstairs and here, too, Diaz and his assistants deployed another four Aero 12A’s. In all cases, the Aero 12A loudspeakers are flown and are supported by D.A.S. Audio AX-Aero12S rigging bumpers. Subwoofers are a proprietary design.

Diaz elaborated on his reasons for deploying D.A.S. Audio’s Aero 12A loudspeakers, “The Aero 12A outputs exceptional sound quality and SPL levels given the surprisingly small size of the enclosure. Speech intelligibility is excellent and the loudspeakers are very musical in nature. I also love the fact that these loudspeakers are self-powered because this not only simplifies system cabling, it also saves considerable space. In the case of the LQ Nightclub, we were able to eliminate three equipment racks that used to reside in the DJ booth since we no longer needed to house power amplifiers.”

In addition to the LQ and Luna nightclubs, Diaz has installed D.A.S. Audio loudspeaker systems at the Monte Carlo Room in the Bronx. “At Monte Carlo,” Diaz reports, “we installed a combination of D.A.S. Aero 28A loudspeakers along with LX-218A subwoofers. In each of these installations, D.A.S. Audio’s customer and technical support services have been exceptional. When I have questions, I get straight through to people who understand what I’m looking to accomplish and get the information I need quickly. The company is very responsive to its customers. I’ve been working with D.A.S. for a good 15 years or so and the company continues to impress me.”

The LQ Nightclub project was installed and placed into service in February and, since that time, Diaz reports it’s been smooth sailing on all fronts. “The new loudspeaker system is working out really well,” he says. “The various bands and DJ’s that work there have been very complimentary, as have been the club’s customers. The owner has told me on numerous occasions how impressed he is with the sound and this has led to new business opportunities. There are plans to add additional loudspeakers in two more areas at the club. When a project leads to new business—as it has in this case—that’s as good as it gets!”

D.A.S. Audio

Thursday, May 03, 2012

Low-Voltage Audio Products: Power & Noise

This article is provided by Rane Corporation.

 
This is an installment in a multi-part series. Additional segments are available here.

Noise
Low noise and low voltage don’t like each other.

Low voltage usually means portable, and portable always means low current to prolong battery life. You can design low noise and low voltage if you can be a current pig, but if you must have low noise, low voltage and low current—well, that’s difficult.

Everything works against you. The easiest way to make a really low noise op amp is to run as much current as possible through the front-end differential-pair until the silicon glows.

As unintuitive as it may be, a plain resistor, hooked up to nothing, generates noise and the larger the value the greater the noise. It is called thermal noise or Johnson noise (John Bertrand Johnson first observed thermal noise while at Bell Labs in 1927, publishing his findings as “Thermal agitation of electricity in conductors,” Phys. Rev., vol. 32, pp. 97-109, 1928), and results from the motion of electron charge of the atoms making up the resistor.

All that moving about is called thermal agitation (caused by heat—the hotter the resistor, the noisier).

Therefore quiet designs should use small resistor values, but, alas, small resistor values draw large current, and there goes the battery life. Compromise must ensue.

It is difficult to find the perfect balance between small resistor values for low noise and large resistor values for low current consumption. To make it even harder, with most analog circuits small resistor values mean correspondingly large capacitor values.

Large capacitor values do not hurt the noise performance but they are physically large and cost more, so you must make a compromise between noise, space and cost (analog design is like that).

The choice of resistor values then becomes the deciding factor in selecting the right op amp for each application. Look at the resistor values; if they are very small (like in a mic preamp) then the noise contributed by the op amp becomes critical.

However, if the application is active filters, say, and the resistors surrounding the op amp are at least 10 k ohm, then the dominate noise factor becomes their thermal noise, not the op amp’s noise. Understanding this simple fact allows you to use low-cost op amps for most of your needs.

Ultimately the performance gets down to how much voltage is available and how low is the noise floor: power supply and noise—the big two in designing quality audio for IAs.

Power Supply Design
Successful IA audio circuits begin with power supply design. Designing low-voltage audio circuits for portable and wireless information appliance products puts severe restrictions on quality.

Sacrifices necessary to keep cost, size, and weight to a minimum often hurt audio quality.

Portable and wireless devices force audio designers to work with very small supply voltages, often just a single 1.5-volt cell. There is just one rule when designing quality audio circuits if you only have 1.5 volts to work with: make more voltage.

