Wednesday, September 28, 2005

MIX Online Review: RAD2 Remote Angle Display

As seen in (Professional Audio and Music Production Magazine, January 2004)and Sound & Video Contractor Magazine (September, 2003 issue)

The RAD2 Remote Angle Display with dual sensors, the latest addition to Rieker’s RDI Series of digital inclinometers, was specifically designed for speaker and speaker array positioning. This battery-powered digital LCD box can be matched with a number of Rieker sensors, which can then be used interchangeably with the display box to coordinate multiple remote speaker locations.

“Our customers in sound system installation or audio/video contracting typically have a number of speakers that they need to level and only want one display box to configure the array,” said senior engineer Joann Coates. “We can supply the LCD as a single-line display reading one sensor at a time [RAS] or a dual-line display to measure two sensors at the same time [RAD].” Both sensor and display box use standard-type audio XLR cord plug connectors for connection. In most cases, the sensors are permanently mounted on the speakers with cables running down to the floor. The cable connector is then plugged into the display box as needed.

The sensor ranges come in either ±30° (60° total range) or ±70° (140° total range). The display can be calibrated to anywhere within the total range of the sensors specified. Skip Gosnell, the marketing director for Rieker, said, “We have a number of V-DOSC or similar line array installers, for example, who are purchasing the RAD2 dual-line display with the REL feature being programmed to read top–bottom and selecting the ±70° sensors and having them scaled to read +50° to -90°.”

The REL button (relative zero) allows the operator to temporarily zero the digital readout to obtain relative angle changes. The Min/Max function provides the smallest and largest angle that the device has sensed since it was last reset. The optional Relative Difference function can be programmed to give the difference between the top and bottom sensors.

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Tuesday, September 27, 2005

INCLINOMETER (in•cli•nom•e•ter)

the definition of...

An Inclinometer (or Clinometer) is an instrument for measuring angles of slope (or tilt), elevation or inclination of an object with respect to gravity. Also known as a tilt sensor, tilt indicator, slope meter, slope gauge, gradient meter, gradiometer, level gauge & level meter.

INCLINOMETER (in•cli•nom•e•ter)
Pronunciation: "in-kli-'nä-me-ter
1. An instrument used to determine the angle of the earth's magnetic field in respect to the horizontal plane.
2. An instrument for showing the inclination of an aircraft or ship relative to the horizontal.
3. An instrument for showing a deviation from the true vertical or horizontal
4. An instrument used by surveyors in order to measure an angle of inclination or elevation [syn: clinometer]
Sound Reinforcement: To check a level as compared with a horizontal plane, or to measure surface inclination or tilt of line array. To assist in setup and alignment for proper sound quality.
Transportation: For measuring the rolling and pitching of a ship or the banking of a road curve.
Mining: A. An instrument for measuring the inclination or slope, as of the ground,

Syn:clinometer b. Any of various instruments for measuring the departure of an object from vertical; a driftmeter.

1. Any device that receives a signal or stimulus (as motion or inclination, etc.) and responds to it.

Friday, September 16, 2005

How does an inclinometer work?: Part 1

An inclinometer (also known as a tilt indicator or level indicator) are mechanical or electronic tools that measure the inclination of a surface relative to the earth's surface (gravity). Levels vary from simple mechanical devices to complex electronic sensors that digitally readout angular level values (as does the RAD2 Remote Angle Display). Mechanical levels use a bubble or ball in a vial (typically curved)-filled with a damping fluid, or a pendulum to indicate level.

Electronic or digital inclinometers put out an electronic signal proportional to the angle of lift, relative to level (or horizon). Internally, the inclinometer consists of a tilt sensor and signal-conditioning electronics. These normally reside in one enclosure with either a cable or connector.

Tilt sensors and inclinometers generate an artificial horizon and measure angular tilt with respect to this horizon. They are used in cameras, aircraft flight controls, automobile security systems, and special switches - or any application that requires a tilt or angle measurement. Important specifications to consider when searching for tilt sensors and inclinometers are the tilt angle range and number of axes. The tilt angle range is the range of desired linear output measured in degrees. The number of axes the inclinometer and tilt sensor measure on is another important specification.

Common sensor technologies for tilt sensors and inclinometers are accelerometer, capacitive, electrolytic, gas bubble in liquid, mercury, and pendulum.

Rieker's RAD2 system uses liquid capacitive inclinometer sensors. Since this type of technology has no moving parts (nothing to break) and has a natural built-in shock and vibration resistance, the sensors are extremely durable for rough environments.

