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Thursday, September 02, 2010

Audio Basics: The Perception Of Sound - A Primer

A comprehensive guide covering the basics of what all engineers should understand of the perception of sound and how that knowledge to better their craft.

 
The goal of this guide is to help develop knowledge of basic acoustic principles.

In turn, this will help you to understand, and eventually master, the basic techniques of sound engineering and recording.

Each section has a theme that is first defined in technical terms and then further explained in practical terms with respect to audio.

timber
timber (pronounced /tam-ber’/) is a sound’s identity. This identity depends on the physical characteristics of the sound’s medium (the matter or substance that supports the sound).

Let’s take an A at 440 Hertz produced at 60 decibels: we can immediately tell if the sound was emitted from a violin, saxophone, or piano.

Yet, even though the instrument is different, it’s the same note and the same amplitude. The difference is in the sound production: string, air column, etc..

Plus, the sound isn’t generated by the same “tool”: a bow for violin strings, a reed and an air column for the sax, and felt covered hammers that strike the piano strings.

It’s the different physical characteristics of the medium and the « tool » that determine the characteristic sound waves in each case. Later we will also see how a sound chamber adds another dimension to this definition.

Waveform
The most basic waveform is a sine wave (sinusoid) (fig. 1). It could be considered the atom of sound. Pure sinusoidal sounds are rare (tuning forks, drinking glasses being rubbed) and were considered to have strange powers over human behavior at one time. Most sounds that surround us are of a more complex nature.

This means that inside a sound, that we perceive as being unique, there is a superposition of many sine waves that have, in a way, fused together to become one sound.

It’s the nature of this superposition itself that determines the resulting waveform (fig. 2) and that is responsible for its timber. This is called a spectrum.

Spectral Representation
There are many ways of graphically representing sound. For instructional purposes we have chosen to use a spectrogram for its clarity and simplicity.

Horizontally: time in seconds. vertically: frequency in Hertz. A sine wave (sinusoid) at 100 Hertz is represented by a horizontal line at a height corresponding to 100.

A harmonic sound at 100 Hertz is represented by superimposed lines corresponding to sine waves of 100, 200, 300: n x 100 Hertz. The length of the lines represent the length of the sound.

Noise
Let’s imagine a case where all sine wave frequencies that are perceptible to the human ear (from 20 Hertz to 20 kHertz) and having the same amplitude, are “mixed” into one sound signal. We get what is called “white noise”, or in other words “hiss”. If the white noise is very short we would perceive it to be a kind of short percussive sound.

Consonants belong to this category, in the same way that a sound medium that receives the attack of the “tool” which “kick-starts” it, produces as noise.

This noise corresponds to the time it takes for the sound wave to stabilize and take its final form. The “rubbing” of a bow on a string is similar to a hissing sound, while a hammer hitting a piano string is similar to a percussive sound.

These notions will be dealt with in greater depth when we get to envelopes and transients. In the case where a series of noise frequencies is contained between certain limits we will refer to them as noise bands.

If a zone is particularly swollen in energy, then we can speak about colored noise around that zone. Pink noise is white noise with a power density that decreases by 3 dB per octave.

Harmonic Sounds
Having already highlighted the superimposed or complex aspect of sound, we are now going to focus on a specific category of frequencies in a sound spectrum: harmonics.

A harmonic sound is a sound which contains sine waves that obey the mathematical law called the Fourier series. This law translates as follows: A complex periodic signal is made up of a certain number of component frequencies that are integers of the fundamental frequency.

An example of a harmonic sound: a sound at 100 Hertz in which the component waves are 100; 200; 300 ; 400 ; 500 ; 600 Hertz. The perceived pitch is the lowest frequency: 100 Hertz. The following component waves (2 x 100, 3 x 100, 4 x 100, etc.) are calculated on integers and are called harmonics.

The lowest frequency, on which they are based, is called the fundamental. The number , or “rank”, of a harmonic is the integer by which the fundamental is multiplied. For example the 3rd harmonic would be the one at 300 Hz. fig.5

The pitch of a harmonic sound is easily perceptible to the ear, and these sounds usually have an “in tune” quality about them. That’s why melodic musical instruments are designed with the goal of producing harmonic spectrums.

Noises, like those we referred to earlier, are aperiodic signals. They are characteristic of percussion instruments for example.

Distribution Of Energy Across The Spectrum
Regions of relatively great intensity in a sound spectrum are called formants. In the case of a band of consecutive frequencies it is referred to as a formant zone between x and y Hertz.

This distribution of energy plays an important role in the perception of timber, as do the number of components in the spectrum, their distribution, and its regularity or non regularity.

• a- violin: a hiss noise at the attack, harmonic spectrum.
• b- flute: harmonic spectrum.
• c- piano: noise of the hammer attack, percussive sound and spectrum not quite regular in its harmonics.
• d- warm sound: few harmonics but regular distribution of the energy from low to high.
• e- piercing sound: harmonic sound with a lot of intensity in the highs.
• f- Hollow sound: few harmonics in the mids.
• g- nasal sound: weak lows, intense mids, weak highs.
• h- non harmonic sound: like a non-tuned bell.
• I- square signal, odd harmonics: like a clarinet sound.

EQ
It’s the EQ section of a console that will allow us to tweak or correct timber. Depending on the model, the EQ section is more or less sophisticated and offers different possibilities of adjustment.

We won’t be dealing with simple high/low EQ knobs or switches that you can find on hi-fi amplifiers or entry level mixers which are only meant to adapt a sound to a specific listening area.

We’re more concerned with the EQ controls that are found on small modern digital models or part of most major recording software.

We must keep in mind that EQ is mainly used for one reason…to correct, and not in the hope of improving the recorded signal: you can never turn a mediocre recorded sound (due to bad placement of the mic or even the quality of the mic itself) into a great sound by just using EQ.

Equalizers split the audible frequency range( 20 Hertz to 20 kHertz…) into many sub-ranges.

Thus one generally talks about highs, medium highs, low mids, and lows. The first thing to do, then, before tweaking any knobs, is to determine in which frequency range the problem lies, then after that, the nature of the problem.

Is it due to too much coloring that wasn’t detected during the recording process, a parasite due to the environment, or a masked effect due to the presence of other instruments…

What Does It Look Like?
Equalizers are…harmonic and partial filters. Their specificity lies in the fact that they not only can get rid of component frequencies, but that they can also amplify chosen frequency zones.