Separate Audio Supply
No matter what the voltage, in order to achieve very high performance levels, audio circuitry must run from dedicated supplies.

Obviously it does no good to select the lowest noise op amps if they are connected to a digitally corrupted power supply.

Single-Supply Design
If the design cannot justify split-supply costs then you must design with a single supply. Since audio is an AC (alternating current) signal, its voltage swings positive and negative about some reference point.

This reference point is normally ground (or common) for a bipolar or dual power supply, i.e., one with positive and negative voltages (e.g. ±15 VDC). If you only have a single supply then you must create a reference point equal to one-half of the available supply.

For example if you have a single 5 volt supply then you create a common reference point at 2.5 volts, which allows the audio to swing ±2.5 volts (from the reference point up 2.5 volts to the +5 volt limit and down 2.5 volts to zero.

Splitting a single supply voltage is not difficult, nor expensive (although in some designs every extra op amp or resistor can mean trouble).

Techniques exist ranging from a simple two-resistor voltage divider to more elaborate buffered op amp designs. Excellent application notes covering all aspects of this topic are available from Texas Instruments, Linear Technology, and Analog Devices.

DC-DC Converters
If the hand you’ve been dealt contains only one AA cell battery then you must become a DC-DC converter designer at once. Luckily there is lots of help in this area. There’s nothing you can do with a single AA battery except use it to create more voltage.

How much voltage depends on the product and the application. If you must create loud audio into big speakers, then life’s going to be a lot harder than if you can get away with driving only headphones.

Low efficiency loudspeakers and headphones are a big obstacle to pristine IA audio. Low efficiency means you need lots of power to drive high-quality speakers to loud levels. And lots of power means lots of voltage and current.

If it is your choice, then chose a pair of nice clean and quiet split supply voltages—as high as you can get them for loud results or if you are going to interconnect with the pro audio world. Most pro audio products use ±15 VDC for their analog audio circuits.

While finding a single IC capable of converting 1.5 VDC to a nice clean and quiet ±15 VDC is difficult (see LTC Design Note) to impossible, several IC companies make converters that will pump up 1.5 volts to 12 volts, and from there you can split that into a useable ±6 VDC. See for instance Analog Devices or Linear Technology, or also Linear Technology.

See also Linear Tech’s latest free design software for DC-DC converters, although it doesn’t help much for single cell converters.

Another free helpful DC-DC converter design program is available from National Semiconductor named Switchers Made Simple , and take a look at the collaborative venture by National, Vishay, and Pioneer-Standard Electronics called Webench , a free on-line tool to design, simulate and order prototype kits for power supplies.

And not-for-free from ON Semiconductor is Power 4-5-6 software for the design, simulation and analysis of power topologies.

Op Amp Specifications Important For Audio
Selecting op amps for audio is a lot easier than it was the first time I wrote about this topic in 1976 (Audio Handbook, National Semiconductor Corporation, 1976. The reprinted version is the last revision published by National Semiconductor in 1980, compiled and edited by Martin Giles who took over as compiler and editor after I left in 1976. Order copies from Old Colony Sound Lab) .

This is primarily due to the quantity of audio specific chips sold into the automotive and PC industries.

Quantity is what IC companies understand. They live and die by quantity, and for the first two decades, audio was pretty much ignored as a product line. Back then selecting good audio op amps took some digging and required the designer to know quite a bit about audio’s specific requirements.

Things are different now. Audio-grade op amps are sold by the millions each day, and it makes selecting them a lot easier since most IC companies have a separate section in the selection guides for audio.

Here is a summary of the most important parameters (in no particular order):

• Gain-Bandwidth Product, or GBW, equal to at least 3 MHz. This gives plenty of open loop gain (>40 dB) for feedback circuits to still work well at 20 kHz. More is better as long as the phase margin does not get compromised. You want to see a solid phase margin of 60 degrees at the unity gain BW crossing point.

• Slew Rate, or SR, equal to at least 1.5 V/microsecond. This value is necessary to prevent slew-limiting at 20 kHz with full output voltage. In a single-cell world you never have large voltage swings so you never need large slew rates, but it’s nice to have some margin.

• Noise, or Noise Density: normally specified at 1 kHz, along with a graph showing wideband performance. Look for spot noise density at 1 kHz less than 15 nV per square-root-Hz (approximately the noise of a 10-kohm resistor) for low gain circuits (like filters) and less than 4 nV per square-root-Hz (noise of a 1-kohm resistor) for high gain circuits (like mic preamps).

