Article

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

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.

Allan & Heath Flys The UK Flag At SXSW Festival

Allen & Heath’s iLive digital mixing system was selected by promoters, Cato Music, to manage FOH and monitors on the British Music Embassy stage at this year’s iconic SXSW festival in Austin TX, USA.

Showcasing the UK’s brightest new bands and artists, the British Music Embassy was exclusively based at downtown Austin’s Latitude 30 club, where an iDR-48 MixRack and iLive-T112 Control Surface was installed at FOH, digitally split using a Dante networking card to a duplicateiDR-48/T112 for monitors.

There were performances from nearly 60 bands over the course of the week-long festival, including emerging talent Django Django, fiN, Maverick Sabre, Benjamin Francis Leftwich, Jonquil, Ben Howard, Slow Club, Twin Atlantic, D/R/U/G/S, Skindred, and Clock Opera.

“I ran the desk in a very analog way, all inputs on the left banks, and master section on the right and some visiting engineers wanted things moved around but we always managed to get the bands ready in the 15 minute changeovers,” explains FOH house engineer, Fabrizio Piazzini. “Some band engineers had never used the desk before but were so stunned by the ease of use, sound quality and format of the system they promptly added iLive to their riders.”

“The festival patch included 37 channels but running iLive Editor meant I could drop all the channels for each performance on a single layer with just a few clicks,” Piazzini continues. “On the monitor side, we were running 5 mixes, with the odd band turning up with full IEM rigs. To ring out the wedges, we used the iLive MixPad app so we could actually EQ the system in the spot the artist would stand.”

The iLive system was supplied by Allen & Heath’s USA distributor, American Music & Sound, which also supplied Turbosound loudspeakers and beyerdynamic mics.

“I think iLive was a critical piece of the set up and it delivered every night,” commented Glen Rowe from Cato Music. “The British Music Embassy was a huge success this year and some sessions were so busy we set up more delays and opened the windows on the venue for people to hear outside!”

Allen & Heath

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.

Patrick Prothe Joins Biamp Systems As Marketing Communications Director

Biamp Systemshas announced that Patrick Prothe has been named the company’s new Marketing Communications Director.

In this role, Prothe will oversee Biamp’s marketing efforts worldwide with a focus on enhancing the company’s online presence, digital marketing assets and creative resources. Prothe will be based in Beaverton, Oregon and will report to Graeme Harrison, Executive Vice President of Marketing, Biamp Systems.

As Marketing Communications Director, Prothe will be responsible for Biamp’s strategic marketing direction, including developing and driving the global marketing plan, overseeing corporate communications, and directing the activities of the company’s external marketing agencies.

“With the continued growth in many of our global markets and the recent introduction of Tesira, it’s vital that our marketing efforts support the needs of our regions,” said Graeme Harrison, Executive Vice President of Marketing, Biamp Systems.  “Patrick is an ideal addition to our team because he understands the importance of closely aligned marketing and sales efforts. Patrick’s extensive experience and knack for creative vision will help us broaden awareness of Biamp and continue to strengthen our leadership position worldwide.”

Prothe comes to Biamp from Viewpoint Construction Software where he was Marketing Communications Manager, responsible for overseeing all marketing and client communications programs including social strategy, content development, public relations, media relations, and trade shows. Prior to this, Prothe served as Manager, Creative Services, at Knowledge Learning Corporation, where he managed corporate branding and creative direction. Prothe has also held various positions at Synesis Design, Xerox and WARN INDUSTRIES.

“Biamp has a solid industry reputation for providing the best support and excellent service to customers,” said Patrick Prothe, Marketing Communications Director, Biamp Systems. “Being able to join this team of smart, sharp-minded individuals is a true privilege. I look forward to working with Graeme and the rest of the Marketing team as Biamp continues to grow, and ensuring that our customers around the world are fully supported by our marketing efforts.”

Biamp Systems

Audio-Technica Debuts New ATH-ANC9 QuietPoint Active Noise-Cancelling Headphones

Audio-Technica today announced the introduction of its ATH-ANC9 QuietPoint active noise-cancelling over-ear headphones, the company’s new top-of-the-line active noise-cancelling (ANC) model.

The ATH-ANC9 offers new features including exclusive Tri-Level Cancellation selectable noise-cancellation settings, an inline microphone and controller for answering calls and controlling music, and additional enhancements.