Understanding Line Array Systems: Part 2

By John Murray

...Understanding Line Arrays> continued...


Isophasic aperture is my current favorite high-tech term. It describes the phase characteristic of the slot that loads the horn bell of some line array box HF sections. The perfect line array driver, particularly for very short wavelengths, is a ribbon driver like those used by SLS Loudspeakers. Compression drivers are more rugged and capable of higher output levels than a ribbon driver, but they do not have a linear phase signal at the mouth of a horn.

Ideally, the signal at both the top and bottom of the driver’s horn mouth would arrive in-phase with the signal at the center of the horn mouth to mimic the ribbon driver’s characteristic. Since the center of the horn is closer to the driver’s diaphragm than the top and bottom, the more central paths to the horn from the driver must delay the signal to arrive in phase with the longer paths to the top and bottom of the horn. There are two ways to accomplish this.

The first is to make the path length progressively longer towards the center of the horn via a phase-plug type of device. This technique was employed in the old JBL “slot tweeter” super-tweeter and was adapted by Heil in the V-DOSC system for wavelengths at 1000 Hz and up. Other line array manufacturers have employed similar devices.

The other method is to use variable density foam, which slows the speed of sound through the more dense foam medium towards the center of the horn. Electro-Voice and McCauley use this technique to provide an isophasic horn section in their line array offerings.

Perhaps the most interesting technique for an isophasic device is the patented mid-high frequency aperture by Adamson. It employs the longer path length method, and utilizes directional vanes to prevent excess vertical dispersion as well. This approach is used for both the highand mid-frequency sections of their line array systems. The mid-frequency energy exits via two vertical slots on either side of the high-frequency exit slot. The paths of the mid-frequencies curve around the HF chamber housing. All slots are isophasic.

With the slots of the MF section on each side of the HF slot, diffractional problems of each slot on the other could be very problematic. However, Brock Adamson came up with a unique solution: overlapping the crossover points between the mids and highs. This provides in-phase pressure fronts from the other slots to prevent diffractional interference in the frequency range where it would be a problem.


The term “tapering” is also commonly called “shading.” They are essentially interchangeable. One of the first tricks used to take advantage of the line array effect was frequency tapering. My earliest exposure to this technique was the Electro-Voice LR-4B column speaker. For low/mids, it used 6-inch by 9-inch cone drivers that had lowpass filters at successively lower frequencies for speakers placed farther out to the ends of the column. This resulted in a longer column at longer wavelengths and a shorter column at shorter wavelengths, producing a similar dispersion pattern and critical distance for all frequencies, which in turns provides a more balanced frequency response at all listening distances.


Another tapering/shading technique is amplitude shading. This is used in many current line array products to accomplish front fill coverage where the bottom hook of a J-Array covers the extreme near-field listeners. This technique is simply lowering the volume of the loudspeakers covering the nearfield seating with respect to the longthrow loudspeakers higher in the array.


Some line array systems offer more than one choice for vertical dispersion of the individual box elements in the array. They do this as a solution to cover the near-field and extreme nearfield seating in most venues. EAW has gone one step further by offering two different models, matching the vertical dispersion and output level so that the drivers produce equal mouth SPL throughout the array. They avoid any amplitude shading for the drivers covering the closer listeners by increasing the coverage angle of those box elements. Why is it important to avoid amplitude shading?

According to David Gunness, EAW director of research and development, whenever two wave fronts with different pressures are combined, there will be a discontinuity at the juncture of the two. This discontinuity will be audible as though it were a separate, non-coherent source (delayed loudspeaker). The result is transient smear and uneven frequency response. Divergence shading provides a wave front whose curvature varies, but whose pressure magnitude does not. Therefore there is no introduced time smear to the signal.


The majority of available line array systems are horizontally symmetric. Ideally, each band pass is a 1/2 wavelength wide strip that runs the entire length of the array. The advantage is that it avoids horizontal lobbing at the crossover-frequency band. It also requires symmetric pairs of inner mid and outer LF drivers flanking the HF sophistic ribbon.

The drawback to this approach is that for the mid-drivers to be within 1/2 wavelength of each other, they must be incorporated into the bell of the HF horn. The normal 90-degree angle causes reflections between the MF drivers and the discontinuous horn walls cause HF problems as well.


EV, Meyer (on their smaller system), and NEXO have opted for an asymmetric design. This approach avoided the mid-frequencies in the horn bell problem and contends with the horizontal lobbing at crossover problem inherent in asymmetric designs. Choose your poison.