Of course, if there isn’t anything in the signal in that range, only hiss will be added!

Good EQ sections generally have 4 bands. Each offers at least 2 controls: frequency adjustment and gain.

These are called semi-parametric. There’s often a third setting called the bandwidth or “Q” which has the purpose of enlarging or tightening the frequency range (bandwidth) of the filter.

When this 3rd control is present, the Equalizer is then called a parametric equalizer. Frequency adjustment will be tweakable between the upper and lower limits of the sub-range of the filter (with software these limits no longer exist!)

The gain knob defines, in dB, how much the filter will effect the chosen frequency. As we can see here in fig. 8, borrowed from cubase, this gain can be positive or negative.

We can also see that the curve of the bandwidth can be wider (hump shape) or narrower (peak shape). This shape corresponds to the bandwidth which is adjusted by the Q setting.

How To Modify timber
You must always keep in mind that all EQing on an instrument will be destructive with respect to the recorded sound, just as the latter is also, in many cases, an imperfect copy, of the original. So one must be careful!

Before touching anything, think about what you want to accomplish with EQing: I want a “warmer” sound, I want to cut the bass, I want my instrument to stand out in the mix, I want to get rid of that annoying resonance that came from the studio…

The spreadsheet below is offered to you as a kind of “quick guide” chart. It will serve as a check-list that will enable you to control and master your timber EQing. Don’t forget, however, to listen: your ears are the ultimate judge.

Advance timber Topics
We’ve discussed a “physical” definition for timber, stressed its “plural” aspect, and discussed the elements which make up and define it.

Above all we discussed the infinite combinations between these spectral components.

Now lets further this analysis by focusing on how different types of parameters such as intensity, duration, and space influence our perception of timber.

In fact, these variables are not isolated from each other: by changing one, we affect another either in a real, tangible way (which means that it can be physically measured) or subjectively ( perceived by our ears and our brains).

Perceiving timber as a whole is the result of our brains assimilating and evaluating all psycho-acoustic parameters.

The Stevens Effect
This experience concerns simple sounds (sine waves). On a given and constant frequency, for example 5000 Hz, intensity will be increased by about 40 to 90 dB.

The listener honestly thinks that the sound has become higher, about 40 Savarts, which means around a whole step (a “tempered” whole-step = 50 Savarts).

This is due to a virtual frequency variation, which only affects the human ear. For frequencies below 1000 Hz, it’s the opposite. You’d have the impression that the sound gets lower when it’s intensity increases (See Fig. 10). This is called the Stevens effect.

Consequences Of The Stevens Effect
A real change in the intensity of the spectrum’s components creates a subjective variation in timber (tone).

In the case of a complex signal, in other words the majority of sounds that surround us, when there’s a progressive variation in the intensity of a sound, each component will undergo its own Stevens effect.

You can see how our ears erroneously perceive the components of a sound when there’s a trumpet crescendo for example.

Can we still speak about harmonic spectrums if certain frequencies make us believe that they are higher or lower than they really are? For our ears there’s in fact a sensation of a real timbrel change when intensity increases.

It can therefore be deduced that all complex signals subject to a variation in intensity will be perceived as having had it’s timber altered.

The Role Of Formants
In part I, we saw that a formant is a frequency of the spectrum which is particularly strong in terms of energy.

Some wind instrument players and classical singers use this fact to gain power (or at least to give you that impression) without really producing more energy. Trying to do so would be really difficult on their lungs.

How does this illusion work? Well, it’s a little bit like the trick-question about which is heavier; a ton of feathers or a ton of lead? The instrumentalist works on his sound (timber) to give the impression that they’re increasing intensity.

Thanks to his/her technique, he/she will change the energy in the sound’s spectrum, concentrating it more specifically around 3000 Hz, there where our ears are more sensitive and react to the most weak intensities.

The sound is perceived as being stronger by the listener, while on a console, we’ll just see a small change on the vu-meter.

Real Modification Of timber & The Sensation Of Frequency Variation
When one part of the spectrum is filtered, which can happen on stage because of an obstacle, you can a get the unpleasant impression that the musician is out of tune: a little sharp or a little flat.

This phenomenon is due to the absorption of a frequency band of the spectrum by the obstacle. This “hole” in the timber can be enough to make us believe that the frequency has changed.

This effect is noticed in the case of separated sounds, which is the case of music in general. It will be more flagrant if the spectrum of the instrument isn’t very, or not at all, harmonic: for example bell sounds or a xylophone.

This phenomenon doesn’t happen when there’s a continuous harmonic sound. Our memories remember the spaces between the harmonics and the in-tune aspect is kept. Only changes in timber, due to filtering, will be perceived.

timber & Duration: The Role Of Transients
Attack: At a later time we will deal with duration, where envelope curves will be discussed in detail. But for the present it’s difficult not to speak about the notion of the evolution of timber in time and not at least stress the importance of the nature of the attack in the determining of the spectrum that follows.

Everyone knows that the tool used to generate a sound releases a certain number of components. A cymbal “attacked” with a drumstick or brushes won’t sound the same.

Sustain: Only synthesizers are able deliver a signal that’s perfectly stable in its timber for the duration of the sound. This is exactly what people don’t like about “samples” that tend to lack contrast. When dealing with acoustic instruments, timber is constantly evolving, with a certain amount of unpredictability due to physical and human aspects of playing technique.

Release: The place in which sound is captured, as well as the listening space, effects timber. We’ll dedicate some articles to architectural acoustics at a later time.

For the moment, we’ll mention that a room’s (or place’s) acoustics (or reverberation) modifies “release” and delays or shortens the time it takes for components to disappear. timber is thus either “dilated” in time or “shortened”. timber from the same source can be altered depending on the listening space, due to the fact that attenuation isn’t linear in frequency.

Law Of timber Lost According To Distance
If you hear an orchestra outside in the distance, you will first perceive the bass.

Then as you get closer, you’ll hear the mids and once your close enough you’ll hear the highs. This phenomenon comes from the fact that each frequency travels at a different speed depending on the speed of sound and the wavelengths.

Depending on how far you are from the sound source, timber is altered. The fact of getting further away is associated more with a « lack » of highs, than a decrease in the sound’s signal. A distant sound can thus be recognized by its timber.