In addition to a low 1 kHz spot noise number, you want to see a low 1/f corner, i.e., you don’t want the low-frequency noise to start rising dramatically until below 20 Hz.

• Total Harmonic Distortion + Noise, or THD+N: This is not a spec to get overly concerned with. As long as the part stays out of whole numbers, you probably don’t have to worry about any audible results. But in the interest of successful marketing, select parts with a THD+N less than 0.1% over the entire 20 Hz - 20 kHz audio range. Today it is very hard to find parts that don’t shine in the THD department.

Low noise, high slew rates, wide bandwidths, and excellent linearity (low distortion) characterize high quality audio op amps. Other important specifications are application driven and include power supply voltage, current consumption, common-mode rejection, power supply rejection, input impedance and size.

The Audio Handbook (see above) describes op amp audio requirements as follows: “The IC must process complex AC signals comprised of frequencies ranging from 20 Hz to 20 kHz, whose amplitudes vary from a few hundred microvolts to several volts, with a transient nature characterized by steep, compound wave fronts separated by unknown periods of absolute silence.

This must be done without adding distortion of any sort, either harmonic, amplitude, or phase; and it must be done noiselessly—in the sun, and in the snow—forever.” Nothing has changed.
Selecting Low-Voltage Op Amps

Good audio requires good parts. Low-voltage information appliances make selecting the right audio ICs even more important—and more difficult. What follows are guidelines and pointers to high-quality audio ICs specifically designed for low voltage designs.

Note: There are too many world wide semiconductor companies to be all-inclusive regarding recommendations. Apologies are made to those left out. The author knows the ICs and companies spotlighted from direct experience. Omission of any company or specific products merely means the author was not aware of them. It is also recognized that many of the ICs mentioned will be outdated immediately upon writing, so always check the manufacturer for the latest part replacing or improving the one discussed.

Stay tuned for the coming articles in this series. Want to get a jump on the reading? Head on over to the Rane Website where you can read this article in its entirety.

Supplied by Rane. For more, go to rane.com

Brooklyn Academy Of Music Chooses QSC Audio KLA System For BAMcafé

BAMcafé, a restaurant and live performance venue located within the main Brooklyn Academy of Music (BAM) building, recently installed a QSC Audio KLA active line arrays to support as many as 75 music, comedy and spoken word events it presents annually in its free BAMcafé Live series.

See Factor Industry, Inc., a touring and installed lighting and sound equipment provider, was tasked with providing a system which sounds great at a reasonable price point.

“BAM came to me asking for a recommendation, as I had done the installation in the Opera House there,” says Greg Wnuk, audio department manager/special events at See Factor. “BAMcafe needed a powerful sound solution which was also cost-effective, and KLA fit their price point while sounding great.”

Such was Wnuk’s confidence in the QSC brand that he recommended the new KLA system sight (and sound) unseen.

“I hadn’t heard the KLAs yet, but I was pretty confident they were going to sound good because the QSC stuff sounds good,” he notes. “This client was aware that I hadn’t heard the KLA yet—but they trusted me to trust QSC. The first time I got to hear them was we did the KLA demo at BAM, and we were all very impressed.”

Originally conceived as BAM’s ballroom and completed in 1908, the room was renovated and renamed the Lepercq Space in honor of the chairman of the board in 1973. BAMcafé opened in the BAM Lepercq Space in 1997 and launched its BAMcafé Live weekend programming two years later.

Depending on the room setup the venue holds between 160 (dinner and dancing) and 325 (standing) people. The KLA12’s are flown above the performance area located at one end of the room, which measures approximately 117 feet by 42 feet, and offer coverage all the way to the bar area, which is 75 feet down the room..

“The KLA system really has taken the venue to the next level of sound,” says Josh Escajeda, BAM associate production manager. “We do so many different types of events, from weekly live bands, to opening night receptions, speaking agendas, film screenings to book readings—we really needed something that could be as versatile as the space.”

“We were constantly pushing our old system to its limits,” adds Escajeda. “Now with the KLA system, we have plenty of headroom. Just testing it, we surprised people in the offices above the space, because they could hear the sound in their office. It was awesome. And the customer service from QSC has also been great.”

