Article

Tuesday, November 22, 2011

What’s The “Right” Wattage For Loudspeakers?

The REAL question is not what is the power handling, but what is the OPTIMUM size power amplifier to use on a loudspeaker?

“So how many watts can this loudspeaker take?” The technical answer is that it depends on the thermal and mechanical limits of the drivers and crossover components.

The practical answer is that it depends on the program material played: its peak/average ratio or transient content and spectral (frequency) content.

The REAL question is not what is the power handling, but what is the OPTIMUM size power amplifier to use on a loudspeaker?

Rule of Thumb
For a rule of thumb the best answer is found using what is commonly called the loudspeaker’s RMS (root mean square) power rating. Use an amplifier that is twice the RMS rating. If you can’t find an amplifier with that exact rating, multiply this power rating by 0.8 and also by 1.25 to find a range of acceptable power.

Example: A loudspeaker has a 250-watt RMS rating. Twice this is 500 watts. Therefore the acceptable range for the power amplifier to use is from 400 watts (0.8 x 500) to 625 watts (1.25 x 500). Anything larger is potentially excessive power. Anything smaller can cause damage from the amplifier clipping. Remember that the power amplifier output you select must be rated for the nominal impedance of the loudspeaker (i.e., 16, 8, 4 or 2 Ohms).

The RMS rating represents the thermal power limit for the loudspeaker. It is also a good number to use for comparing products. Twice the RMS rating represents a realistic scenario for most actual audio signals. This is 3 dB more power than the RMS rating. An amplifier’s RMS rating is based on a sine wave measurement. The peak power in a sine wave is 3 dB more than the RMS power. Therefore using twice the RMS power rating for the loudspeaker provides 6 dB more power for short term power peaks.

A good loudspeaker can easily handle this. Why? The RMS rating for professional loudspeakers is almost always measured using pink noise for a test signal. The content of this test signal is an RMS level with peaks that are 6 dB above the RMS level. Thus testing a loudspeaker with pink noise requires an amplifier that can produce power peaks of 6 dB above RMS level of the input signal to the loudspeaker without clipping. The rule of thumb provides an amplifier with this capability.

Real World Audio Signals
Real audio signals usually have peaks at least 10 dB peaks their RMS level. Therefore, with a properly sized amplifier just below clipping on those peaks, the RMS value of typical audio signals will be at least 3 dB below or half of the loudspeaker’s RMS rating. This provides a margin of safety.

Still Possible to Damage a Loudspeaker
Having said this, it is entirely possible damage a loudspeaker with an amplifier that is in the “rule of thumb” power range. Why? Because power handling depends on the type of input signal and the user - not the manufacturer - controls the input signal in actual use.

For example, the RMS and peak levels can be about equal on compressed audio signals and for certain signals from instruments like synthesizers or highly processed electric guitars. This means any amplifier power capability above the loudspeaker’s RMS rating can damage it!

Also, no matter what size the amplifier is, clipped signals are death to loudspeakers, even if the clipping occurs in the mixer, equalizer or other signal processor.

There is not a perfect answer to the power handling question. This rule of thumb is a realistic guide for the optimum size power amplifier to use on a loudspeaker for MOST live audio signals. It allows the loudspeaker to be used to its maximum specified power rating.

Caveat About Distortion
Almost no loudspeakers are rated for their distortion at maximum power. For this reason there is nothing implied by the manufacturer in the maximum power rating that says a loudspeaker will still sound good at its maximum power rating. If you find that a loudspeaker “sounds bad” when run near its maximum rating, then the maximum distortion that you find tolerable will be the limiting factor rather than the maximum power rating.

Note: This applies to professional loudspeakers from reputable manufacturers, and includes drivers as well as complete loudspeaker systems.

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Friday, November 18, 2011

Knowing Cone Drivers: How They Work, Understanding Key Data & Specs

What's really going on with woofers, and what are the important factors in how they perform as well as how they impact the performance of loudspeaker systems

(Editor’s Note: Eminence Speaker LLC contributed to this report.)

Cone drivers (also referred to as woofers and transducers in this article) are not overly complex. When an electrical current passes through a wire coil (the voice coil) in a magnetic field, it produces a force that varies with the current applied.

The cone, connected to the voice coil, moves in and out, creating waves of high and low air pressure. The coil and magnet assembly are the “motor structure” of the loudspeaker.

The movement is controlled by the loudspeaker’s suspension, which comprises the cone surround and the “spider”.

The surround and spider allow the coil to move freely along the axis of the magnet’s core (or “pole”) without touching the sides of the magnetic gap.

More important than knowing the details of how cone drivers work is the understanding of key data and what it means. Prior to 1970, there were no easy or affordable methods accepted as standard in the industry for obtaining comparative data about loudspeaker performance.

Recognized laboratory tests were expensive and unrealistic for the thousands of individuals needing performance information.

Standard measurement criteria were required to enable manufacturers to publish consistent data for customers to make comparisons between various loudspeakers.

Things began changing in the early 1970s, however, when several technical papers were presented to the Audio Engineering Society (AES) that resulted in the development of what we know today as Thiele-Small Parameters.

The authors of the papers – A.N. Thiele and Richard H. Small – devoted considerable effort to showing how the following parameters define the relationship between a cone driver and a particular enclosure.

These parameters can be invaluable in making choices because they can tell you far more about the transducer’s real performance than the basic benchmarks of size, maximum power rating or average sensitivity.

Let’s have a look at the parameters defined by Mr. Small and Mr. Thiele. (And note that we listed Mr. Small first this time – bet he doesn’t get that very often!)

Fs: The free-air resonant frequency of a cone driver. Simply stated, it’s the point at which the weight of the moving parts of the speaker becomes balanced with the force of the driver suspension when in motion.

If you’ve ever seen a piece of string start humming uncontrollably in the wind, you have seen the effect of reaching a resonant frequency. It’s important to know this information so that you can prevent your enclosure from ‘ringing’.

With a cone driver, the mass of the moving parts, and the stiffness of the suspension (surround and spider), are the key elements that affect the resonant frequency.

As a general rule of thumb, a lower Fs indicates a woofer that would be better for low-frequency reproduction than a woofer with a higher Fs. This is not always the case though, because other parameters affect the ultimate performance as well.

Re: DC resistance of the driver measured with an ohm meter, and often referred to as the “DCR.” This measurement will almost always be less than the driver’s nominal impedance.

Some users sometimes get concerned the Re is less than the published impedance and fear that amplifiers will be overloaded. Due to the fact that the inductance of a speaker rises with a rise in frequency, it is unlikely that the amplifier will often see the DC resistance as its load.

Le: Voice coil inductance measured in millihenries (mH). The industry standard is to measure inductance at 1 kHz. As frequencies get higher, there will be a rise in impedance above Re, because the voice coil is acting as an inductor.

Consequently, the impedance of a cone driver is not a fixed resistance, but can be represented as a curve that changes as the input frequency changes. Maximum impedance (Zmax) occurs at Fs.

Q Parameters: Qms, Qes, and Qts are measurements related to the control of a transducer’s suspension when it reaches the resonant frequency (Fs). The suspension must prevent any lateral motion that might allow the voice coil and pole to touch (this would destroy the driver). The suspension must also act like a shock absorber.

Qms is a measurement of the control coming from the driver’s mechanical suspension system (the surround and spider). View these components like springs.

Qes is a measurement of the control coming from the driver’s electrical suspension system (the voice coil and magnet). Opposing forces from the mechanical and electrical suspensions act to absorb shock.

Qts is called the “Total Q” of the driver and is derived from an equation where Qes is multiplied by Qms and the result is divided by the sum of the same.

As a general guideline, Qts of 0.4 or below indicates a transducer well suited to a vented enclosure. Qts between 0.4 and 0.7 indicates suitability for a sealed enclosure, and Qts of 0.7 or above indicates suitability for free-air or infinite baffle applications.

Vas/Cms: Vas represents the volume of air that when compressed to one cubic meter exerts the same force as the compliance (Cms) of the suspension in a particular speaker.

Vas is one of the trickiest parameters to measure because air pressure changes relative to humidity and temperature – a precisely controlled lab environment is essential.

Cms is measured in meters per Newton, and is the force exerted by the mechanical suspension of the speaker. It is simply a measurement of its stiffness.