Distortion: When you measure an amplifier’s efficiency, one of the things that is measured is Total Harmonic Distortion (THD). When a signal passes through a non-linear device, additional content is added at the harmonics of the original frequencies. THD is a measurement of the extent of that distortion.

This value is expressed as a percentage (%) and represents the quantity of undesirable content (harmonic frequencies of the signal, noise, parasites, etc.) that are added to the signal at the device‘s output. The higher the value, the lower the quality of the device.

Yet, musicians sometimes seek distortion out: many guitarists tend to love distorted sounds and distortion pedals.

If you pump a sine wave into a device at a greater level than it can take, you’ll saturate the input and create a type of musical distortion: the energy that’s lost in amplitude will transform itself into harmonic components and will enrich the timber (Fig. 13).

Figure 13 shows the result, with an incoming sine wave. If we apply a complex sound wave, the increase in richness is much greater.

The Impact Of timber On Signal Level
We’ve already seen that if we move a signal’s energy zones towards the zone to which our ears are sensitive, we get the impression that the signal has intensified.

But…when you EQ you add or take away real electrical energy that has a real effect on the signal’s level. You must therefore keep an eye on the input gain level, which you might have to adjust in order to avoid clipping.

However, you could also bring a certain sound in a mix closer by simply increasing one of the EQ levels slightly (more likely in the mid-highs…).

Changes In timber Perception Due To Distance

As we’ve said, you can make a signal more present in a mix by adding highs to it. This is, as we’ve said before, because high frequency harmonics die out sooner.

We can then deduce that a distant source will have a more muted sound then the same source which is closer.

This is one way of mixing that favors realistic parameters by altering timber instead of levels: the foreground will be more brilliant while the background will be less brilliant.

This mix’s realism will benefit from keeping a homogenous sound that would not have been achieved by just adjusting faders .

Time In Acoustics
With stopwatch in hand, our perception of time seems straightforward. But in everyday life we’re not always watching the clock, and everyone knows that the passage of time is relative.

It differs from one person to another and especially from one activity to another: An hour spent watching a great movie doesn’t feel as long as an hour in traffic.

Scientists may conceive time in seconds, but most musicians feel it in a more fluctuating manner: either in the speeding up of a tempo or the slightly off-pitch note due to stress.

In fact, pitch, which is defined by frequency, is a value linked to time and depends on our perception of a second. If it seems longer or shorter, the note can seem sharp or flat.

It’s said that during the middle ages, long before the invention of the metronome, a person’s pulse was used as the reference. It was therefore better to choose a musician who was calm.

An Experiment
For those of you who can remember magnetic tape, a piano note played in reverse doesn’t sound at all like a piano, and a verse of Shakespeare in reverse sounds strangely like…Swedish.

In fact, what our ears perceive as a single homogenous sound is really like a small train made up of four different cars: if we watch it as it moves forwards or backwards, the order of arrival won’t be the same and therefore our perception of the sound will be different.

It’s this idea that’s expressed through the notion of the A.D.S.R curve, also called envelope curve. A « reverse » preset found on some reverbs manipulates nothing but the reverb envelope. It will probably be a decreasing sound and look like figure 14. If it’s played in reverse, the end will therefore be played before the beginning (figure 14b).

A.D.
For a sound to take place, you need two agents:

• An exciter: it brings in energy.
• The excited element: that which receives the energy and starts to vibrate, creating the sound wave.

For example, in the case of a violin, the exciter is the bow, the excited element is the string. For a drum, the exciter is the drumstick, and the excited element is the skin. Each phase of the A.D.S.R. will measure the rapport time/energy of each of the four phases.

A For Attack
The Attack is the transfer time of the energy as it passes from the exciter to the excited element. The importance of the Attack is fundamental for all instruments especially percussion.

In the case of a piano (a little particular), a big part of the instrument’s timbral identity is determined precisely by the type of attack, in other words, by the player’s technique.

Wind players have also learned to develop tonguing techniques by using, as the name implies, their tongues which let them create percussive-like attacks.

When recording a voice, sometimes you get a transitory that’s too loud, either because the vocalist sings too loud, or because when speaking , a speaker clips the mic by hitting the occlusive consonants (d t p).

A compressor lets you limit the damage, but an anti-pop filter in front of the mic will more than likely fix the problem… The Kleenex in front of the mic myth, wasn’t always very effective, but we can do better now.

D For Decay
Decay could be defined as the time a signal takes to stabilize itself; we can represent it as being the difference between the initial energy of the attack and that used to maintain it.

With an instrument with a non-sustained sound, in other words, without a sustain phase, the difference between decay and release isn’t necessarily noticeable…The last phases continue without us being able to perceive them (fig. 16).

S.R.

S For Sustain
It’s the period during which energy is maintained.

There are two possibilities: that in which the sound is maintained like in figure 15 (for example, wind instruments where you have to continue blowing in order to keep the note), and a dying sound (a drum, once the skin is struck, the sound ,will last as long as the energy accumulated by the skin holds)

The first category creates harmonic spectrums, the second, non harmonic spectrums. (see article on timber).

R For Release
Once there’s no more energy coming in, the sound dies down until it extinguishes itself. For non-sustained signals, this phase just continues from the last phase. Release is rather complex.

It starts, in theory, once the musician doesn’t have any more control over the signal and finishes when the energy producing the sound is completely exhausted.

For a piano, the note continues to sound, once a key is released, if the sustain pedal is pressed and lets the strings vibrate naturally until they stop.

Plus, in a certain way we have to consider the fact that the room also effects decay. The same instrument, played in a small room won’t sound the same, in terms of decay, as it would in a cathedral. The resulting trailing effect from a long release might sound nice for certain synthetic sounds, but is difficult to control during recording.

Synthesize An Envelope

There are two specific cases for manipulating ADSR. The first brings us to the synthesizer.

The first analog models were designed to be able to adjust each section of a sound, in the image of acoustic instrument models.

In a synthesizer, the envelope filter (or EG, for envelope generator) controls a VCA (Voltage Controlled Filter) and subjects the control of the sound level to the ADSR settings.

This means, the oscillator produces a constant power signal that will be routed towards a VCA which will receive its instructions from the ADSR settings: the VCA gets modified in regards to the current that powers it.

This is delivered by the ADSR. The envelope in fig. 2 would correspond to someone progressively turning up the volume knob of an amp from zero to the chosen level (A), then reducing it quickly (D) until a second stable level (S), then progressively decreasing until zero (R ).