BAM, which is currently celebrating its 150th anniversary, is America’s oldest performing arts center, and has hosted wide variety of celebrities, from performers to presidents, during its years of service.

Gala at the Brooklyn Academy of Music BAMcafe. (Photo credit: Pascal Perich)

See Factor Industry, Inc.
QSC Audio

St. Leander Updates Sanctuary With Tannoy Qflex

St. Leander Catholic Church in San Leandro, California, recently reconfigured their sanctuary to bring the worship experience closer to their members. The redesign included the installation of a new sound reinforcement system featuring Tannoy Qflex array loudspeakers.

Leo’s Professional Audio, based in Oakland, California, was asked to assist in designing and installing the new sound system.

“The redesign involved moving the alter from the back of the rectangular sanctuary to the center of one of the longer walls,” explains Graham Cooper, vice president of the contracting division at Leo’s Professional Audio. “The pews were moved to create semi-circular seating around the new alter area. As a result, the new system was designed to cover an area that was much wider than deep.”

The sanctuary, approximately 60’ x 120’, was a “vast cavernous space” with all of the hard, reflective surfaces one would expect in an older, traditional sanctuary. It was imperative to steer the sound to the seating areas and minimize any bounce back from the hard surfaces.

Cooper placed two Qflex 24 array powered loudspeakers to the left and right of the alter. The structure of the alter platform and the location of the choir area stage right dictated the placement of the loudspeakers at 15-feet above the floor.

The Qflex 24s, just under 5-feet tall with less than a 7-inch width and depth, were painted to match the décor of the church making their presence barely noticeable. With a horizontal dispersion of up to 120 degrees, Cooper was able to focus the output to cover the majority of the seating area and maximize the vocal intelligibility of the system.

“Because the room was very wide, there were a few seats in the sanctuary that were not covered by the Qflex system ,” adds Cooper. “To ensure every seat was a “good” seat we installed four Tannoy Di5DCs spread out along the front wall so that they would hit the areas at the far ends of the room as well as right under the Qflex lousdpeakers. The end result was very impressive.” 

The original alter area, set off from the main sanctuary, was converted to additional seating which required extra audio support. A single Tannoy i7 line array column bracket-mounted to an adjacent column, provides ample coverage.

A nearby confessional turned rack room is home to a Lab.gruppen 5.4x amplifier to power the Di5DCs and i7 boxes. The rack includes a Rane digital signal processor for delay and EQ supplemental and two Rane auto mixers control the microphones and CD player. To provide easy on/off of the system and minimal interference to system set-up, Cooper used a Lyntek PD510-4 sequenced power system with the on/off panel located outside of the locked rack room.

“The system is literally push and play,” he explains. “They don’t even have to open the rack room to get everything up and running. The coverage is outstanding and the vocal intelligibility better than ever before. The parishioners are absolutely delighted with the new system.”

Tannoy

Wednesday, May 02, 2012

RTI Now Shipping New CP Series Multichannel Power Amplifiers

Remote Technologies Incorporated (RTI) today announced that the company is now shipping its new four-channel CP-450 and 16-channel CP-1650 audio amplifiers.

Designed to complement RTI’s eight-zone AD-8 distributed audio system, these new amplifiers provide an additional 50 watts per channel to extend speaker outputs in larger installations.

“Our AD-8 distributed audio system provides 25 watts per channel, which is ideal for most installations, but larger rooms and subzones might require more speakers and more power,” says Pete Baker, vice president of sales and marketing for RTI. “We are pleased to offer our dealers a powerful, cost-effective solution to this installation issue with our new CP-450 and CP-1650 amplifiers.”

For optimal performance, the CP-450 and CP-1650 feature the same proprietary Cool Power technology used by the AD-8’s built-in class D amplifier. In addition, the CP-1650 offers audio input level adjustment on each channel, unique loop-in/loop-out connections for simple daisy chaining, and bridgeable audio outputs for increased power to a whopping 100 W.

For added flexibility, the CP-450 and CP-1650 amplifiers are suitable for rack-mount or freestanding installations. Convenient power control from different sources is made possible with voltage trigger, RS-232, and IR power control options. For situations that require more than one amplifier, voltage trigger and IR pass-through allow for control of an additional unit.

The CP-450 and CP-1650 are now available at MSRPs of $699 and $1,499, respectively.