Considering stiffness (Cms), in conjunction with the Q parameters, gives rise to the kind of subjective decisions made by car manufacturers when tuning cars between comfort to carry a family and precision to go racing.

Think of the peaks and valleys of audio signals like a road surface, then consider that the ideal driver suspension is like car suspension that can traverse the rockiest terrain with race-car precision and sensitivity at the speed of a jet plane.

Vd: Peak Diaphragm Displacement Volume – in other words, the volume of air the cone will move. It is calculated by multiplying Xmax (voice coil overhang of the driver) by Sd (Surface area of the cone). Vd is noted in cc, and the highest Vd figure is desirable for a sub-bass transducer.

BL: Expressed in Tesla meters, this is a measurement of the motor strength of a driver. Think of this in terms of how good a “weightlifter” the transducer can be. A measured mass is applied to the cone, forcing it back, while the current required for the motor to force the mass back is measured.

The formula is mass in grams divided by the current in amperes. A high BL figure indicates a very strong transducer that moves the cone with authority.

Mms: The combination of the weight of the cone assembly plus the “driver radiation mass load.” The weight of the cone assembly is easy: it’s just the sum of the weight of the cone assembly components.

The driver radiation mass load is the confusing part. In simple terminology, it is the weight of the air (the amount calculated in Vd) that the cone will have to push.

Rms: Represents the mechanical resistance of a driver’s suspension losses. It is a measurement of the absorption qualities of the driver suspension and is stated in N*sec/m.

EBP: Calculated by dividing Fs by Qes. The EBP figure is used in many enclosure design formulas to determine if a driver is more suitable for a closed or vented design.

An EBP close to 100 usually indicates a driver that is best suited for a vented enclosure. On the contrary, an EBP closer to 50 usually indicates a speaker best suited for a closed box design.

This is merely a starting point. Many well-designed loudspeaker systems have violated this rule of thumb! Qts should also be considered.

Xmax/Xmech: Short for “maximum linear excursion.” Driver output becomes non-linear when the voice coil begins to leave the magnetic gap.

Although suspensions can create non-linearity in output, the point at which the number of turns in the gap (see BL) begins to decrease is when distortion starts to increase.

Xmax is voice coil height minus top plate thickness, divided by two, while Xmech (as expressed by Eminence) is the lowest of four potential failure condition measurements times two: Spider crashing on top plate, and/or voice coil bottoming on back plate. Voice coil coming out of gap above core; physical limitation of cone.

Take the lowest of these measurements and then multiply it by two. This gives a distance that describes the maximum mechanical movement of the cone. (For Eminence transducers, half the Xmech value represents the one-way excursion limit that if exceeded would cause permanent damage.)

Sd: This is the actual surface area of the cone, normally given in square centimeters.

Zmax: Represents the driver’s impedance at resonance.

Usable frequency range: Manufacturers use different techniques for determining this, and most are recognized as acceptable in the industry. However, they can arrive at
different results.

Technically, many drivers are used to produce frequencies in ranges where they would theoretically be of little use. As frequencies increase, the off-axis coverage of a transducer decreases relative to its diameter.

At a certain point, the coverage becomes ‘beamy’ or narrow like the beam of a flashlight.

See the chart at left – it demonstrates at what frequency this phenomenon occurs relative to the size of the transducer. If you’ve ever stood in front of a guitar amplifier or loudspeaker cabinet, then moved slightly to one side or the other and noticed a different sound, you have experienced this phenomenon.

Clearly, most two-way loudspeaker systems ignore the theory and still perform quite well.

Power handling: A transducer needs to be capable of handling the input power it’s provided. The general rule of thumb is that a power amplifier, when reproducing any program source, “provides” long term- thermal power that is approximately 1/8 its maximum rated output before clipping (rap music excluded).

This is why even UL testing for power amps is done, and listed for on the back of the amp, at 1/8 the rated output power of the amp.

Typically, a loudspeaker will handle somewhere between 6 dB to 10 dB higher peaks than its long-term-average power rating, particularly in the case of the conservative EIA-426A standard used by several manufacturers.

This means that if a loudspeaker is rated for 100 watts long-term-average power, the amp driving it should be rated between 400 and 1000 watts – if the user does not compress the source signal. Once compression is used, all bets are off.

Generally speaking, the number one contributor to a transducer’s power rating is its ability to release thermal energy. This is affected by several design choices, but most notably voice coil size, magnet size, venting, and the adhesives used in voice coil construction.

Larger coil and magnet sizes provide more area for heat to dissipate, while venting allows thermal energy to escape and cooler air to enter the motor structure. Equally important is the ability of the voice coil to handle thermal energy.

Mechanical factors must also be considered when determining power handling. A transducer might be able to handle 1000 watts from a thermal perspective, but would fail long before that level was reached from a mechanical issue such as the coil hitting the back plate, the coil coming out of the gap, the cone buckling from too much outward movement, or the
spider bottoming on the top plate.

The most common cause of such a failure would be asking the speaker to produce more low frequencies than it could mechanically produce at the rated power. Be sure to consider the suggested usable frequency range and the Xmech parameter in conjunction with the power rating to avoid such failures.

Sensitivity: One of the most useful specifications published for any transducer, it’s a representation of the efficiency and volume you can expect from a device relative to the input power.

Manufacturers follow different rules when obtaining this information – there is not an exact standard accepted by the industry. As a result, it is often the case that loudspeaker users are unable to accurately compare the sensitivities of different products.

Eminence Speaker LLC and Live Sound/ProSoundWeb Senior Technical Editor John Murray contributed this article.

Also be sure to read Real World Gear: The Latest In Loudspeaker Drivers and take our Photo Gallery Tour of the latest driver models.

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Thursday, November 17, 2011

An Endurance Test Of 9-Volt And AA Batteries

Higher voltage? Longer life? Bringing some facts to the process

9V Transmitter
For the 9V transmitter I used the Shure SC1-TA. It has a relatively low current drain (50mA) that remains relatively constant with increasing time until the battery voltage is insufficient (6.5V) to keep the unit on. This is a VHF system.

AA Transmitter
For the AA testing I used the Lectrosonics SM. It is an ultra-miniature digital hybrid UHF transmitter than runs from a single AA cell. The current draw is voltage-dependent. The transmitter turns off when the battery voltage drops to 0.9VDC.

The Test Setup
To test the batteries I drove the wireless transmitters with a balanced aux level signal from an iPod interfaced to each belt pack via a passive summing/balancing interface (Figure 1). Each was driven with program to produce a full-scale receive signal level on its receiver for the duration of the test.

The turn-off voltage and current draw for each transmitter was determined with a variable DC supply (with on-board ammeter) and an oscilloscope. The battery voltage was monitored with a programmable USB data logger, chosen for its very high impedance and minimal loading of the battery.

Voltage samples were gathered at 1-minute intervals and the test duration extended well beyond the transmitter turn-off time. The results were plotted using a graphing program.

Figure 2 shows the results of the tests. I included a plot of cost vs. time, as well as an estimate of how many charges would be required for the rechargeable batteries to become cheaper than the disposables (not counting the cost of the charger).

The 500mA Lithium Polymer 9V battery (www.ipower.com) is a relatively new development and was suggested by Gordon Moore of Lectrosonics (the 400mAh unit from the same vendor should be avoided for high current drain applications).

Specifications
The two major specifications given for batteries are the voltage (in volts DC) and the available current (in ampere hours).

If a battery can provide one ampere (1 A) of current (flow) for one hour, it has a capacity of 1 A·h. If it can provide 1 A for 100 hours, its capacity is 100 A·h. Most small batteries are rated in mAh.

As with audio gear, “one number” specs can be deceiving.

For example, the NiMH 9V had the highest voltage when fully charged (about 9.5VDC). Yet, it had the shortest life. The Li-Polymer 9V had the lowest full-charge voltage (about 8.2VDC) but lasted the longest.

So, higher voltage is not necessarily better.

The mAh rating of a battery should only be used as a rough estimate of its suitability for an application. For example, the Li-Polymer 9V has a mAh rating that is about three times the NiMH 9V, yet it lasted about six times longer in the endurance test.

Also the disposable AA batteries have higher mAh ratings than the rechargeables, but did not last nearly as long.

Test 123DC
So what can you do with a box of exhausted batteries?

You can test the battery tester! In the run down tests I often let the test rig run overnight to exhaust the battery-under-test. The data downloaded from the data logger showed what the voltage did as a function of time. Most batteries returned to near their full charge voltage once the transmitter shut off. Figure 3 shows the time record for one of the batteries.