The ATH-ANC9 blocks up to 95% of outside noise—the highest ANC performance ever achieved by Audio-Technica QuietPoint headphones, while delivering superlative sound quality.

Audio-Technica’s new Tri-Level Cancellation provides three preset filters for noise reduction of up to 30 dB over a wide range of environmental noise conditions that are experienced in everyday life.
Mode 1 is ideal for use on airplanes, trains and buses and applies maximum noise-cancellation to low frequencies. Mode 2 is designed especially for use in noisy offices and crowded places, and targets midrange frequencies. Mode 3 is best for already-quiet locations like libraries and creates a pristine, peaceful environment ideal for study.

The ATH-ANC9 is the first over-ear QuietPoint model to feature a cable with an inline microphone and controller for answering calls and controlling music. The mic and controller support select products including the iPhone(TM), iPad(R)and many iPod(R)models. The microphone has an omnidirectional pickup pattern (it picks up sound from all directions) and is designed for high-quality, intelligible response, enabling the wearer’s voice to be clearly transmitted without having to speak directly into the mic. The controller enables the user to play or pause music, answer and end calls, and go to the next or previous track.

The ATH-ANC9 has replaceable memory foam earpads for unmatched comfort, and is designed for the exceptional audio quality that Audio-Technica has offered for 50 years. Its precision 40 mm drivers and newly developed electronics provide clear, natural full range sound with authoritative bass, a detailed midrange, smooth, extended treble and precise imaging. The headphones offer an input sensitivity of 100 dB that will provide an ample listening level from portable music sources.

The ATH-ANC9 also works when the noise-cancelling function is turned off, and operates in passive mode without batteries.

The headphones fold flat for storage and come with two detachable cables (with and without inline controller), a 1/4-inch adapter, an airline adapter, a hard carrying case and an AAA battery.

The Audio-Technica ATH-ANC9 QuietPoint active noise-cancelling headphones are available now at a suggested retail price of US$349.95 at http://www.shopaudiotechnica.com, Best Buy Magnolia Design Centers, Airport Wireless stores and other select authorized retailers.

Audio-Technica

Friday, May 04, 2012

NSCA Presenting Digital Marketing Boot Camp In Las Vegas

The National Systems Contractors Association (NSCA) is hosting a two-day boot camp featuring keynote speaker, Scott Klososky, from the 2012 NSCA Business & Leadership Conference (BLC).

The boot camp will be held June 11-12 at the Las Vegas Convention Center, Las Vegas, NV.

Coming off of the most popular BLC, of which many topics centered around social media and marketing techniques, this boot camp will provide strategy-based curriculum on social and digital marketing strategies for marketing professionals and senior executives.

Attendees will understand the full breadth of resources available in the social media world in additional to the dynamics, processes and trends available today. Klososky notes that attendees will leave with “more knowledge to drive a noticeable, measurable impact on their organization’s bottom line and online marketing strategies over the next few years.”

The two-days will be spent on instruction and concept delivery in these key areas:

• Why social technologies are exploding, their role in today’s organizations and why leaders should pay attention;
• 15 unique social dynamics web 2.0 has delivered and you can apply to your organization’s strategy;
• Processes of social media implementation; and
• Future path of social technologies.

Klososky, a former CEO of three successful startup companies and current founder and Chairman of the Board of Alkami Technology, specializes in looking over the horizon with how technology is changing the world. His vision and ability to see trends in emerging technologies allow him to be a thought leader who applies his skills to help organizations thrive, leaders prosper, and entire industries move forward.

NSCA members receive this two-day boot camp at 40 percent off the original costs. Additionally, NSCA members can apply up to $400 in NSCA Education Credits towards the cost of the boot camp. 

The training includes meals, a USB drive with process implementation documents, tools and templates supporting the strategies discussed, and all the presentations for a total cost of $1,800 per participant.

For more information visit www.nsca.org/bootcamp or contact Bonnie Taylor, NSCA Events Specialist via email at .(JavaScript must be enabled to view this email address). 

NSCA

Thursday, May 03, 2012

3D Printing Capabilities Enhance Fishman R&D, Prototyping Processes

In recent years, Fishman has been growing while also expanding into new categories such as the Triple Play wireless guitar controller introduced at the 2012 Winter NAMM show.