Remote Technologies Incorporated (RTI)

Tuesday, May 01, 2012

Community’s VERIS Is Part Of Classic Church’s Rebirth

Founded nearly half a century ago in Northampton Township, Saint Bede the Venerable has grown to more than 10,000 parishioners. The church recently underwent a $5 million renovation and expansion, incorporating a classic Italian marble altar, magnificent stained glass, and a new sound system based around Community VERIS 8 two-way full-range systems.

As Audiobahn’s Tony Hersch explains, the new sound system design called for versatility. “Unlike some of the more traditional Catholic churches, St. Bede offers a mix of traditional and semi-contemporary services, with a choral and live musical accompaniment, as well as traditional organ and choir,” he says. “So ultimately we had to design a system that offered more than simply spoken word intelligibility - it had to offer good musicality as well.”

The distributed audio system includes 14 VERIS 8 cabinets - seven on each side of the seating area, as well as two additional VERIS 8 boxes for platform monitors. A pair of Community CPL27 dual eight-inch systems provide monitoring to the choir loft, and two MVP12M boxes provide additional monitoring. The system is powered by QSC amplification. A Soundcraft LX7II 16-channel console and Shure RF systems complete the signal chain.

One of the biggest challenges in installing the system, says Hersch, was coordinating the work around the ongoing construction. “It was a pretty extensive renovation,” he says. “They brought in the altar from another church in Philadelphia that had closed its doors, with massive marble columns, and gorgeous stained glass windows. They were still holding services during the renovation, with the construction behind a false wall, and we were running cable and installing the speakers within a very tight schedule, alongside the general contractors.”

Hersch reports that the newly redone sanctuary is nothing short of stunning, with sound to match. “It sounds fantastic, and the coverage is great,” he says. “There’s not a bad seat in the house. And it looks terrific.”

Community

Hosa Technology Introduces CBT-500 Cable Tester

Hosa Technology is pleased to announce the introduction of the CBT-500 Cable Tester.

Designed to quickly and easily check a variety of cable types, the new CBT-500 Cable Tester enables one to verify the integrity of a cable before interconnecting equipment.

The CBT-500 makes an invaluable addition to one’s ‘tools of the trade’ and is ideal for use when preparing for a concert, a studio recording session, or an installation as well as checking cables afterwards in order to ensure working operation the next time.

The new CBT-500 Cable Tester makes testing a wide range of cables commonly encountered in the music / pro audio environment a snap. The compact, handheld unit is capable of testing the following cable terminations:

• XLR (3-pin and 5-pin)
• Balanced and unbalanced phone (standard guitar-type cable)
• Phono (RCA)
• speakON® (with support for 2-pole, 4-pole, and 8-pole connectors)
• DIN (multi-pin, including 5-pin DIN commonly used for MIDI equipment)
• Ethernet (RJ-45)
• USB Type A to Type B (standard USB to the square USB connector found on most audio interfaces)

In addition to providing the necessary connectors to test the various cable types, the Hosa CBT-500 Cable Tester is also easy to use, consisting of a front panel rotary knob and two rows of 8 LEDs—one row amber colored, the other row green. These LED’s facilitate easy checking of the individual pins found in any of the supported cable types. The rotary knob enables one to switch from one pin to the next. As an example, by connecting both ends of an XLR cable to the unit, one can systematically check pin 1 to pin 1, pin 2 to pin 2, and pin 3 to pin 3. If both LEDs illuminate, the connection is passing signal. If not, the connection is broken.

The new CBT-500 is also equipped with removable leads—commonly referred to as test probes—to verify the continuity of additional connectors and jacks. This capability is particularly convenient when making cable repairs, as it enables one to check the actual replacement connector.
The Hosa CBT-500 Cable Tester is constructed of metal to withstand field abuse and operates with a standard 9-volt battery (included). The device also provides a battery check function to ensure proper working condition prior to use.

Jose Gonzalez, Hosa Technology’s Product Manager, commented on the new CBT-500 Cable Tester, “The CBT-500 is an incredibly versatile and capable tool that no electronic musician or audio professional should be without. With support for most connector types that one is likely to encounter, this rugged, handheld tester makes it easy to know whether a particular cable or connector will pass signal—before it becomes a problem in the field. With the included test probes, continuity testing of additional connectors, jacks, or bare wire is equally quick and easy. I’m confident our new cable tester will help people avoid signal transmission failures in their work.”