If the exhausted battery were tested with a simple DC voltmeter, one could conclude that it was fresh. This is because a voltmeter has an extremely high input impedance to minimize loading on the circuit being tested. A good battery tester will load the battery prior to reading the voltage.

Larry Pajakowski turned the SynAudCon listserv on to an excellent battery tester, the ZTS Pulse Load battery tester (www.ztsinc.com, shown in Figure 4).

After a fully automatic test cycle, percentage of remaining battery capacity is indicated on the LED bar display.

There are several models available and all are relatively low cost (less than $50 USD). Unlike my DC voltmeter, the ZTS correctly indicated the depleted state of each battery after the run down test.

Pat and Brenda Brown own and operate SynAudCon, conducting training seminars around the world in addition to providing in-depth web-based training.

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Friday, November 04, 2011

Elusive Quests: What’s All The Buzz About?

If all else fails, don a bee bonnet, and convince your audience that the sound they’re hearing is a recent insect infestation

If you’ve been in the live sound business for any time at all, or working in any field related to sound, you’re quite likely to agree that the most maddening technical anomaly of all is the elusive “hum” or “buzz.”

These two terms can be related or separate, but they’re both nasty business just the same.

We’ve all set systems - large and small - only to power up and discover the beast within. Is it 60 Hz? 120? 240? A combination? Is it mains? Monitors? Backline? Lights? 

First, do no harm! (Yep, this applies to us too.) Don’t start undoing your stage work in a panic. Try to discern what frequency(ies) are the problem. A Real-Time Analyzer can come in quite handy - even a rudimentary online RTA app is worth the investment.

Let’s focus on the common 60 Hz hum. Hopefully you’ve taken the time to verify the integrity of the house power. This is a must. A simple line checker from the hardware store is a cheap way to make sure the house is “keeping it clean” and feeding the proper line voltage – and with the hot, cold and ground in the right order.

Make sure (especially in small/medium house applications) to connect all PA power feeds, as well as backline, to the same source. Don’t mix and match! It’s a sure-fire ground loop headache waiting to happen.

Invest (wisely) in proper power conditioning and sequencing devices. These protect and catch big voltage issues long before they reach your gear. Also be sure to reduce any capacitive coupling by keeping parallel runs of audio and AC power separated as far as possible from one another. Do a little homework on parasitic coupling. It’s a fascinating subject and very important to us sound people who are prone to running lots of wire.

I know, I know, some of you are saying “go digital and clear up a lot of these issues.” True, some of this applies more to us old (and young) “analog dogs,” but digital offers its own sets of problems. I also don’t think I’m speaking out of turn in either case when I say that you’re only as good as what you’re being fed from the house.

Maybe you’ll get lucky and be able to isolate the problem to one or two input channels. And don’t forget the backline. Sometimes a bad filter capacitor in a backline amp can wreak havoc on your system and the stage audio as well. Use those DI ground lifts as needed. Keep XLR cables in mint condition. A few hours of work in the shop can save lots of headaches in the field.

There’s nothing wrong with older power amplifiers, but you might want to have a look inside. Change those old filter caps, or if you’re not budget-challenged, get some new amps. (They’re lighter weight too!)

Watch out for older small-venue fluorescent lighting or newer compact fluorescents. They and their respective ballasts have a way of getting into your wired (and wireless) systems. Most of the time the problem is a poorly grounded “main,” but nothing substitutes for a clean, stable and well-grounded source. (Not always possible, I well know.)

If all else fails, don a bee bonnet. (I always keep one handy in the gear box.) Convince your audience that the sound they’re hearing is a recent insect infestation, smile, and (try) to have a great show anyway!

Greg Stone has worked in live sound since 1976 and is the owner of Hill Country Ears Sound Company in South Texas.

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Powersoft K10 Amplifiers Drive Low End For Opulent Temple At Burning Man Festival

"The subs provide more than enough low end and the amps offer an incredible amount of headroom." - Parker Walters, Showtech Productions

For the last six years Keith Kettrey and Parker Walters, owners of Showtech Productions based in Dallas, have provided the audio and video system utilized by the Opulent Temple camp at the Burning Man Festival, and this year they supplemented the low end of the left-right JBL Vertec line array system with 12 EAW SB2001 subwoofers (six per side), powered by six Powersoft K10 amplifiers equipped with DSP+KAESOP plug-in boards and Armonía Pro Audio Suite software.

The Opulent Temple, one of hundreds of camps located in Black Rock City, Burning Man’s temporary city located in an ancient lake bed in the Black Rock Desert in Nevada, features live DJ music and video from dusk to dawn along with ample room for dancing. 

“The Opulent Temple space was roughly defined by the DJ booth, or pod, flanked by 12-foot video screens and then 18-foot towers for the sound system. The outer edges of the area were loosely defined by some art structures,” explains Kettrey. “Overall our goal was to cover an area that was roughly 100 feet wide by 150 feet deep. Because we were in the middle of the dessert we wanted to make sure that we brought enough system for the space.”

The EAW SB2001 subs feature dual 21-inch woofers capable of up 3600 watts at 2 ohms which provide substantial low end for the electronic/DJ/Dance music.

The K10 provides 12,000 watts from a single rack space chassis. Operation with 2-ohm loads, like those presented by the SB2001s, is not only safely possible but recommended, substantially reducing the number of amplifiers the system required. Each K10 easily powered two SB2001 subs.

“Because of the music style we wanted enough power to ensure a driving low end,” Kettrey continues. “We discovered the Powersoft amps when we attended an EAW event where they used them to power the SB2000 subs – and we were amazed.”

Walter adds, “The subs provide more than enough low end and the amps offer an incredible amount of headroom. Although we were running the amps at 208 volts, the average current draw preamp was less than 10 amps per K10 during the peak performance time slots. Simply amazing.”

“By using the K10s with DSP+KAESOP and Armonía software we were able to walk out on the dance floor and make any modifications needed,” explains Kyle Kettrey, Keith’s twin brother and team advance man. “I even set up a network bridge so that we could use the tablet PC from our campsite – if some DJ got out of hand with his volume levels we could handle it.”

Because the Black Rock Dessert features sand that closely resembles sheetrock powder, occasional wind storms and temperatures that can reach 120-plus degrees during the day, Kettrey knew that the system had to be protected from the elements if it was to survive the week – so all of the Powersoft K10s along with system processing were enclosed in an sealed tent with self-contained air conditioning to keep the space cool and the air filtered.

“The Powersoft amps run pretty cool, so that wasn’t a serious concern,” Kettrey explains. “It wasn’t the heat as much as keeping the equipment clean so that they run well – the tent was the easiest solution. The rest of the gear just got really, really dirty. When we got back we literally dumped it all in the parking lot and pressure-washed everything. The amps stayed spotless and worked extremely well.”

Powersoft

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Wednesday, November 02, 2011

Clear Path: The “Right” DI For Computers In Audio

When it comes to best audio quality practices, they’re sometimes not ideal.

Ready, brace yourself, this one is going to hurt: Computers are not made for audio. There, I finally said it!

Computers are made for crunching numbers, and they also happen to be able to manage audio and video tasks really well because of their tremendous processing power.

However, when it comes to best audio quality practices, they’re sometimes not ideal. Often we use computers to feed tracks to PA systems and as playback machines for things such as backing tracks. Getting the sound from the computer into a sound system is relatively easy: Connect the 3.5-mm (1/8-inch) unbalanced output jack and away you go. If only it were that simple…

Anyone who has done this knows that more often than not, it can introduce a ground loop or induce noise via the unbalanced line. Even PA system noise can find its way into the computer, adding noise to the program material output. Amplify any of this with 20,000 watts and you have a problem. 

Passive Boxes
Several companies produce direct boxes that are specifically designed for computers. These are usually stereo, and more often than not, are passive or transformer based.

In other words, the transformer not only converts the unbalanced signal into a balanced one, but also introduces galvanic isolation to eliminate stray DC currents from traveling in between the computer and the audio system. And when the ground is lifted, all of the audio passes through the transformer disconnecting the ground thus eliminating the ground loop. 

Because the computer’s output is buffered (usually by a -10 dB consumer level or headphone jack), a passive DI is perfectly suitable for computers. Transformers can usually handle a lot more signal before distortion when compared to phantom powered active DI boxes. This makes them a better choice when using the headphone jack.