If you think about it, ADSR controls let us adjust the time it takes to go from A to D, then from D to S and finally, from R to zero, with the relative level of S (Sustain) coming from the difference between the end of the attack and the end of the decay.

A Little Bit Of History

This approach was for a long time considered sufficient, even if a certain amount of approximation had to be admitted: in fact, the absolute levels of A,D, and R aren’t modified; only the relative levels defined by the level of Sustain.

When the DX7 came out in the early 80s, there were finally 8 segment envelope generators that separated level from time (fig. 17).

Even if such a system has brought new possibilities in terms of adjustment capabilities and sound creation, transitory recreation through simple filters is still unsatisfactory. The digitalization of samples (Roland D50) permitted the use of real sampled attacks with synthetic sounds.

All manufacturers then used equivalent technologies on their instruments. These days, physical modeling lets us recreate an instrument’s original envelope and its interaction with the created sound , but ADSR parameters are still present to allow adjustment of the dynamic envelopes of the sounds.

Compression

The second case in which a musician-technician might find themselves confronted with having to manipulate an envelope generator: a compressor. A compressor usually has envelope adjustments that change the action time of the compression effect.

Depending on the gear, you’ll usually find an Attack adjustment, which corresponds to the time the compression kicks in once the signal reaches the limit of compression.

By putting this setting on slow, the compression will be much more discrete and lets you assure a certain amount of compression without it being too sensitive (for classical music for example).

But on the other hand, all sudden peaks corresponding to short attacks will escape treatment. A short Attack adjustment will allow the compressor to react instantly , but that typical compressed “punchy” sound will be heard. In today’s music this can be a desired and interesting effect, if used with moderation.

You can also modify the Release which adjusts the time it takes the compressor to bring the level back to its initial level. As with the attack settings, a middle setting will be more discrete and will be more delicate in bringing the level back down. The opposite, a release set to zero can, if the compressor intervenes often, give a disagreeable wave effect.

For more audio/sound related content and resources, go to AudioFanzine.

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JBL Professional CBT Column Loudspeakers Aid In Transformation Of Faith Evangelical Free Church

“We were amazed at the smooth horizontal coverage and the well-defined vertical coverage." - Heidi Samuel, Owner of Maine Pro Audio

In an audio system redesign that marks the complete transformation of a former cinema to a house of worship, the Faith Evangelical Free Church of Waterville, Maine is now equipped with a pair of JBL Professional CBT (Constant Bandwidth Technology) 70J column loudspeakers, each with three CBT 70JE extender cabinets. 

After Faith Evangelical moved into the cinema space seven years ago, it was served only by a low-quality public address system. A deeply curved wall behind the stage dampened the sound even more, making it very difficult to hear anything on or off the stage.

Invited to consult and plan the new sound design, close friend of the church and owner of Maine Pro Audio Heidi Samuel recommended the JBL CBT loudspeakers.

“The CBT product is a great fit in terms of coverage and sound quality,” explains Samuel. “We were amazed at the smooth horizontal coverage and the well-defined vertical coverage. When fully extended with the CBT 70JE, we finally have a system that provides pattern control well down in the low voice range, which we would have never accomplished with the previous installation.”

One of the biggest challenges during the installation was the inner construction of the church. Multiple fabrications were engineered to ensure the safety of the congregants as well as proper equipment fit. 

After determining the mount height of the CBT, contractors cut holes in the walls for subs and added blocking to the steel framing to further support the mounting brackets. An independent fabricator built the mounting brackets to allow for a few degrees leeway for horizontal aiming.

With limited space, the cabinets were assembled on the ground, as opposed to being constructed on the mounting bracket itself, and lifted into place. 

“The CBT speakers can save many churches looking for an improved system from overspending on a system that extends far beyond their needs, while performing on par with systems that cost much more,” continues Samuel. “The CBT’s keep going, clear out of the far wall of the lobby, 120 feet away. If the wall wasn’t there, I’m sure you could hear them well into the yard, another 50 or 60 feet away.”

Using an SPL meter, there is only a 2 dB difference between the area directly under the loudspeaker and the rear of the auditorium, 80 feet away. The ratio between the highs and lows remains consistent and the two CBT arrays cover 100 percent of the seating area with zero front fills.

JBL Professional Website
Harman Professional Website

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Maestro Sound & Lighting Deploys Adamson Y-Axis Series For Sonu Niigam Concert In Qatar

Mains, sides and fills all utilize Y-Axis modules, backed by T21 subwoofers and M15 stage monitors

Doha International Tennis Court in Qatar recently hosted a concert for an audience of more than 10,000 by acclaimed Indian Bollywood pop singer Sonu Niigam, noted for his singing performances in a number of blockbuster Hindi movies.

Maestro Sound & Lighting Company of Bahrain was the provider of sound, lighting and video backdrops for the event, with the help of tour manager Vincent Rodrigues and event manager Shafiq of Regency group.

Doha International Tennis Court was not originally designed for outdoor music events, primarily suffering from extensive reverberation issues. Maestro overcame these difficulties with the help of it’s recently acquired loudspeakers from Adamson Systems, including Y-Axis Series, T21 subwoofers, and M15 stage monitors.

Specifically, the line arrays serving the event were comprised of Y18 modules and Y10 modules. Left and right hangs flanking the main stage consisted of four Y18 with four Y10 beneath. Sides were covered by stage-located arrays of four Y18 (per side) tilted upward.

Fill was handled by an array of four Y10 per side positioned on the balcony and firing at the upper bleachers. consisting of 4 Y10 per side was installed on the balcony to cover the upper bleachers. Multiple stacks of 21-inch-loaded T21 subs were distributed in multiple stacks on the ground.

David Dohrman, Adamson Europe applications engineer, provided support for the overall system design, helping to address the numerous challenges presented by the venue. Adamson Shooter predictive software provided an assist to his efforts.

Adamson Website

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Korg Unveils microKEY USB Powered Keyboard/Controller

Portable design, 37 compact keys, dual USB hub plus extensive value-added software extras

Korg has unveiled the new microKEY USB powered keyboard and MIDI controller, offering a great way to create a compact and customized MIDI command center.

Unlike its competitors, the microKEY features a built-in two-port hub to connect other controllers, computer peripherals or Korg nanoSERIES controllers for added flexibility and control of all the users’ software. Free editing software and a number of included software bundle licenses and discount coupons make the microKEY even more cost effective.