Line arrays have great directional control in the vertical axis. Subwoofer systems, by nature of the very long wavelengths involved, do not have any directional control unless arrayed. Even then, because of the omni-directional nature of each element in the array, there is no front-to-back directionality. This causes muddiness on stage and low-frequency feedback problems. Enter cardioid and hypercardioid low-frequency sections.

Cardioid and hypercardioid loudspeaker systems are similar to microphones, just in reverse. In the case of loudspeakers, two transducers, separated by an exact distance within the enclosure, with delay on the rear driver, create the directional radiation pattern. The cardioids type has maximum level cancellation straight back at 180 degrees behind them and the hypercardioid have maximum level cancellation at about 120 degrees off-axis. As examples, Meyer employs cardioid low-frequency sections while NEXO employs the hypercardioid.


IIR (Infinite Impulse Response) filters in a DSP processor act just like analog crossover and equalization filters. Their amplitude and phase characteristics are in a fixed relationship. So much boost or cut produces an exact corresponding change to the phase response.

FIR (Finite Impulse Response) filters are able to manipulate phase independently of amplitude and correct for distance-related cancellations between drivers if each driver is under individual DSP control. Some systems, like Intellivox, employ separate DSP processing and amplification for each driver in the array. These types of systems will define the next big step forward in loudspeaker technology.


So, the next time you want to impress the ladies at the local hall, tell ‘em “We’re gonna hang a logarithmicspaced, articulated spiral array in a horizontally asymmetric configuration employing frequency tapering and divergence shading, which will include isophasic high-frequency and mid-frequency apertures, hyper-cardioid low-frequency transducer sections, is controlled by finite-impulse response filtering digital signal processing, and works well with a psychoacoustic infector.” You might just get lucky…

Live Sound Technical Editor John Murray is a 26-year industry veteran working for EV, Midas, MediaMatrix and TOA. John has presented two AES papers, chaired three Syn-Aud-Con workshops and is a member of the TEF Advisory Committee and ICIA adjunct faculty. If you have a question you’d like to ask John, e-mail him at .

Wednesday, September 14, 2005

Understanding Line Array Systems: Part 1

By John Murray

...this is a great article by John Murray/ProSoundWeb that provides a great overview of the different types of line arrays...

Behind the buzz, there are a lot of factors at work in line arrays. An explanation and comparison of current models.

At last count, I found at least 19 companies offering line array loudspeaker systems that are more than simple column designs. Rather than discussing over a dozen different product types, I thought we might approach the subject by defining the technological terms of line arrays. This way, we get a better grasp of the issues involved with line array systems and will be able to discern both the similarities of, and unique differences between, the products being supplied by manufacturers today.

This discussion can’t be contained in just a few paragraphs, so we must start with the more basic issues of line arrays and then follow with more esoteric topics that build on these basics.


Line arrays have been around for over a half of a century as column speakers, and other than those made by Rudy Bozak here in the US, most were voice-range only. Their application was generally for highly reverberant spaces, where a narrow vertical dispersion avoided exciting the reverberant field, provided a higher Q (narrower dispersion pattern) and, as a result, improved intelligibility of the spoken word.

Never losing popularity in Europe as they did in America, it’s no wonder that L-Acoustics V-DOSC loudspeakers from France were the first to show the concert sound world that more level and smoother frequency response can come from fewer drivers in a line array. After everyone realized that for a given listening area, the drivers have no destructive interference in the horizontal plane and combine mostly inphase in the vertical plane, the race was on.


Basically, a line of sources will create a wavefront of sound pressure that is loosely cylindrical in nature at a particular range of wavelengths (frequencies). Its idealized shape is actually more like a section of a cake, and the wavefront surface area, as it expands only in the horizontal plane, doubles in area for every doubling of distance. This equates to a 3dB SPL loss of level for every doubling of distance.


An idealized point source, imperfectly represented by a loudspeaker or nonlinear cluster of loudspeakers, radiates in a spherical waveform rather than cylindrical. This wavefront expands to four times the area with each doubling of distance, which equates to a 6dB SPL loss for every doubling of distance. This is commonly known as the inverse-square law, and it applies to all point-source radiant energy. Hence the big advantage for a line array is that for a given number of transducers, the resulting long throw level can be much greater than for a non-line array, or point-source, loudspeaker system.


This is the term applied to the dispersion pattern, or response balloon of a line array. It simply means that when you stack a bunch of loudspeakers, the vertical dispersion angle decreases because the individual drivers are outof- phase with each other at positions off-axis in the vertical plane. The taller the stack is, the narrower the vertical dispersion will be and the higher the sensitivity will be on-axis. In the horizontal plane, an array of like drivers will have the same polar pattern as a single driver. Some believe that the horizontal pattern is wider than for a single driver, but they are mistaken, likely fooled by the fact that the level is louder off to the side due to the higher sensitivity of multiple drivers. However, the actual polar pattern remains the same as for a single driver.