Active Boxes
The active direct box was originally developed as a means to eliminate loading that would occur on low output electric bass pickups. By introducing a buffer, the bass signal going to the artist’s stage amp would not be affected thus conserving his sound while the PA system would be fed a hotter signal.

Buffers are essentially amplifiers. This means that they need power (voltage and current) to make them work. The preferred power source is 48-volt phantom because it does not require running separate AC for the DI box.

The other hidden advantage of a buffer is that the signal will only go one way. Unlike a transformer that is bi-directional, buffers do not allow signals to go backwards. Where this matters in our world is preventing noise from polluting the computer. 

And because most program material is limited during the mastering process, one can get sufficient headroom using phantom power to generate a relatively clean signal. The problem, unless dealt with, is the lack of galvanic isolation; active DIs don’t solve ground loop problems.

There are some DI boxes that combine the benefit of an active direct box with transformer isolation. These are usually a little more expensive than a simple passive or active DI because they offer the best of both worlds. The transformers isolate the computer from the PA, while the buffers inhibit PA noise from polluting the computer.

Peter Janis is the president of Radial Engineering and has worked in professional audio for more than 30 years.

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Friday, October 28, 2011

Willow Creek Church Improves System Efficiency With Powersoft Amplifiers

Subwoofer power consumption cut by two-thirds

Willow Creek Church in South Barrington, IL, recently installed two Powersoft K3 amplifiers to power the six Danley Sound Labs TH-115 tapped horn subwoofers that provide dynamic low-frequency support to the sanctuary sound system.

Seven years ago, Willow Creek Church, one of the largest churches in the U.S. in terms of both membership and facilities, commissioned a new sound reinforcement system to serve its main sanctuary, which has a capacity of more than 7,200.

Until recently, the subwoofers were driven by power amplifiers retained from a previous system. Time took its toll, however, with the amplifiers increasingly plagued by systematic failure, to the point where Matt Satorius, audio system engineer at Willow Creek, began researching options for an upgrade.

“Specifically, we were looking for amplifiers offering increased power efficiency, digital signal processing, and the ability to be controlled and managed over a network,” Satorius explains. The thorough evaluation process eventually led him and the church production team to the Powersoft K Series, which provide exceptional power and efficiency from a very compact footprint.

“The efficiency of the Powersoft amplifiers is the ticket,” Satorius states. “We were able to replace the existing six amplifiers with just two Powersoft K3 amplifiers – they’re amazing. The original setup required three 20-amp AC circuits per side, but we’ve been able to cut that by two-thirds with these new amplifiers. That’s efficient.”

A single K3 amplifier per side, specified as being capable of generating more than 5,000 watts is all that is required to drive its respective set of three subwoofers.

In addition, both K3 amplifiers are outfitted with the optional KDSP board that provides a suite of digital signal processing, further enhanced with the inclusion of the optional KAESOP Ethernet/AES3 interface. They team up to offer sound shaping and system management capabilities, remotely controllable via Powersoft Armonía Pro Audio Suite software.

A proprietary algorithm onboard each unit fosters a tight and controlled low end, which is a must for optimized subwoofer performance.

“The amps are rack mounted under the stage with the subwoofers,” concludes Satorius. “Being able to remotely monitor and control them is extremely nice. As an added bonus the install and setup was a breeze – we had them swapped out and working in an afternoon.”

Powersoft

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Tuesday, October 25, 2011

In Profile: Technologist & Powersoft Co-Founder Claudio Lastrucci

“If you desire to do a lot of things you can find time to do everything.” - Claudio Lastrucci

Claudio Lastrucci has always felt driven to explore undiscovered territory.

It’s a quality that has informed all of his innovations as head of research and development and managing director of Italy-based Powersoft, and perhaps nowhere more than in the creation of DIGAM, the company’s signature.

Short for Digital Audio Amplifiers, DIGAM is a patented amplifier technology employing Power Factor Correction (PFC) that has enabled Powersoft to create products so compact, powerful and efficient that some in the industry once believed their claims were simply too good to be true.

Lastrucci understands the skepticism: “Changes in technology require time to show that the principles behind their development are correct. It’s not something that happens in months. It takes years.”

Indeed, in the June 2003 issue of Live Sound International, amplifier designer Jeff Kuells presented a thorough evaluation of the Powersoft model Q4002, leading his report with the statement, “How small can power amplifiers get before they become ergonomically difficult to use? I believe Powersoft might have found the limit.”

The Q4002 is only 1RU, weighs just 21 pounds, but it packs four independent channels that deliver 950 watts per channel (RMS). It was mind-boggling at the time, and only a bit less-so now.

Still, Lastrucci and company haven’t found the limit on the size-to-power ratio front, with the more recent M50Q 4-channel model for touring and install applications capable of delivering 1,250 watts per channel (at 4 ohms) from a 1RU package that weighs just a bit more than 16 pounds.

Although PFC was a familiar concept in other industries, it was new to professional audio when it was first introduced by Powersoft 15 or so years ago. Combined with PWM (Pulse Width Modulation) technology, the result was one of the first true high-power, full-bandwidth Class D professional amplifiers, a product roughly a fifth the size and weight of traditional amplifiers while offering a significant increase in efficiency.

“It’s a factor of 10,” Lastrucci notes about the efficiency. “In 1995, our technology was outside the view of anybody in the business, and we were the first company to offer it, but now that it’s commonly known we have something like 250,000 units worldwide.”

PFC also allows the amplifiers to operate at line voltage around the world, from 95 volts to 265 volts. The power supply remains stable under any AC line condition while maintaining a consistent power output from the amplifier. And perhaps most significantly to Lastrucci, it helps produce energy savings of approximately 40 percent.

In addition to DIGAM, which was patented worldwide in 1997, Lastrucci also holds international patents for a variety of power electronic topologies, control methods, acoustic transducers and other technologies, and as a result, he is often referred to as the inspiration behind Powersoft.

In conversation, however, he is quick to point out the contributions of the company’s staff, insisting they all share a common drive to change the rules of the audio industry with the work they pursue. “Most of them are musicians and sound engineers, so music and sound are their life. They are very passionate about what they do. I don’t know, maybe it’s something that is in the blood.”

Not So Different
Given the company’s base in Lastrucci’s native Florence, that could very well be the case.

Widely considered the birthplace of the Renaissance, the city has been home to a number of individuals whose contributions to art, architecture and technology have been dramatic to say the least - perhaps most famously, Leonardo da Vinci.

“That’s my first example,” Lastrucci laughs, noting that da Vinci is a favored point of reference for him, an example of someone whose legacy underpins his own belief that art and technology are not so different from one another as some might think.

Both involve an initial moment of raw inspiration and a similar process of research, and development to achieve a finished product – regardless of whether the result is a work of art, an architectural achievement, or the establishment of a new scientific principle or technology.

“Perhaps art can be more readily understood by some, but those who do not understand there is some art in technology do not understand technology,” he says simply.

For Lastrucci the connection between art and technology has been evident since he was a child, a product of his own long held passion for both music and engineering. “I’ve been a piano player for 30 years, since I was six. I’ve been a drummer for seven and a saxophone player for many years as well, but also I was always very curious about electronics, physics and chemistry.”

While that combination of interests fueled the initial inspiration for DIGAM and ultimately led to the founding of Powersoft, Lastrucci’s family and their willingness to support both his technological and musical aspirations were extremely important as well.

In fact, Powersoft began very much as a family business and remains so to this day, with Lastrucci’s brother Luca and their mutual friend, Antonio Peruch, also serving as managing directors and Lastrucci’s father, Carlo, as president.

Ideas Into Reality
Lastrucci, his brother and Peruch actually formed the company two months before their graduation from the University of Florence, where both Lastrucci and Peruch studied electronic engineering. They started out with limited resources, developing the technology for the products that are now their core business in a small apartment while providing consultancy services for others for the first year and a half of their existence.

“We had no employees, but we had a lot of ideas,” Lastrucci says, and steadily, they worked to make those ideas reality, consistently expanding their offerings and consistently growing 20 to 30 percent annually for 16 years. The second generation of DIGAM was followed by D and Q Series amplifiers in 1999, then the installation-specific QTU1400, and by 2003, mass production of the company’s PowerMod and DigiMod DSP hardware.