The microKEY features 37 velocity-sensing mini keys, using the same Natural Touch keybed found on the Korg microKORG XL and microSTATION. This keybed has been designed with careful attention to the touch and feel: the proportions of the black keys and white “waterfall” keys have been adjusted for optimal playability, and the key touch makes it easy to play chords, glissandos and rapid-fire phrases. The microKEY accurately conveys the dynamics of the user’s performance to any software package.

The Octave Shift buttons extend the range of the microKEY by four octaves in either direction – up or down. The Key Transpose function allows users to play in any key instantly. Used together, these two features provide access to the entire MIDI note range. For adding expression to users’ performance, the microKEY is also equipped with both a Pitch Bend wheel and a Modulation wheel.

The microKEY serves double-duty as a USB hub. The two USB ports (Type A) allow users to expand their custom control center by adding on a Korg nanoPAD or nanoKONTROL – or any other USB device. The microKEY is compatible with Mac™ OSX 10.4 or later, along with Windows XP SP3 (32-bit), Windows Vista SP2 (32-bit, 64-bit) and Windows 7 (32-bit, 64-bit).

Users can download the free “Korg KONTROL Editor” software, available from the Korg website, and customize the microKEY for their production or performance system. With the editor software, users can set the keyboard to respond to one of eight distinct velocity curves, or set a fixed velocity value. In addition, the range of the modulation wheel can be customized by setting minimum and maximum control change values.

Weighing only 2.21 pounds, measuring (W x D x H): 22.24 x 5.47 x 2.13 inches and running on USB power, the microKEY is well suited for the on-the-go laptop musician as well.

The microKEY ships with the following software licenses and discount coupons:

1. A license to download the “M1 Le,” a limited edition of Korg’s M1 software synthesizer (included in the Korg Legacy Collection – Digital Edition) that brings to users’ computers the sounds of the ground-breaking M1 Music Workstation. Users may also choose to upgrade to the “M1 software synthesizer,” “Wavestation v1.6 software synthesizer,” or “MDE-X v1.2 multi-effect plug-in” bundles.

2. A license to download Toontrack’s software drum sound module “EZdrummer Lite,” giving users access to numerous high-quality drum sounds. Visit http://www.toontrack.com for details.

3. A discount coupon for Ableton’s “Live,” “Live Suite,” and “Live LE” DAW software, widely popular for its sophisticated functionality. For details on this software, please refer to http://www.ableton.com or http://www.h-resolution.com.

4. A license for “Lounge Lizard Session,” the physical modeling sound module from Applied Acoustics Systems, famed for its richly expressive electric piano sounds. For details on this software, please visit http://www.minet.jp/aas/lounge-lizard-session.

The Korg microKEY keyboard and MIDI controller will be available November 2010 with U.S. MSRP to be determined.

Korg Website

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Turbosound The Choice Of Charlotte Club

TCS speakers were selected to combat the clubs challenging acoustics, which included numerous curves, cement, and countless reflective surfaces.

Located in Charlotte’s Music Factory entertainment district, Butter NC is an 8,300 square-foot space with lush design concepts and a sound system that envelopes up to 400 guests with electronica, hip-hop and top 40 music.

Local design and install firm Eye Dialogue chose Turbosound TCS speakers to provide the distinctly different vibes on Butter’s two levels, each with its own DJ.

Mention the name “Butter” as a night club and most people think New York, where that nightspot revolutionized the evening out by combining a high-end restaurant with an exquisite nightclub experience on the same premises.

With his North Carolina roots and a passion for entrepreneurship, Butter co-owner Scott Sartiano decided to bring his distinctive nightclub concept back home.

“I wanted something intimate and classy, yet not overdone. It has a touch of New York, South Beach and Vegas all rolled into one,” he said.

“It’s a total visual experience; they pulled out all the stops in designing this space,” says Entertainment Technology Designer Jack Kelly of Eye Dialogue Lighting & Sound in Charlotte, who designed and installed the club’s audio systems.

“But the big thing for the owners was the sound. They wanted it to feel like the sound was all around you instead of originating from one end of the room. That presented a lot of interesting problems.”

To address this “directionless” directive within a fairly limited budget and specific limitations on the number and locations of loudspeakers, Kelly settled on a combination of Turbosound TCS Series speakers to meet the club’s goals.

“Replication of audio in a club setting isn’t like a concert or trying to reproduce a CD. People expect to hear a lot of sub,” he said. “But at the same time, it’s got to be clean and distortion-free at any volume.”

“It’s a challenge. I was looking at several different manufacturers, but when I heard a demo of these speakers, especially the subs, I really liked what I heard – big sound, tons of headroom, and super clean. And they’re really not as expensive as their reputation. They just sound that way!”

Butter is divided into two seperate areas, each with a separate DJ and sound system.

The main dance club is on the bottom floor, while the upper level houses a lounge with VIP area. Clubgoers enter on the second floor, which features curving walls and 16-foot ceilings.

“It’s a difficult space. The curves help and they softened it up with fabric treatments to make it less reverberant, but the building is an old warehouse with cement floors, so there are a lot of reflective surfaces.

Basically, we have one speaker up in each corner of the lounge, aimed down and in. The subs are on the ground, inset in the walls,” said Kelly.

The four main speakers are Turbosound TCS122s, which radiate in a 60x40-degree pattern.

The two TCS-B218 subwoofers feature dual 18-inch drivers that can reproduce frequencies down to 25 Hz at volumes up to 142 dB.

“The 122 horns delivered exactly what we needed, and those new TCS B218 boxes subs give us a really full sound, even with the tall ceiling,” Kelly enthuses.

“The sound of that sub just blows me away, and it’s really not that expensive. It’s really an awesome box.”

On the first floor, Kelly used a similar scheme for both the main dance floor and the lounge, adjusting his product choices to fit the size and shape of the spaces.

The dance floor is the loudest room in the club, requiring an immersive audio experience that remains clean and undistorted, even when the room hits 115 dB.

This is accomplished with Turbosound TCS-1561 3-way speakers with 15-inch horns above each corner of the main dance floor, radiating in a 70x40-degree pattern.

These are augmented by four floor-standing TCS-B218 subs spread evenly around the periphery to create the bass-heavy club sound the club was striving for.

The first floor lounge, a smaller space with 9-foot ceilings and a slightly more restrained vibe, employs four TCS-121C mains with a pair of TCS-218C subwoofers.