The new Hosa CBT-500 Cable Tester carries an MSRP of $69.95. The unit is expected to be available May 2012.

Hosa Technology

Rebuilt Spa Resort Hawaiians In Japan Features Concert System Headed By QSC Audio ILA

Spa Resort Hawaiians, a family water resort in Fukishima prefecture, Japan, had not long completed an extensive refurbishment, including the commissioning of a new sound reinforcement system headed by QSC Audio Installation Line Array (ILA) loudspeakers, when the terrible earthquake and tsunami of March 2011 wreaked havoc in the area.

The resort had to close, but reconstruction work began almost immediately, and the ILA is now back in full service on the venue’s recently completed entertainment stage.

Japanese QSC distributor Onkyo Tokki Ltd (OTK) was responsible for the original ILA installation as well as the reconstruction work. The entertainment area at the spa, which hosts evening Hula dance and Polynesian live music events for the spa’s customers, is located behind an indoor lido pool area, and is surrounded on three sides by two-tiered audience seating with a maximum capacity of 700 people.

Prior to the refurbishment, the resort’s management was unsatisfied with the sound of the original PA. Due to the expansion of the entertainment area over the years, the existing 14-year-old system no longer provided an even dispersal of sound across the seating area. The managers knew it was time for a system capable of achieving consistent coverage both laterally and vertically.

OTK recommended the QSC ILA as both the most efficient and cost-effective solution for the resort. As the ILA provides an impressive 140-degree horizontal range of coverage, no out fill or center clusters were needed, further minimising cost and complexity.

OTK decided to install 12 WL2082-i loudspeakers and two pairs of WL118-sw subwoofers. A dedicated SC28 system controller was chosen to optimise the performance of the system, in which the speakers were driven 4-way (3-way plus subs) by a combination of QSC PowerLight PL325 and PL340 amplifiers.

Installed in two arrays mounted five meters from the ground, one on either side of the stage and each consisting of six WL2082-i elements and a pair of WL118-sw subs, the stage sound system succeeded in reproducing on-stage sound with vastly improved clarity and presence, delivering it evenly to the entire coverage area— even the rear seats on the top tier, which are around 30 meters away from the twin QSC arrays.

Following the earthquake, reconstruction at Spa Resort Hawaiians took just under a year. According to OTK, the goal of the reconstruction work was to restore the systems at the resort to the state they were in the day before the disaster, and this was achieved by the time the resort reopened in February. To the delight of OTK’s engineers, when the structural works on the resort following the earthquake were complete, the QSC ILA and its constituent elements were found to be in full working order, requiring no repairs.

QSC Audio

New System Application Profile: Desert Episcopal Church Of Scottsdale, AZ

Church sanctuaries are often aesthetically beautiful spaces. Church sanctuaries are often acoustically challenging spaces. Navigating between these two extremes is the sound system designer.

It’s a path recently tread by Rod Andrewson and the team at CCS Presentations Systems in formulating and implementing a new sound reinforcement system at St. Barnabas on the Desert Episcopal Church in Scottsdale, AZ.

Built in 1960 under the direction of noted architectural firm of T.S. Montgomery, the 350-seat sanctuary features columns and arches, a high ceiling, and a raised circular altar with choir risers and a pipe organ located directly behind it.

“The sanctuary was originally designed to have the acoustic principles and characteristics of larger cathedrals,” explains Andrewson, who serves as chief engineer at CCS. “As a result, it’s a highly reverberant space that had been plagued with vocal intelligibility issues for decades. Over the years a number of acoustical treatments had been applied in an attempt to resolve the problem, with little to no success.”

The system project was part of a complete renovation of the sanctuary that was spurred by the donation of a new organ, with an overall goal of returning the room to its original form. As a result, church leadership wanted worship services to feature organ music, but they also wanted a system to deliver dramatically improved vocal intelligibility.

An added caveat was that all acoustic treatments would be removed during the renovation with the intent of making the room as reverberant as possible to further enhance the organ music. The sanctuary has a single level of seating in an area measuring roughly 60 feet wide by 100 feet deep under a ceiling that reaches 20 feet.