The company has had to devise new ways to shorten the R&D and time-to-market process without compromising quality, with a major part of this initiative Fishman’s acquisition of an Objet 3D desktop printer so that it can rapidly create fully functional and true-to-life three-dimensional prototypes and parts.

The printer uses patented PolyJet Matrix 3D printing technology, which works by jetting special photopolymer materials in ultra-thin layers (16µ) layer by layer onto a build tray in a process managed by the company’s studio software until the part is completed.

Each layer is then cured (dried) by UV light immediately after jetting, which produces models that can be handled and used immediately.

Having its own in-house desktop 3D printer allows Fishman to go through multiple design iterations in a single day so the R&D department can quickly and efficiently finish new designs without missing windows of opportunity within the guitar market sales cycle.

“The Objet printer has had an incredible impact on our ability to be more predictable on time to market,” explains Ian Popken, Fishman director of product development. “It’s very exciting for mechanical engineers and others involved with the product to realize their ideas a few hours later.

“The fit and finish we’re able to achieve with these resins from the Objet is just remarkable,” he continues. “They so closely resemble the final product that we once took them to a trade show and no one could distinguish that they were final prototypes!

“So not only do we have the amazing cosmetics and the way that it looks on the outside, but also complete inside works and functionality of the product. When it comes time to do our pilot run, it goes very smoothly because we’ve already demonstrated that we can build the product exactly the way we want to instead of waiting for every part to come off tool. We are already ahead of the game because of the printer.”

In addition to shortening the R&D and production cycle, the Objet printer has proved to be a valuable tool for optimizing Fishman’s relationship with its customers.

VP of OEM sales Rob Ketch points out, “The parts we get from the Objet printer are of such good quality, we’re able to send them to customers after our shop has applied paint and graphics and they look very close to actual production parts. In the past, there would be such a long time between a drawing, stereolithography and a molded part; we would often miss a cycle for one of our customers.

“With Objet,” Ketch concludes, “they have a real example of the product, which can be very powerful in closing the sale. The Objet printer’s impact on the timeline and the impression that it makes in terms of parts and products has been really valuable to us at Fishman.”

Fishman

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

Line 6 Now Shipping XD-V55 Digital Wireless Systems

Line 6 is now shipping the new XD-V55 digital wireless handheld, lavalier, and headset microphone systems.

Featuring microphone modeling technology, XD-V systems offer 24-bit, 10 Hz–20 kHz, compander-free performance. .

The family offers a full complement of professional features including signal encryption, dynamic filters, gain control, channel scanning and more.

Utilizing the same 4th-generation digital wireless platform as the flagship XD-V75, the XD-V55 family offers handheld, lavalier and headset systems and the compact, portable XD-V35 family includes handheld and lavalier systems.

“For performers who want wired mic audio performance and wireless freedom, the combination of Line 6 modeling and our class-leading digital wireless platform makes the latest XD-V systems the only choice,” says Steve Devino, live sound product manager at Line 6.

“Proven on countless stages and tours worldwide,” he continues, “fourth-generation Line 6 digital wireless technology ensures the best possible performance experience with crystal-clear audio, rock-solid reliability and simple, license-free operation – worldwide.”

Ensuring faithful reproduction and full-range audio clarity, XD-V systems all provide 10 Hz - 20 kHz frequency response and wide dynamic range (up to >120 dB). They do not use companders or compress the audio signal in any way, and audio quality does not degrade with distance.

XD-V systems operate in the 2.4GHz band, which is free from interference due to TV broadcast, public safety announcements, cell phone towers and other transmitting devices. Encoded DC (Digital Channel Lock) technology prevents reception of any audio interference from other 2.4 GHz devices.

XD-V handheld systems feature a selection of up to 10 models of popular vocal microphones.  Using this incredibly diverse sonic palette, vocalists can choose the perfect microphone sound to match their voice and style of performance.

For active spoken-word performers, instrumentalists or singers who require a hands-free solution, XD-V bodypack systems offer selectable EQ filter models, tailored for a wide range of vocal and instrumental applications. 

XD-V55 bodypack systems have three selectable vocal EQ filter models.

XD-V series digital wireless systems are incredibly easy to operate. Simply choose a channel on the transmitter and receiver and they lock together automatically. There is no need for RF tuning or intermodulation calculators.

XD-V55 family features: 12 channels, 300-foot range; 1/2U desktop receiver with externally mounted antennas; heavy-duty metal chassis.

Line 6