Aside from DIGAM, Lastrucci believes that the K Series, a line of amplifiers unveiled in 2004, represented a major technological step forward, and it also opened up the U.S. market.

Even so, he maintains that every new product and innovation represents an important milestone, from the 2008 launch of the IPAL Integrated Powered Adaptive Loudspeaker in cooperation with B&C Speakers and the 2010 release of Armonia remote control and monitoring software, to the most recent offering, the M Series, in 2011.

The apartment workshop gave way to a larger production space in 2002, and to their current complex shortly thereafter.

Now, in addition to the administration and engineering facilities in Florance, Powersoft runs dedicated manufacturing plants in Bologna and Gorizia, employing roughly 100 people in total.

A third building is currently under construction in Florence to serve as a main demo facility and provide warehouse space, and earlier this year, the company set up a dedicated U.S. office and service center in Southern California under the direction of pro audio veteran Ken Blecher.

Future Look
At the time Powersoft came into being, “green” products, generally speaking, were not prevalent. “My perception is that people don’t understand what kind of impact a specific product can have. For instance, some will argue about whether or not an efficient amplifier will have a measurable environmental benefit, but the same person would never think of opening the window during the winter with the heat on.” For Lastrucci, the equation is simple: “If it’s green, it’s better.”

As much as that perspective is informed by his explorations of uncharted territory as a technologist, it is also rooted firmly in a deep-seated fascination with nature. It often leads he and his family to explore decidedly less traveled regions far removed from the cultured environment of Florence - trekking and diving, for example, in Africa, Indonesia and South America. “We go on long, often complex trips,” he says. “And in very harsh conditions, let’s say.”

This passion continues to drive Powersoft’s creation of new products like the Deva 1S, a solar powered, wireless A/V communications device for audio distribution and surveillance in public spaces ranging from theme parks, to transport hubs, to live venues. While not necessarily representative of their core business, finding compact, energy-efficient solutions for a variety of applications is part of the DNA of the company, Lastrucci says.

“If you desire to do a lot of things you can find time to do everything,” Lastrucci concludes, a philosophy informed in part by the fact that he and his wife of 12 years recently welcomed a new addition to their family. “With two boys and a baby girl, there’s a lot to do,” he adds, laughing.

Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.

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Monday, October 24, 2011

Gepco Launches RunONE Powered Loudspeaker Cables

Combine audio, power and optional data under one jacket

Gepco International has introduced RunONE powered loudspeaker cables, which combine audio and power, along with optional data, under one durable yet flexible jacket.

Each RunONE cable combines one channel of power with two, eight or 12 channels of 110-Ohm balanced audio for line level, mic level or digital AES audio signals and can be used with self-powered loudspeakers or in DMX lighting control.

Additional configurations include two channels of Category 5e cable that can be used for data drops in remote power and audio applications.

Snakes with optional data can also be used for digital audio transmission while running power to Front of House for remote locations. 

Shielding around the power channels eliminates power noise from interrupting the audio/data signal, ensuring high-quality performance.

Terminated with industry-standard connectors, RunONE cables offer the option of Edison, IEC and Neutrik powerCON connectors for the power channel; 3-pin XLR, 5-pin XLR (for DMX lighting), TRS and Neutrik convertCON connectors for audio channels; and RJ45 and Neutrik etherCON connectors for optional data channels.

RunONE cables are available in pre-defined and custom configurations.

“The RunONE cables are a great solution for anyone looking to save time,” says Joe Zajac, market development manager for Gepco Brand products. “With up to 12 channels of audio combined with power and the optional two channels of data, the RunONE cables will also provide for much cleaner set-ups.”

Gepco International

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Wednesday, October 19, 2011

Pesky Ground Loop Problems Plaguing You?

Although ground loops often involve power line safety ground connections, disabling them is both highly dangerous and illegal...

When a system contains two or more pieces of equipment that are grounded, whether via power cords or other ground connections, a “ground loop” will likely be formed. See Figure 1, below.

Although ground loops often involve power line safety ground connections, disabling them is both highly dangerous and illegal.

However, devices called “ground isolators” can be inserted in the signal path to break the loop safely.

This approach attacks the problem at its fundamental roots, while tampering with safety ground does not. In simple language, a ground isolator is a device that transfers a signal across an electrically insulated barrier.

This is how it stops the flow of power-line currents that would otherwise generate noise as they flow through signal cables.

Because an isolator is not a filter that recognizes and removes noise, it must be inserted in the signal path at the point where the noise coupling actually occurs.

On the other hand, a transformer can serve as an extremely effective ground isolator.

As shown in Figure 2, it transfers signal voltage from one winding to the other without an electrical connection between them.

This electrical isolation blocks the flow of ground noise current in the signal cable.

While the isolation would be total for an ideal transformer, physics imposes limitations on real-world transformers.

Two Basic Types
In practice, noise reduction depends critically on the design of the transformer. Audio transformers fall into two basic types.

The first, known as an output transformer, is by far the cheapest and easiest to build. Because its primary and secondary windings are physically interleaved, considerable capacitance is created which allows noise currents, especially at higher audio frequencies, to flow between windings.

This limits its ability to stop ground noise.

The second type, known as an input transformer, is built with internal metal foil shielding between its windings.

This “Faraday shield” effectively eliminates capacitive coupling and vastly improves noise rejection.

A magnetic shield serves a completely different purpose and, if used, is on the outside of a transformer surrounding both the core and the windings.

Figure 3 shows noise rejection versus frequency for a typical unbalanced interface.

With no isolator, by definition, there is 0 dB of rejection in the interface, as shown in the upper plot.

The middle plot shows results for a typical isolator using an output transformer. Hum at 60 Hz is cut by 70 dB, but buzz artifacts around 3 kHz are reduced by only 35 dB.

The lower plot shows results for a typical isolator using an input transformer. Hum is cut by over 100 dB and buzz by over 65 dB.

The overwhelming majority of “black boxes” intended to solve ground loop problems use output transformers.

One advantage of these boxes is that they can be installed anywhere along the length of a cable or can be used at patch-bays.

Although boxes made with input have some 30 dB better noise rejection, they must be installed thoughtfully.

In Figure 4, we see a commercial black box. Because high-frequency response can be degraded by excessive cable capacitance at their outputs, these types of boxes must be installed near the equipment input they drive, generally through no more than 3 feet of cable.

Some commercial interface devices are “active” (i.e., powered) devices. Although these often have useful features, they invariably use differential amplifier circuits to “isolate” their unbalanced inputs.

In a future discussion of balanced interfaces, we’ll find that ordinary diff-amps do this job very poorly. Typical products in this vein often deliver only 15 dB to 30 dB of noise rejection under typical real-world conditions.

Incidentally, to eliminate noise in an unbalanced cable run, it’s not necessary to “balance” the line (using a converter at the driving end) and then “unbalance” it (using another converter at the receiving end).

The noise rejection of such a scheme is no better, and often worse, than that of a single high-performance isolator (i.e., input transformer) installed at the receiving end.

Check performance data for isolators carefully. Many have scanty, vague or conspicuously non-existent specifications, and many use cheap, telephone-grade transformers.

These can cause loss of deep bass, bass distortion, and poor transient response. Data for high-quality isolators is complete, unambiguous, and verifiable. Input-transformer-based isolators have other benefits, too, including:

• Their inputs are truly universal, accepting signals from either unbalanced or balanced sources, while maintaining extremely high noise rejection

• They provide inherent suppression of RF and ultrasonic interference. The subsequent reduction of “spectral contamination” is often described as a marvelous new sonic clarity (Reference 2)

• They are passive, requiring no power

• They are inherently robust, reliable, and virtually immune to transient over-voltages.

Explore The Options
In many systems, including the one seen earlier in this article in Figure 1, there is more than one way to break the ground loop.

Observe that the noise voltage between the CATV ground and the ac power safety ground at the subwoofer causes noise current flow in the shield of all the signal cables between the CATV ground and the sub-woofer.

Common-impedance coupling will induce noise in both audio cables in the path, generally in proportion to their lengths.

CATV feeds are notorious for having “ground” at their shield several volts different from utility ac power ground, this system might exhibit a very loud hum regardless of preamp control settings because of coupling in the 20-foot cable.

Of course, the loop could be broken by defeating the subwoofer safety ground - but don’t do it! Remember, audio cables that connect equipment together will also carry lethal voltages throughout the system or could start a fire if the sub-woofer develops a defect.