“Those are Turbosound’s Contractor Series,” explains Kelly. “The first floor lounge is a smaller space, so it didn’t require quite the kick we have on the dance floor, and these boxes were perfect for that.

With an open staircase connecting the two levels, the final key was finding a way to get Butter’s two main sound systems to play nicely together.

“Most of the time, they are playing two different styles of music. The stairwell is the key, because there’s a landing area where the bathrooms are, and that’s where the sound intersects,” said Kelly. “We had to find a way to make it the sound intelligible there without jumping through too many hoops.”

The solution was to add a single 8-inch speaker on the landing, with a wall switch to select the music from either space.

“If you reinforce the mids and highs of the music that is dominant on the landing – which varies from night to night – suddenly you’ve got clear sound,” he notes. “Theoretically, if both sources are the same volume on the landing, it wouldn’t work. But that hasn’t been an issue.”

The installation at Butter is aligned with Eye Dialogue’s emphasis on the use of new technologies and proven equipment to create solutions in both lighting and sound. “I like the scientific nature of audio design,” said Kelly.

“You’ve got a space, you’ve got a budget, and you’ve got a goal. So really, for any given situation, there’s always a best solution. At Butter, we achieved the owner’s vision of world-class audio in a visually oriented venue, and the new Turbosound TCS Series speakers were a critical part of that equation.”

Co-owner Scott Sartiano agreed. “Great sound is vital in every club, so we always use the best available,” he said.

“In this case, that meant Eye Dialogue and, in turn, Turbosound, and it has not disappointed. The sound in Butter is clean and crisp, yet powerful. You can’t help but notice the sound quality while you are there. I’d like to think it has become a welcome fixture and staple in Charlotte nightlife.”

Turbosound

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Roland Debuts OCTA-CAPTURE USB 2.0 Audio Interface

Offers eight professional-grade preamps, with each channel providing an combo-XLR input jack, phantom power, low-cut filter, phase invert, and digital compression

Roland U.S. has introduced the OCTA-CAPTURE USB 2.0 audio interface, a 10- input/10-output device for computer-based multi-channel audio production.

DOCTA-CAPTURE features eight professional-grade VS Preamps with each channel providing an combo-XLR input jack, phantom power, low-cut filter, phase invert, and digital compression. These digitally controlled microphone preamps are built with premium components, resulting in transparent audio with a superior signal-to-noise ratio.

The space-saving combo jacks also accommodate 1/4-inch connectors, with Hi-Z instrument inputs on channels 1 and 2 and balanced TRS inputs on channels 3-8. In addition, channels 7 and 8 are well suited for high input levels associated with kick drum and snare drum recording. Additional connectivity includes eight 1/4-inch TRS output jacks, coaxial S/PDIF I/O, and MIDI I/O.

The unique AUTO-SENS function makes it simple for users to set the preamps’ input levels when using multiple microphones. For example, when recording a drum set, pressing the AUTO-SENS button automatically sets the optimum input levels for all mic inputs as the musician plays the drums. Never before has it been this fast and easy to set the perfect levels for multiple inputs at once.

OCTA-CAPTURE features four software-controlled Direct Mixers, which allow users to set up independent monitor mixes and route them to headphones, loudspeakers, or other recording and mixing devices. The internal 40-bit DSP engine provides superb audio quality, and any input can be routed to multiple outputs for the ultimate in monitoring flexibility.

Roland’s VS Streaming driver offers the latest technology for powerful low-latency performance with 48-sample ASIO buffers (44.1kHz/48kHz) and unprecedented stability.

All popular DAW platforms are supported via ASIO 2.0/WDM (Windows) and Core Audio (Mac) driver versions. The VS Streaming driver is compatible with the very latest computer operating systems, including Windows 7 and Mac OS X 10.6.

OCTA-CAPTURE can also be used as an expansion I/O unit with Cakewalk’s V-Studio 700 and 100 systems. Thanks to VS Streaming, multiple devices (two OCTA-CAPTURE units or one OCTA-CAPTURE plus a VS-100 or VS-700) can be used without compromising stability, performance, or audio quality.

OCTA-CAPTURE can easily be used as a desktop interface, or mounted in a single rack space with the included rack ears. It comes bundled with the Cakewalk Production Plus Pack, a powerful software suite that includes SONAR 8.5 LE, Rapture LE, Studio Instruments Drums, and more.

The OCTA-CAPTURE is expected to ship in October with an MSRP of $699.

Roland Connect Website

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NOA Audio Solutions To Debut NOA Record 4 Digital Migration System At IBC2010

Record 4 integrates with other NOA workflow systems to form a unified solution for digitization and archive management.

NOA Audio Solutions will introduce NOA Record 4 at IBC2010, part of its IngestLine family of systems for fast, cost-effective, accurate migration of audio from analog to digital storage.

Developed in response to the needs of enterprises with a great deal of audio stored on long-playing magnetic tape, vinyl, or both, the NOA Record 4 extends the speed and capacity of the system by one-third — from three to four parallel stereo streams.

“Our customers asked, and we are responding,” said Jean-Christophe Kummer, NOA managing partner.

“NOA’s MediaLector and CD Lector products already meet the mass migration needs of customers with large archives stored on DAT and CD.”

“Now, we add higher throughput to the Record system to suit customers whose analog audio archive material is currently carried on cassette, vinyl, open reel tape, and 78s.”

NOA Record integrates seamlessly with both NOA mediARC and JobDatabase workflow systems to form a unified solution for digitization and archive management.

Working within the mediARC or JobDatabase environment, NOA Record 4 can increase throughput from the current 2.7 hours of sound digitized per hour to 3.6 hours of sound digitized per hour — an impressive 33 percent increase in capacity.

The NOA Record ingest system comprises NOA Record software, the N6071 Workstation, and N6000A hardware and replayer communication modules. The NOA Record features tightly integrated software-controlled A/D converters that pack an excess of 125 dB dynamic range.

With an easy-to-read user interface for multiple simultaneous sources, NOA Record displays relevant recording and playback parameters of all stations on a single screen divided into section panels, thereby affording the user an at-a-glance view of the most important functions. More features include Traces Aided Spot listening (TAS), remote control of hardware, and the world’s top Azimuth monitoring tool.

NOA will demonstrate its upgraded NOA Record system at at IBC2010 in Amsterdam, the Netherlands, Sept. 9-14 and NOA Record 4 will be released on Sept. 30, 2010.