The sanctuary is designed to have the reverberant of a much larger space, with main loudspeakers barely visible on columns at far left and right. (click to enlarge)

Acoustical Imprint

CCS, also based in Scottsdale, was one of a handful of firms invited to submit a bid on the project. “When we were told that the sanctuary would be stripped of
all acoustic treatments, I felt like we had a real challenge on our hands,” Andrewson notes. “As a matter of fact I contemplated not having anything to do with the project at all. But before walking away, I
wanted to do a little research.”

He had successfully implemented Tannoy Qflex digitally steerable arrays in other applications, and thought they might provide a solution in this application. The straightforward goal was focusing as much sonic energy on the audience as possible while keeping it off of the bountiful reflective surfaces.

Qflex also has the added advantage of a slim profile, and the cabinets and grills can be painted, helping ease concerns about aesthetics. The CCS team arranged a demonstration of a Qflex array within the sanctuary, and the client came away impressed.

“We showed them that we could very precisely divine and steer the acoustical imprint on the audience,” Andrewson explains. “When we demonstrated the Qflex loudspeakers and VNET software to our client, it was very clear that none of our competition had even considered doing the same. I think that demo was, hands down, what won us the job.”

The system design is led by two Qflex 40 self-powered loudspeakers mounted on left and right columns about 20 feet in front of the altar, the best option available to covering the majority of seats. Each loudspeaker, measuring 83 inches high by just seven inches wide, is loaded with eight 4-inch LF drivers, sixteen 3-inch LF drivers and sixteen 1-inch HF drivers. Every driver has its own discrete amplification channel.

A key part of the equation is VNET, Tannoy’s proprietary digital processing and network protocol, hosted on a PC located in the system’s rack room which can be found behind the back wall of the sanctuary. VNET is linked to the loudspeakers via a VNET USB and RS232 interface.

With VNET, both main loudspeakers are acoustically profiled and individually optimized, taking into account their exact positions within the venue relative to the room boundary and acoustic properties of the room. A control panel allows VNET parameters to be viewed and adjusted, including overall system status as well as specific component performance parameters, while the control area provides access for adjustment of all loudspeaker DSP parameters including crossover/delay, EQ, gain and power.

Additional Aspects

While the Qflex/VNET tandem handles the vast majority of coverage needs, there are still a couple of regions requiring support. Tannoy Di5 compact loudspeakers are discretely mounted left and right (and also painted to match) to provide coverage to the narthex in the very front area.

Two more Di5s, also column mounted, are utilized to extend coverage to the choir, which is located behind the pulpit, with one more for the pulpit for “preview and confidence” (a.k.a. monitoring). These loudspeakers are driven by eight speakers in three zones, two in the narthex, four at the pulpit and two for the choir.

QSC Audio ISA-280 amplifiers in the rack room, joined by a Biamp AudiaFlex, provides processing and delay for the support loudspeakers as well as overall system control and touring. Settings are locked down so they can’t be changed, except volume levels.

Vocal reinforcement is provided by Shure wireless transmitters outfitted with Countryman lavalier microphones.Six Shure ULXP receivers are rack mounted in the rack room to support the handheld and body worn transmitters. Both podiums are equipped with Clockaudio microphones.

A closer look at one of the Tannoy Qflex loudspeakers with slim profile concealing 40 transducers that have individual power and processing. (click to enlarge)

With the system largely devoted to spoken word reinforcement, there’s also no need for a mixing console. Audio levels can be adjusted via a wireless remote that interfaces with the AudioFlex platform, and this remote also works with the Crestron-controlled lighting system (also installed by CCS).

The rack room is also home to a plethora of video streaming gear. There are two additional buildings on the church campus that are utilized for other styles of worship, and via a NewTek TriCaster Studio and three Vaddio high definition pan tilt zoom cameras, can transmit video streams (via Ethernet) of the services to use within their worship program or to home bound parishioners.

The TriCaster is equipped with an offset tool that allows for a 6-minute delay, enough time for the A/V tech to optimize the audio via a Mackie 802- VLZ3 8-channel mixer and caption the
video with TriCaster’s video effects prior to distribution. Services are also made available for download via the Internet.

“Meeting the specific aesthetic and sonic requirements of this project was paramount, but it was all under one overall goal of the client – provide the absolute best solution for everyone to receive information from a variety of media sources,” concludes Andrewson. “That’s what we sought to deliver, and the client, as well as the patrons, indicate we hit the mark.”

Julie McLean Clark is a writer and marketing consultant working who has worked in the pro audio industry for more than 15 years.