A safe way to break the ground loop is to install a ground isolator somewhere in the audio signal path from TV to subwoofer.

Because longer cables are more likely to couple more noise, the preferred location in this system would be at the receive end of the 20-foot cable (Figure 5).

Another safe, and potentially less expensive, solution is to break the loop by installing a ground isolator in the CATV feed as shown in Figure 6.

CATV isolators must be installed downstream of the lightning ground and should generally be installed where the cable first connects to the audio or video system, such as at a VCR or TV input.

Additional installments by Bill Whitlock are available here.

 
Bill Whitlock has served as president of Jensen Transformers for more than 20 years and is recognized as one of the foremost technical writers in professional audio.

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Wednesday, October 12, 2011

Rain Computers Debuts Nimbus M2 Workstation Combining Pro Audio & Video Production

All-in-one creative computer designed for everything from Pro Tools to Adobe Premiere

Rain Computers has released a follow up to the Nimbus Multimedia Computer - the new Nimbus M2 is a versatile workstation offering excellent power versus performance ratio by offering a standard 6-core processor, NVIDIA Quadro 600 workstation video card and 8GB or 16GB of memory.

Rain worked with notable software engineers from companies such as Adobe Systems to create a stable and reliable platform. The result is a computer that “just works”, even with the tricky software and hardware found in recording studios and graphics departments.

“The idea was to design the Swiss Army Knife equivalent of a workstation. Something just as good at crunching guitar tracks as it was at figuring out the lighting in a 3D model.” comments Kevin Jacoby, Rain Computers CEO. “But we also wanted it to be really affordable so we could offer it to the next generation of creative geniuses. We definitely got there and that’s the part we’re most excited about.”

Nimbus M2 is built, in part, on the success of the Phenom II X6 6-Core processor developed by AMD, a noted superconductor manufacturer.

Nimbus M2 also sports some significant upgrades since the last version. Of particular note is the ability to expand to 32 gigabytes of memory, the external hard drive bay allowing for a fast “hot swap” drive to be installed without any technical knowledge, and the new cooling system developed by NZXT that combines modular cooling fans with advanced acoustic insulation.

Like all Rain computers, Nimbus M2 is available with RainCare Encompass Creative Computer Support, a support system designed to cover not just the computer itself, but the “grey area” between the computer, software and hardware. A free 30-day trial of RainCare Encompass is always included.

Nimbus M2 Multimedia Workstation is available now starting at $1,599 through the Rain Custom Shop at RainComputers.com as well as through select Rain Authorized Dealers like American Musical Supply (AMS) and B&H Photo in New York.

Rain Computers

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Tuesday, October 04, 2011

Middle Atlantic Introduces Industry’s First Hardwired Rackmount UPS System

Also provides a code compliant solution for extending UPS backed power outside the rack

Middle Atlantic Products announced the addition of a hardwired rackmount UPS to its family of UPS systems that provides a code-compliant, cost-effective solution for seamlessly distributing battery backup power both inside and outside the rack. 

The first of its kind to offer hardwired input and output connections, this new UPS minimizes the chance of inadvertent or malicious disconnection, making it ideal for critical applications subject to high security, those utilizing point-of-service devices like cash registers, or any environment where uptime is critical.

The new hardwired UPS also provides a code compliant solution for extending UPS backed power outside the rack, and allows an installer to securely terminate remote hardwired branch circuits directly to the UPS without requiring a transition box, saving time and money on the purchase and installation of extra materials. 

This feature is ideal when maintaining power to remote projectors, protecting the device and bulb from damage due to power outages.

Hardwired UPS models include those that provide hardwired input and hardwired output connections, as well as models that provide corded input and hardwired output connections.  All new hardwired UPS systems feature additional local output receptacles.

Middle Atlantic supports the UPS line with a 3-year warranty program.

Middle Atlantic

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Tuesday, September 27, 2011

Components & Techniques For Getting The Best Results With Portable PA

So many options, and so many of them great. Here's what to look for and how it can work in meeting your specific application needs

Technically, lot of sound reinforcement systems are “portable.”

Get together enough hands and/or the right heavy equipment, and almost anything is portable.

But in pro audio, the term portable PA refers to compact systems that can be easily transported in a small truck, van, or even the trunk/back seat of a car, and then hand-carried (or hand-trucked or rolled) into a venue and set up quickly by one person (if need be).

Most often, we’re probably first think of loudspeakers - compact, 2-way models that are also sometimes referred to as “speakers on sticks,” but there are other components such as powered and non-powered mixers in the genre, as well as packaged systems that come complete with stands, cables and even microphones. 

Portable PA systems have been around for decades, and while the basic form has remained the same, they’ve come a long way in terms of performance capability and feature sets.

A popular model in the 1950s was the Knight system, a total package that included a 32-watt power amplifier, a choice of microphones, and even the option of a 4-speed record player!

By the 1970s, the genre had really moved forward in terms of quality and output, with many of the larger manufacturers offering portable components and systems.

One that stood out was the TAPCO Entertainer system from Electro-Voice, which earned a well-deserved reputation as an excellent portable PA system, comprised of the 100M powered mixer and 100S loudspeakers.

In fact, Entertainer systems can still be found performing today, and the heritage of the system continues at EV, exemplified by the new ZXA1 loudspeaker introduced earlier this year.

The 2-way ZxA1 is powered by an integrated 800-watt, 2-channel amplifier module, and it also has built-in steep crossover slopes and woofer excursion protection as well as a switchable high-pass filter allows for use with a subwoofer.

Further, it’s outfitted with XLR microphone and line level inputs, along with an XLR line level output for daisy chaining several amplified loudspeakers or a sub.

Staying with modern portable loudspeakers, the JBL EON has undoubtedly become one of the world’s all-time best-selling models over the past decade or so, and these days it seems like every other event has a stage flanked by Mackie SRM loudspeakers on stands.

Other portable components continue to evolve to a higher standard as well. Soundcraft just released the Notepad Series of multipurpose mixers, with all models outfitted with the company’s GB30 mic preamp and four stereo line inputs.

The Notepad 124FX also has an integral digital effects processor, which has a feed from every input and over 100 effects, including a pink noise and test setting.

Meanwhile, QSC Audio GX Series amplifiers have become a staple in driving portable systems, featuring Class H topology that’s based on key elements of the premium PLX Series, while the recently introduced Crown Audio XLS Series amplifiers integrate advanced crossover, limiting and DSP. 

When it comes to packages, there are a plethora of packages.

The Yamaha STAGEPAS Series offers passive loudspeakers, powered mixer and cables. A nifty facet is that the powered mixer is actually housed in one of the loudspeakers, and is detachable.

There are dozens of choices in this genre, ranging from professional caliber down to more “prosumer” models that even incorporate CD and mp3 players and a wireless microphone receiver within the loudspeaker system cabinet.

Sometimes, just a powered loudspeaker about the size of a loaf of bread equipped with a mic input will do the trick, and these can also come in handy for spot monitoring in support of larger systems.

Plenty Of Choices
Whether it’s independent components or a package, every portable PA system usually includes these basic items:
• Microphones (and sometimes direct boxes)
• Mixer and power amplifier, or powered mixer, or mixer and powered loudspeakers
• A pair of loudspeakers
• Loudspeaker stands, microphone stands
• Mic, line-level and loudspeaker cables
• Optional items include floor monitors, direct boxes and a mic snake

Let’s look more closely at each component.

Microphones. Step one is figuring out how many mics (and mic inputs) are needed. Unidirectional dynamic mics work well for speech, singers, guitar amps and drums, while unidirectional condenser stand-mounted mics are a solid choice for acoustic instruments as well as singers. Presenters and singer can also be outfitted with headphone mics, and acoustic instruments with mini clip-ons.

Power Amplifiers. How much power do you need? If you’ve already got loudspeakers or are going to purchase them, follow the recommended power guidelines provided by the manufacturer. Levels of a powerful system can always be tuned down to match the application, so be sure not to go too light in terms of output.