NOA Audio Solutions Website

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Tech Tip Of The Day: Creating A Spacious Rhythm Guitar Sound

I tracked my rhythm guitar in mono but I think it needs to found more full. What can I do?

 
Q: I’m working on an album and I’m really having alot of trouble getting the rhytmeh guitar to sound just “right”.

What i’ve discoverd is that it needs to sound more open, with maybe a more stereo feel.

However, the problem I’m having is that when I tracked the instruments I only threw one mic on the amp.

Of course, basically everything else was recorded stereo!

So, now I’m at a loss. What can I do?

A: No doubt about it, this is one of those things that most easily could have been solved in tracking.

Back when tracks were valuable real estate (stop laughing, that time existed!), few instruments got recorded in stereo. Instead, they were recorded in mono, then panned to a position within the left-to-right sound field.

Today, nearly everyone has access to more than we ever dreamed possible, and those holds true whether you’re using a hardware or software based DAWs.

All this really just means that we have more at our disposal to create a more spacious (and more natural) stereo spread, on really most anything, as track count becomes less and less of a consideration.

Since the rhythm guitar is often the instrument around which all other tracks are built, it’s important to give it that big sound field.

Thankfully, however, there is a way to accomplish this aside from initially recording the instrument in stereo.

With an electric guitar, you can usually use a multi-effects processor to convert a mono input signal into an enhanced stereo output via a stereo chorus or panning tremolo effect or by using a delay in which the left side delay is shorter or longer than the right side delay.

Some processors allow you to assign a dry signal to one output and the delayed signal to another.

By using a very short delay, you can fool the listener’s ear into believing it’s hearing two guitars, just because the left image is offset in time a bit from the right.

As always, we welcome input from the PSW community and would love to know how you would solve this mixing issue. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

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Meyer Sound Constellation Provides Texas High Schools The Ideal Performance Environment

The auditoriums of two Texas HIgh Schools have utilized Constellation to improve the success of their students and the quality of performances.

The award-winning music programs at two Texas high schools—one in the community of Spring and the other 140 miles away in Temple—are both powered by Meyer Sound Constellation acoustic systems.

According to the schools’ music faculties, Constellation’s flexibility and natural acoustical characteristics not only enhance the value of the auditorium as a community resource but also improve the learning experience and even lend a competitive advantage to the school’s choirs as they prepare for interscholastic competitions.

The active acoustic systems—a keystone element in the auditorium renovations at both schools—were specified by BAi, an Austin, Texas-based acoustical consulting firm. Hairel Enterprises of Conroe, Texas was awarded the installation contract for both systems following a competitive bid process.

Constellation’s enhancements at Temple High School’s 1,200-seat venue quickly earned praise from James Pfeiffer, director of fine arts for the Temple Independent School District. “Constellation really opens up the room to whole new realms of what we can do,” he said.

“We can instantly adjust the acoustics for any type or size of ensemble, from a full choir with orchestra to a large band, to a small jazz combo, or a pop choir of 12 singers. It transforms the auditorium into a tremendous resource for all the schools in Temple and our surrounding community. It just gives us enormous flexibility.”

At the heart of the Temple system are the primary Constellation processor and three VRAS reverberation processors.

Acoustic energy throughout the room is picked up by 22 miniature cardioid microphones and, after processing by the patented VRAS reverberation algorithms, by a total of 95 self-powered Meyer Sound loudspeakers and 12 subwoofers.

The system was configured to provide six presets that allow the room reverberation to vary from its nominal 1.3 seconds to 3.5 seconds for choral music. A stage system supplies early reflections to create an active electronic orchestra shell.

The overall effect, said Pfeiffer, is a dramatic transformation of both the sound and the utility of the space. “It naturally balances the sound in the auditorium, so you hear the same thing in every seat. And it works everywhere, even when the performers are up in the balcony. It’s really hard to describe. You have to hear it to appreciate what it does.”

Constellation at Spring High School is similar in many respects, though scaled to a 700-seat auditorium. The primary processor and two VRAS processors work in conjunction with 24 microphones, 52mid-high loudspeakers, and eight ultra-compact subwoofers.

Six presets allow the room’s mid-band reverberation time to vary from 1.5 seconds with the system off to the longest setting of 4.2 seconds. This “very long” setting was included at the request of David Landgrebe, Spring’s assistant choral director.

“Our choirs compete in regional competitions usually held at a newer church that has the acoustics of a European cathedral,” said Landgrebe. “

You don’t get a chance to warm up on stage, so for some younger singers that really throws them for a loop. Being able to create that very live acoustic here for rehearsals definitely gives us an advantage in preparing our students for the event.”

The Constellation systems at both schools includes the voice lift feature, which augments the early reflections from the stage to better disseminate vocal sounds throughout the auditorium—reducing the need for a conventional PA system.

Both schools have found the effect particularly useful for dramatic presentations and musicals. “It projects voices into the room much better,” comments Landgrebe,” and that helps negate the problematic wireless microphone issues we’d have to deal with otherwise.”

The acoustical redesign at Temple was directed by BAi President and Principal Consultant Charles Bonner, with assistance from Senior Acoustician Andy Miller. BAi Principal Consultant Richard Boner headed the team for Spring High School, also with Miller’s assistance.

A range of compact loudspeaker models were chosen for the Temple installation, including 26 Stella-8C and 31 Stella-4C, 20 MM-4XP, nine UPM-1P, and nine UPJunior VariO loudspeakers along with 12 UMS-SM subwoofers.

The Spring installation used 20 Stella-8C, 20 MM-4XP, and 12 UPM-1P loudspeakers, in addition to eight MM-10 subwoofers.

Meyer Sound Website

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TC Group Americas Names Frank Loyko As Vice President Of Sales For Tour & Install

Industry veteran brings more than 30 years of sales and marketing experience to new role

TC Group Americas Inc., distributor of TC Electronic, TC Helicon, Dynaudio Acoustics Tannoy, Lab.gruppen, Linn and Audica, has appointed Frank Loyko to the position of vice president of sales for tour and install.

Most recently, Loyko served as president of RCF USA, and prior to that, he was the director of live sound for Digidesign, a division of AVID Techonolgies. He was also vice president of sales at EAW, and then served as senior vice president of worldwide sales for LOUD Technologies.