In addition to the narrowing vertical coverage angles, the array length also determines what wavelengths will be affected by this narrowing of dispersion. The longer the array, the lower in frequency (longer in wavelength) the pattern control will occur.


There is a limit to the 3dB per doubling loss, and it’s at this point where the array is far enough away to appear to be more of a point source and its level begins to attenuate according to the inverse-square law at 6dB per doubling of distance. The transition between these two regions is known as the critical distance for the line array. The region closer than critical distance, and the region beyond it, is termed as the Fresnel and Fraunhofer regions, respectively, so named by Christian Heil of L-Acoustics. Unless you’re a true math dweeb, near-field region and far-field region roll off the tongue a bit easier.

The critical distance for a given line array length varies inversely with wavelength (frequency). This was also discussed in depth in the last issue. Shorter wavelengths (higher frequencies) have much farther critical distances than longer wavelengths (lower frequencies). In theory this means, at greater distances, a line array will maintain more high-frequency content than low. However, air attenuation of the highs will counteract this characteristic.


Articulated is the ten dollar term for curved. This describes the very-popular J-Array shape that most manufacturers currently offer, save one. To date, the Duran Audio Intellivox system is the only line array that covers from extreme near-field to far-field seating with a straight-line dead-hang approach. (Talking about articulated arrays with your clients is what gets your day rate increased and your job title changed from “sound tech” to “audio engineer.”)


This is also a term for curved arrays of a particular type. Spiral arrays describe a curve that is increasing in the rotational angle from one end to the other, just as the common J-Array does from top to bottom.


Mark Ureda, consultant to JBL, mathematically determined that spiral arrays that increase their angle of curvature in even increments perform better. For example, at the top of a line array, the splay between cabinets is 0 degrees. Going down the array, the element boxes are successively splayed at 1 degree, 2 degrees, 3 degrees, etc. Or it could go in increments of 2 degrees (i.e.: 2 degrees, 4 degrees, 6 degrees, etc.). These are arithmetically increasing spiral arrays.


Lobes describe all the acoustical energy that emanates from a loudspeaker or group of loudspeakers. The specified coverage angle of a horn is its main lobe. Spurious lobes are those that emanate out in a non-useful direction from the source.


Much ado has been made about lobe steering. Visions come to mind of FOH guys moving loudspeaker coverage around with a joystick. Lobe steering is generally done by incrementally delaying drivers in a line array. This can only be done when the sources, (the drivers), are about 1/2 wavelength apart for a given frequency, and only in the direction of the line array’s axis. For typical live sound HF drivers with a 9-inch diameter, this means that they cannot be positioned close enough together to steer anything above 750 Hz. However, using adaptive apertures to mimic a long line of smaller sources enables some steering at shorter wavelengths.


Side lobes are artifacts of line arrays. They are called side lobes but actually emanate from the ends of the array, at the top and bottom, as a typical line array is viewed in use. They are caused by the individual elements being in-phase at a particular angle and wavelength at some off-axis position from the array’s main lobe. It is possible to eliminate side lobes, but there are limits and consequences to side-lobe elimination in line arrays.


This is a synonymous term for side lobes. Gradient describes how these lobes occur at particular angles or grades with respect to the line array’s orientation. Professional progress terminology tip: use gradient side lobes rather than side lobes in your technospeak. Chicks dig it.


Another of the fundamental parameters of line arrays is the spacing between individual elements. The accepted limit is that for good line array behavior, the sources should be no more than 1/2 wavelength apart for a given frequency. This means that loudspeakers reproducing longer wavelengths can be spaced farther apart without any deterioration in performance. But since 1/2 wavelength at 15 kHz is just under one-half of an inch, HF devices can never be close enough. One manufacturer maintains that because of this, line arrays do not really work at very high frequencies. However, I disagree, because even at very short wavelengths, the 3dB loss per doubling of distance still holds true, and this is what defines the line array effect. (In my humble opinion.) What does result from driver spacing of more than 1/2 wavelength is more pronounced gradient side lobing.


Duran’s Intellivox Series line array loudspeakers employ the logarithmic driver spacing technique. This provides denser driver spacing at short wavelengths and economizes on the number of drivers needed for longer wavelengths by spacing them in larger and larger logarithmic increments.