That said, here are some very loose guidelines:
• Speech-only system in medium room: 50 watts continuous per channel
• Folk music in a coffee shop with 50 seats: 25 to 250 watts
• Folk music in a medium-size auditorium, club or house of worship with 150 to 250 seats: 95 to 250 watts
• Folk music at a small outdoor festival (50 feet from loudspeaker to audience): 250 watts
• Pop or jazz music in a medium-size auditorium, club or house of worship with 150 to 250 seats: 250 to 750 watts
• Pop or jazz music in a 2,000-seat concert hall: 400 to 1,200 watts
• Rock music in a medium-size auditorium, club or house of worship with 150 to 250 seats: at least 1,500 watts
• Rock music at a small outdoor festival (50 feet from loudspeaker to audience): At least 1,000 to 3,000 watts

As previously noted, some power amplifiers now also offer built-in DSP, and these packages are increasingly competitive from a price standpoint.

The digital processing can serve as a substitute for active crossovers and delays.

Crossover-filter presets for specific loudspeakers can make it a snap to set up a multi-way system.

High-pass filters prevent harm to loudspeakers, often due to powerful lows in music, mic-stand thumps or accidental DC at the amp output.

Mixers. Self-powered or not? A mixer that incorporates power amplification is usually easier to carry and set up - it’s more plug and play.

An increasing number of powered mixers also include onboard effects, and some offer a graphic equalizer, which is useful for tuning the frequency response of the loudspeakers in different environments.

An advantage of a separate mixer and power amp is that if either one fails, only one component needs to be replaced.

Stand-alone mixers also tend to be more ergonomic for the user, and there are more size options - no need to purchase a unit with 8 channels if the maximum number of channels needed isn’t going to exceed 4 or 5. 

Be sure that the mixer has enough balanced XLR mic inputs to handle any possible application you have in mind for the system. RCA jacks for CD and MP3 players come in handy.

Loudspeakers. Full-range loudspeakers for portable PA are usually 2-way designs, either powered or passive. Most applications require just two loudspeakers, but there may be times when additional loudspeakers are needed.

Flexible connectivity is something to keep in mind.

For example, the latest generation of JBL EON portable loudspeakers have an XLR output where the output signal is selectable, either the whole mix may be looped to another loudspeaker (or sent to a mixing console), or simply the primary input for traditional “daisy-chaining” of additional loudspeakers.

These also offer one XLR/ quarter-inch combo connector and additional quarter-inch inputs providing input flexibility and the ability to mix multiple sources.

Some applications may require a stage monitor or two, in addition to two or more main/full-range loudspeakers. An increasing number of portable cabinets offer a side that angles the drivers more steeply upward toward the performers.

Typically, full-range loudspeakers are available with a choice of 12-inch or 15-inch cone woofer that is ported, joined by a driver on a horn or waveguide.

Horn dispersion (6 dB-down points) is commonly 40 degrees vertical x 90 degrees horizontal, or 40 degrees vertical x 120 degrees horizontal, but again, we see an increasing variety of coverage patterns available.

Keep in mind that the overall goal is to focus as much direct sound on the audience (and off of surrounding hard surfaces). Thus the polar pattern is important, particularly for indoor applications and especially in highly reverberant spaces like gymnasiums and some worship sanctuaries.

The frequency response of the loudspeakers should be wide enough to reproduce the sound source accurately.

For speech only, 100 Hz - 12 kHz is usually sufficient, but a guitar-singer application is better served by 80 Hz - 15 kHz and a rock band really should have 40 Hz - 15 kHz or higher.

Those frequency limits are typically 10 dB down or less from the level at 1 kHz.

Of course, the flatter the response over the passband, the more accurate the reproduction.

Boosting The Boom
This brings us to subwoofers. Dynamic music performances can really benefit from extended low-end energy.

The majority of full-range portable loudspeakers also offer at least one companion subwoofer, typically loaded with a 15-inch or 18-inch woofer.

These usually include mounts to accommodate stand poles, and sometimes have wheels on their back floor edge for easier transport.

An option for music applications is a subwoofer/satellite configuration, where one or two subwoofers on the floor provide deep bass, while two compact satellite loudspeakers on stands provide the rest of the spectrum.

Because our ears don’t localize extreme low frequencies, all sound “appears” to come from the satellites. The advantage of this approach is not having to lift large, heavy loudspeakers for positioning on stands.

Column loudspeakers are having an influence on portable PA designs, and that’s no surprise considering that the Shure Vocalmaster system, one of the most popular portable systems in the 1960s-70s, also featured a column approach.

A recent example is the Fishman SA220, which incorporates six 4-inch mid-woofers in a vertical line topped by a 1-inch soft-dome tweeter, housed in a cabinet measure just over 5 inches wide by 6 inches deep.

It’s self-powered and comes equipped with two mic/instrument channels with high-quality preamps, each with 3-band EQ, phantom power, built-in reverb, effects loop, notch filtering and phase controls.

A popular configuration is to place the SA220 behind the performer/group, where it can serve as both mains and monitors. Another common set is one SA220 per performer.

The advantages are that the performer(s) hear the same mix that the audience hears, and the sound level is more constant with distance than with a woofer/horn system.

Further Considerations
These days, most powered PA loudspeakers are bi-amplified: they have one amplifier for the woofer and another for the tweeter.

Advantages of bi-amplification include:
•  Distortion frequencies caused by clipping the woofer power amplifier will not reach the tweeter, so there is less likelihood of tweeter burnout if the amplifier clips. In addition, clipping distortion in the woofer amplifier is made less audible.
•  Intermodulation distortion is reduced.
•  Peak power output is greater than that of a single amplifier of equivalent power.
•  Direct coupling of amplifiers to speakers improves transient response—especially at low frequencies.
•  Bi-amping reduces the inductive and capacitive loading of the power amplifier.
•  The full power of the tweeter amplifier is available regardless of the power required by the woofer amplifier.

Loudspeaker cabinets come in both wood and molded plastic.

Wood cabinets are covered with either scratch-resistant heavy-duty paint or durable fabric that’s often called, in slang, carpet.

Molded plastic cabinets have also proven durable, can feature attractive styling, and are often lighter than their wood counterparts.

For example, a quality 12-inch, 2-way model with plastic cabinet might weigh about 25 pounds, while it’s wood counterpart could be double that figure. (You can come to appreciate lower weight after several nights of muscling loudspeakers on stands.)

Some plastic cabinets, however, have a tendency to “leak” low frequencies or resonate, possibly degrading both frequency and time response.

Most cabinets include a pole cup on the bottom that accepts a 1-3/8-inch or 1-1/2-inch pole for mounting.

The pole cups are vertically aligned, with some having a second angled cup to aim the loudspeaker down toward the audience when raised.

Two models in the new QSC Audio KW Series offer a proprietary Tilt-Direct pole cup mounting system.

A turn of a dial engages a 7.5 degree downward tilt of the loudspeaker, directing more of the acoustic energy toward the audience.

Moving & Mounting
Manufacturers of portable loudspeakers typically provide stands, either included or available as an option, and of course, these work well for the vast majority of applications.

Be vigilant about stands - they should be constructed of metal (usually aluminum), offer a solid tripod base and a secure collar, and be rated to comfortably handle more than the weight of the loudspeaker.

There have been some interesting developments in the world of stands as well.

The Ultimate Support Systems Air-Powered Series have an internal shock that lifts and lowers loudspeakers weighing 50 pounds and less with virtually no effort.

The Ultimate TeleLock Series has a collar that gives the user the ability to safely raise or lower the stand while a loudspeaker is on it. 

Both Air-Powered and TeleLock poles are also available for subwoofer mounting.

Portable PA loudspeakers can also be a good choice for permanent and semi-permanent installations.

For example, a church may find that its portable loudspeakers on stands are doing a fine job, so well in fact, that there’s the desire to get them out of the way and permanently mount them.

If you suspect this might be a future possibility, look for loudspeakers outfitted with mounting points that bolt onto steel cables or yokes that can be aimed as needed.

Don’t skimp on loudspeaker cables. Relatively low gauge cable is good practice, with #14 or #12 zip cord or zip cord, PVC or SJO cable of the same gauge seen most often.

Neutrik Speakon connectors are both more reliable than phone plugs (1/4-inch TRS), they lock into place, and pass more current.

To avoid tripping that could result in injury to a person or a loudspeaker or both, tape cables securely to the floor, using gaffe tape.

Extra cable length can be coiled and stored under the mixer or under the loudspeaker stand.

A very helpful item is a dolly, wheeled cart or hand-truck to transport equipment into venues. Consider lightweight tubular carts such as those from Ultimate Support Rock n Roller and Kart-A-Bag - being collapsible, they store easily in your car or truck.