”Frank’s background and experience are a great blend with the extensive talent we have here at TCGA,” states Marc Bertrand, CEO of TC Group Americas. “It really is a great opportunity for us to take our organization and specifically, the two market verticals he’s responsible for, to greater levels of sales, support and market position more quickly and effectively.”

Loyko adds, “My philosophy has always been to surround myself with the best in the industry. The TC Group has a strong, professional and knowledgeable team with superior products.”

TC Group Americas Website

 

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NEXO Loudspeaker Demo At Paramount Arts Centre Theatre

Demo to showcase NEXO and Yamaha Commercial Audio products including loudspeakers and audio consoles.

Yamaha Commercial Audio Systems has announced it will hold a NEXO Loudspeaker Demo at the Paramount Arts Centre Theatre in Aurora, Illinois.

The event is free of charge and open to all audio professionals, and will be held on Friday, September 10 from 10:00 am – 4:00 pm.

The demo will showcase both NEXO and Yamaha Commercial Audio products including NEXO GEO-S12 Loudspeaker Array System, NEXO RS15 Dual-15” Subwoofer, NEXO RS18 Dual-18” Subwoofer, NXAMP 4X4 Controller Amplifier, and NEXO PS15R2 Loudspeakers.

Yamaha Commercial Audio Products featured include Yamaha M7CL-48 Digital Mixing Console and SB168-ES Digital Stage Box.

As well, EtherSound and DANTE Networking will be used throughout the demonstrations.

The Paramount Arts Centre is located 23 East Galena Boulevard in Aurora, Illinois.

Yamaha Commercial Audio Systems Website

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Electro-Voice The Choice Of Ohio’s GaREAT Complex

Paladin Professional worked closely with C. L. Pugh & Associates to provide the best design and installation possible at this GaREAT facility.

Geneva, Ohio is a small city with a history of thinking big. So it’s no surprise that the local GaREAT Sports Complex, operated by the non-profit Geneva Area Recreational, Educational, Athletic Trust, is an ambitious undertaking.

Started just two years ago, the multi-sport recreational facility on a 175-acre campus already encompasses more than 450,000 square feet of indoor fields, courts, and tracks for year-round practices and competitions, as well as an outdoor stadium with a field and a track.

Sound throughout the facility, which also includes dining and meeting rooms, has been handled by Paladin Professional Sound of Valley View, Ohio, working in close cooperation with manufacturers’ representatives C. L. Pugh & Associates of Brunswick, Ohio.

With two decades of experience serving clients including the Rock and Roll Hall of Fame, the Cleveland Browns, The Ohio State University, Quicken Loans Arena, and Kent State University, Paladin is no stranger to the design, installation, and service of large-scale systems.

Even so, Paladin’s John Davidson, who led the design and installation effort, is impressed with the high bar the Trust has set for the GaREAT Complex. “They’ve got a facility that is second to none in the state of Ohio,” Davidson says, “and they want sound that is as state-of-the-art as everything else.”

To meet GaREAT’s standards, Paladin has relied on sound systems built around Electro-Voice loudspeakers, amplifiers, and controllers. “Electro-Voice is my preferred speaker line,” said Davidson.

“For projects that merit modeling in EASE, I will always choose to model a system based on Electro-Voice over anything else, unless there’s no EV product that fits a specific application. And when we get an EV system installed and running, it not only covers the way it should according to EASE, but more often than not we are surprised that the overall sound is even better than we expected. So, having a positive history with the EV line, I’m not typically going to choose anything else.”

A case in point is the latest stage of the multi-phase build-out of the GaREAT Complex, which included installation of the sound system for the stadium.

Used for football, lacrosse, soccer, and track, the stadium is home to the region’s semi-pro football team and also hosts practices, games, and events for local leagues and high school and collegiate teams. The outdoor system was configured as a combination of wall-mounted and pole-mounted Electro-Voice loudspeaker systems.

“Geneva is in the snow belt,” Davidson says, “and these speakers are going to have to stand up to some of the harshest weather in the country. These EV speakers can take the weather extremes, and they can also be pole mounted, which was an important consideration in our design.”

For bleacher coverage, Paladin chose 18 Electro-Voice ZX5 15-inch, two-way, composite-enclosure loudspeakers – nine each on the stadium-facing walls of the buildings that bracket the long sides of the space.

“We wanted the system to sound noticeably better than what people would typically run into, even at college and professional sports facilities,” said Davidson, “and the sonic quality of the ZX5 is awesome. We also needed speakers that don’t weigh too much, to avoid harming the steel siding, so the composite body of the ZX series is a great advantage. And the design of the mounting bracket also helped us achieve a stable mount without damaging the siding.”

For on-field coverage, including events for which a stage is set on the field and amphitheater seating is set up in an end zone, Paladin pole-mounted more composite-enclosure systems from Electro-Voice: a total of eight SX600 high-output, dual 12-inch, two-way, full-range systems and six Electro-Voice Sb122 12-inch subwoofers (in a directional cluster of three in each end zone).

The setup is powered by Electro-Voice CPS 4.10 Contractor Precision Series amplifiers.

“The CPS 4.10s are perfect for the job,” Davidson says, “because each amp gives us four channels of 1000 watts each into 4 ohms, so we need only one amp for each pole.”

The amplifiers are each outfitted with optional RCM-810 remote control modules that allow them to be operated via an Electro-Voice NetMax N8000 digital matrix controller that Paladin set up to give the customer easy touch-screen control over the system’s various zones.

“Because of the cable distances involved,” said Davidson, “a football field would typically be handled with a constant-voltage, high-impedance distributed system, possibly 100 volts. But because we had the ability to control the amplifiers via NetMax, we were able to put each amp into a thermally-controlled Hoffman enclosure at the base of each pole and drive all the speakers at low impedance.”

By eliminating step-up and step-down transformers, we have a signal that is much cleaner and has no core saturation at higher levels. NetMax allows us to make the system as high-fidelity as possible.”

The overall result, Davidson says, is “fantastic — really good sound. There’s not a bad seat in the house, sonically. Every seat is consistent within plus or minus 3dB.”

“You’ve got plenty of bass thump for modern music, incredible clarity from the announcer, the wireless referee mic is crystal clear, and there are no feedback issues whatsoever. It’s just an awesome system. We’ve now done four different Electro-Voice systems for the GaREAT Complex, and they keep coming back to us for more.”

Electro-Voice Website

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