Live Sound Technical Editor John Murray is a 26-year industry veteran working for EV, Midas, MediaMatrix and TOA. John has presented two AES papers, chaired three Syn-Aud-Con workshops and is a member of the TEF Advisory Committee and ICIA adjunct faculty. If you have a question you’d like to ask John, e-mail him at .

Tuesday, September 13, 2005

Speaker Positioning: Part 2

Speaker Positioning (part 2): One thing that will affect the sound from your line arrays is overall position when hanging: size and space in relation to the venue, the surrounding stage objects, and what if anything is blocking the sound. Experimenting with positioning offers near limitless possibilities for tweaking your sound, however, unless your paid by the hour (and have a day or more) this may be a bit consuming.

Hanging speakers in a real world space is not brain surgery, however it does require a bit of planning and some knowledge of basic geometry. Sound engineers typically need a basic floorplan, distances from the stage the sound will be traveling, and where various objects are likely to block the sound. Not all line arrays use the same sound producing technologies, however, it helps to think of invisible cones of sound emanating from the speakers themselves - the angle of every speaker in the line array will be pointing along an individual axis based on the overall angle the line array is hung. Anything coming in contact with that cone before and after it reaches the audience is going to affect the sound. Things likely to come in contact with that cone of sound include walls, floors, seats, and columns (or structural supports). Try to keep anything from blocking sound before it reaches the seating area. Obvious problems might include a part of the stage in front of a speaker, even if it is below the speaker it will affect the ability to move the air in front of it.

Speaker positioning basics:

* Experiment and then experiment some more with positioning - it will start becoming second nature - however with setup time being critical, use tools of the trade: certain line array manufacturers supply software making it easy to calculate speaker position to a given venue size; a tool to remotely monitor the angle of the line array during install certainly speeds this up - see the RAD2 line array; a laser distance finder of sorts.
* Sound path: Don’t let objects block the path of sound before it reaches the audience.
* Interior dampening/reflective material: Every space is different when it comes to acoustic properties, which will affect the sound quality. Learn as much as you can about the interplay between acoustically dampening and reflective materials and how it works. A venue with lots of reflective materials such as paneling and glass will lead to overly bright sound and can contribute to signal cancellations. Conversely it is possible to go overboard with acoustic dampening too, and muffle or drown certain sounds or voices.

Monday, September 12, 2005

What is a Line Array?

What is a Line Array?

A line array is a group of radiating elements arrayed in a straight line, closely spaced and operating with equal amplitude and in phase. Described by Olson in his 1957 classic text, Acoustical Engineering, line arrays are useful in applications where sound must be projected over long distances. This is because line arrays afford very directional vertical coverage and thus project sound effectively.

Again, proper alignment is the key to getting the best sound from any line array.

Wednesday, September 07, 2005

Speaker Placement

Speaker Placement Basics (101)

Most of you sound engineers are probably beyond this entry level tip but somebody out there has to start somewhere...

Each venue or club space is different acoustically. In general, you want to aim the Main Speakers at the audience. If you point them toward the stage (and microphones) it leads to feedback problems. If you point them at the ceiling it wastes power and muddies the sound. In a small setting, you might position the main speakers 5 feet off to either side, or possibly just use one speaker. In larger settings, you might set speakers on each side of the stage, anywhere from 10–30 feet away. Ideally you should raise the main speakers several feet over the level of the audience’s heads. In some settings it helps to angle the speakers slightly downward, so the sound goes directly into the audience instead of bouncing around off the ceiling and back walls. Putting speakers against a back wall or in corners increases bass response. Sometimes more bass occurs than is desired. In this case, you need to experiment, listen, and adjust the angle of the speakers.

Monitor Speakers should be close to the performer and aimed up toward his ears. These speakers are often wedge shaped. These speakers should not be too load because this leads to feedback problems and muddies the sound. It often helps to turn down the bass in the monitors too.

Avoiding Feedback
The best way to avoid feedback is by doing a thorough sound check ahead of time. Proper microphone selection and set up are essential. If howling/hum begins during a concert: turn down the master volume at once. Set the speakers further away from microphones, use better microphones and/or set them closer to singers/instruments so you don’t have to turn up the gain as much. Use mixer EQ section and graphic equalizers to reduce the problem frequencies during the sound check. If feedback is still a recurrent problem, you may need an auto feedback control unit and you will have to adjust the angle of the speakers.

The Rieker RAD2 Remote Angle Display significantly increases the efficiency of hanging speakers (line arrays), adjusting the angle, and decreasing the amount of time it takes to set up the sound system - for that precise fine tuning!

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