Any additional racks and trunks should have casters for rolling, and lockable casters will keep them in place once positioned. Some trunks have trays and lid storage to make it easier to organize items.

Protective covers (usually available from the manufacturer or from Under Cover or Cloud 9) should be used to prevent damage of components, including for the loudspeakers. No one likes to see ugly, scratched-up cabinets, and it’s simply not a professional look. 

Placement Strategies
Typically, main left and right loudspeakers on stands are placed to either side of the stage/presentation area, aimed at the audience. (Figure 1, Option 1).

Make sure the loudspeakers are “in front” of the microphones, positioned toward the “dead” rear of the cardioid mic pattern, thus reducing the potential for feedback.

Raise the loudspeakers high enough to clear the crowd. Otherwise, people in the back will hear muffled sound because the crowd attenuates the high frequencies.

Also, raising the loudspeakers prevents blasting the front row of the audience.

Articulation is best if the direct-sound level is high relative to the reflected-sound level. This happens if you place the loudspeakers closer to the audience and take care to aim them.

An alternative to loudspeakers at either side of the stage is suggested by veteran audio consultant Ray Rayburn: Mount a single loudspeaker at one front corner of the audience, shooting across to the opposite corner (Figure 1, option 2).

Further, stack the two loudspeakers vertically (horn to horn) which narrows their coverage angle in the vertical plane. Clamp them solidly together.

The advantages of this arrangement include no comb filtering from hearing two loudspeakers at different distances, and a clearer overall sound with less reverb because of reduced ceiling reflections.

For sporting events, try to place loudspeakers so they aim across the playing field at the bleachers. This way, the players can hear what’s going on, and the people in the bleachers will absorb some of the sound and reduce reflections.

Final tips: wind screens on microphones outdoors really help eliminate noise, pop filters on microphones help prevent breath pops, which are particularly common among less experienced presenters, and a headworn mic allows a presenter to turn his/her head without getting off-mic.

Bruce Bartlett is a microphone engineer, sound mixer, recording engineer, and audio journalist. His latest books are “Practical Recording Techniques 5th Ed.” and “Recording Music On Location.”

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Wednesday, September 21, 2011

Riding The (Moving) Rails: Detailing Class-G And Class-H Amplifier Topologies

These two designs are fairly distinct

Among lesser-known amplifier classes – yet viable and employed in pro audio applications - are Class-G and Class-H topologies.

Unlike some of the “hybrid” designs touted as “unique” topologies by some in the amplifier development community, these two designs are fairly distinct.

That said, there is some disagreement among the amplifier “experts” on the exact definitions of Class-G and Class-H. For purposes of general discussion, when more than one voltage rail is employed, or a pair of rails is used in the bipolar supply case, the unit fits a Class-G designation.

On the other hand, Class-H can be thought of as a more refined version of Class-G. Some claim if the transistors for each rail are connected in series, then it’s Class-G, and if the transistors for each rail are connected in parallel, it’s Class-H.

Yet some say that if the rail voltage is switched, it’s Class-G. If the rail voltage is modulated, it’s Class-H. (We’ll talk about moving rails a bit later.), However, when the transistors are in series, it can be said that the lower transistor is actually running from a modulated supply.

The two points of view presented here disagree, and there are others not included. And some even say Class-G and Class-H represent the same thing.

The general consensus is that Class-G runs from a low voltage rail until the signal goes beyond a certain voltage, and a higher rail (or rails) is switched in. Class-H refines this to use a variable higher voltage rail (or rails), also known as a modulated rail. Let’s simplify the matter and refer to both Class-G and Class-H as “multiple rail amplifiers.”

Both are actually forms of Class-A or Class-AB as used in audio applications: Class-G and Class-H require multiple rail voltages, and both are intended to improve upon the efficiency of Class-A or Class-AB. (Note that Class-A is impractical for this type of amplifier because we’re trying to improve efficiency.)

Multiple rail amplifiers usually allow significantly higher efficiency than single-rail Class-AB designs, and thus the interest in these sub-classes. The more rail levels are used, the higher the efficiency, assuming they’re spaced properly. While impossible in reality, efficiencies can reach 100 percent with an infinite number of voltage rails.

Multiple rail amplifiers typically use only two voltage rails, which amounts to more than 80 percent theoretical efficiency at maximum power. Using more rails becomes impractical due to added power supply complexity, but with four rail voltages, efficiency at maximum power can reach 90 percent (again, theoretically).

More common in high-power pro amplifiers, the sonic characteristics of multiple rail amplifiers are varied due to the wide array of possible implementations. (They’re said to be similar to Class-AB when good design practices are used.)

However, multiple rail amplifiers suffer from the same problems as single rail amplifiers when it comes to driving reactive loads. The added dissipation usually occurring in one output transistor simply migrates another.

What does moving rail design really mean? Simply, the voltage rail moves with respect to the signal being reproduced - “tracking” it, in a sense. There are many particulars to this type of design, but the idea is to minimize the voltage across the output transistors and approach 100 percent efficiency.

Keep in mind that this is the efficiency of the output stage only. The power supply driving the output stage has its own efficiency, and the compound efficiency of the amplifier as a total package depends largely on the power supply (particularly in this type of amplifier.)

There are other caveats to contend with, such as the unpredictability of the input signal and the complexity of the power supply. The power supply rails in a moving rail amplifier can also cross zero in some implementations, allowing the output stage to act as a low-voltage amplifier at all times, regardless of output amplitude. Facilitating this type of operation is not trivial and can be quite expensive.

The power supply of a moving rail amplifier can be of the linear or switching variety as well, adding to the confusion. In the audio world, moving rail amplifiers are either modified Class-A or modified Class-AB, and suffer from the same problems when driving reactive loads. It should be noted, however, that the lowered voltage across the transistors minimizes these problems.

The hidden problem is that the power supply itself may be affected by load reactance since it has to drive the output stage. This is where the choice of power supply implementation has a profound effect on real world efficiency, and switching supplies are often the clear choice.

Tommy O’Brien is a power amplifier designer.

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Friday, September 16, 2011

Adamson Systems Unveils “Project Energia”

Key components include loudspeaker systems with networkable Class D amplifier modules, DSP, cable and power distribution, AVB network hardware with software integration of control, and 3-D simulation and diagnostics

Adamson Systems Engineering has announced the commencement of the much anticipated Project Energia. 

The project was conceived after years of analyzing inefficiencies found in sound reinforcement and with performance venue optimization. 

The key components of Energia include a new series of loudspeaker systems with networkable Class D amplifier modules, DSP, cable and power distribution, AVB network hardware with software integration of control, and three-dimensional (3-D) simulation and diagnostics. 

To ensure reliability and a smooth integration of Adamson technology, there are three phases involved in the release of the first loudspeaker system in Project Energia:

Phase 1 - Mechanical Field Testing
Phase 2 - Class D Amplifier, Power Distribution and Ground Control Field Testing
Phase 3 - Network and Network Hardware Field Testing

In July of this year, Phase 1 began when Adamson unveiled the E15 Line Source system with a series of strategic beta partners. Eighth Day Sound (USA), Wigwam Acoustics (UK), Fluge (Spain) and Big Daddy Productions (Indonesia) have all taken delivery of the E15 Line Source Array. The new system will be found on a variety of fall tours and large format events around the world. 

Additional Energia beta systems are now in place with distribution partners DV2 in France, Adamson Europe GmbH in Germany and in Singapore with long time partner Team 108.

Adamson Systems President and CEO Brock Adamson states, “Phase 1 of Energia is designed to ensure the highest acoustic and mechanical performance of the E15, while we simultaneously begin testing our new three-dimensional Blueprint simulation software. Once we’re finished evaluating transducers, sound chambers and the mechanical elements of the system, we will introduce our new Class D amplifier modules, power distribution and ground control system for Phase 2. 

“We did not want to simply re-package our old systems and offer them as though they were something new,” Adamson continues. “On the contrary, we wanted to provide something exciting that hasn’t been done. Pushing the boundaries of technology is what makes our industry great. Our team is very pleased with the feedback and support coming from our field test partners so far. We couldn’t ask for a better team of industry professionals to help us bring Project Energia to life.”

A sneak peak of the new system can be found on the Adamson web site and Facebook page.

Adamson Systems

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