Monday, June 18, 2012
Auralex Unveils QuadFusor Sound Diffusor At InfoComm 2012
At the InfoComm 2012 show in Las Vegas, Auralex Acoustics introduced the QuadFusor sound diffusor, combining four of the company’s MiniFusors and arrays them in an attractive 2-foot-by-2-foot pattern.
The Class A fire-rated acoustical diffusor is suitable for commercial spaces and can be dropped into a ceiling grid or mounted to a wall or ceiling.
The QuadFusor’s surface-variable design includes a proprietary ledge and rear cavity that accommodate the insertion of a rigid substrate such as Auralex PlatFoam or acoustical fiberglass, thus improving the device’s diffusion characteristics and adding significant low-frequency trapping.
The QuadFusor can easily be installed in new or existing construction. The QuadFusor is Auralex’s latest addition to their complete line of acoustical treatment solutions, perfect for contractors and system integrators at any budgetary level.
For more information, please visit .
BSS Audio Broadens Network Interoperability With Dante
In a move to broaden network interoperability of its category-leading Soundweb London audio networking and processing platform and provide integrators with more comprehensive choices, Harman’s BSS Audio today added Audinate Dante enabled processors to a line that also includes Cirrus Logic CobraNet and AVB.
As a result, Harman Professional and BSS Audio now provide deeper technology solutions and support for legacy projects, current large format fixed installations and progressive AVB-enabled networks in the pipeline.
The BLU-806 offers the same configurable signal processing capability as the existing BLU-800 device. Both new devices offer configurable inputs and outputs, compatibility with all Soundweb London input and output card options, logic processing capability, the 256-channel Soundweb London digital audio bus and GPIO.
Each device offers up to 16 inputs and outputs, configurable in banks of four. Card options include analog mic/line inputs with Phantom Power, analog outputs, digital inputs (AES/EBU and S/PDIF), digital outputs, the Soundweb London AEC Input Card and the Soundweb London Telephone Hybrid Card.
The BLU-806 and BLU-326 devices allow 64 incoming Dante channels and 64 outgoing Dante channels.
“System designers using Soundweb London will now have the choice of AVB, CobraNet or Dante as the digital audio transport, using the Soundweb London digital audio bus to complement the application-specific transports. Most importantly, system designers are able to deploy each of these transports using a single software application,” stated Iain Gregory, Marketing Lead for Installed Sound at Harman Signal Processing.
The BLU-806 and BLU-326 along with the other members of the Soundweb London family provide the building blocks of the perfectly tailored system solution.
Extron Now Shipping HDCP-Compliant HDMI Fiber Optic Extenders
Extron Electronics is pleased to announce the immediate availability of the FOXBOX HDMI fiber optic transmitter and receiver set for long haul transmission of HDCP-compliant HDMI video, audio, and RS-232 control signals over fiber optic cabling.
Engineered for reliability and exceptional high resolution image performance, the fiber optic transmitter and receiver use Extron-exclusive all digital technology to deliver pixel-for-pixel performance with signals up to 1920x1200, including HDTV 1080p/60. Available in multimode and singlemode models, FOXBOX HDMI products include Key Minder, EDID Minder, Auto Input Memory, RS-232 control from multiple locations, internal test patterns, and real-time system monitoring.
EDID Minder manages EDID communication and Key Minder supports continuous authentication of HDCP compliance, providing enhanced and simplified integration. Compact, low profile enclosures allow for discreet installation such as behind a flat-panel display.
“AV system designers have been attracted by the multiple benefits of fiber optic signal transmission, and now they have a solution for transmitting protected HDMI content,” says Casey Hall, Vice President of Sales and Marketing for Extron. “FOXBOX HDMI fiber optic transmitters and receivers ensure perfect pixel-for-pixel accuracy over extended distances.”
FOXBOX HDMI transmitters and receivers can be used in combination with FOX Series matrix switchers to create HDCP compliant signal distribution systems up to 1000x1000 and larger, supporting enterprise installations with fast and reliable switching. In addition, they are compatible with FOX Series VGA and DVI extenders when transmitting non-HDCP content.
Units can be paired with FOX Series VGA transmitters and receivers for easy conversion between RGB and HDMI in AV systems that include a variety of analog and digital displays and sources. FOXBOX HDMI is ideal for a wide range of applications requiring long distance transmission of high resolution content with the highest quality.
Posted by Keith Clark on 06/18 at 10:26 AM
Carl Tatz Design Chooses Argosy Dual 15 For New Village East Studio
Village East, the personal mix room of renowned multi-platinum/Grammy-winning engineer/producer Bob Bullock, recently came online showcasing the world-class performance of the Carl Tatz Design PhantomFocus System monitor tuning protocol, as well as a number of performance and design elements hand-picked by Tatz, including the Argosy Dual 15 DR 800 studio workstation.
Bullock, who has engineered and mixed for artists such as Reba McEntire, George Strait, Travis Tritt, and Shania Twain, realized that with the caliber of his client work, he needed a personal studio environment which could sonically and aesthetically rival any high-end studio, and called on personal studio guru Carl Tatz to design it.
Tatz chose the Argosy Dual 15 DR 800 studio workstation to be the centerpiece as Bullock’s control surface and to compliment his acclaimed PhantomFocus System tuning.
“My workstation of choice for engineers working in the box, which may be a majority nowadays, is the Argosy Dual 15 with the 800 rack modules, “ says Tatz. “It has a minimum monitor reflection architecture and handsome silhouette making it the perfect complement to the PhantomFocus System.”
Along with the new Carl Tatz Signature Series by Auralex turnkey acoustical treatment system that emulates the design and performance of Tatz’s custom-designed control rooms, he also employed Dynaudio Professional M1 reference monitors driven by a Bryston, 4B SST2 power amplifier.
“When I design a studio, I choose partners who understand my vision and the standard for design and performance of the PhantomFocus system,” says Tatz. “I use elements which are both aesthetically pleasing and high performing.
The Argosy Dual 15 DR 800 has the slanted back and I love the way it looks – it takes the control room to a serious level in terms of design. And it provides the perfect positioning environment for the sweet spot.”
The Argosy Dual 15 DR-800 has a sleek, symmetrical design, featuring a front desk area and two 15-dgree angles for a wrap-around feel. The Dual 15 features two DR 800 rack modules, with a slanted top to defeat early reflections. The Dual 15 is designed to enhance the studio workspace by bringing essential order to all the equipment, putting all the controls within easy reach of the busy engineer. A padded armrest offers comfort for long studio sessions.
According to Tatz, all the manufacturers that he worked with on the project can share in the stunning results at Village East. Now music professionals can have a truly state-of-the-art mix room second to none at an affordable price point.
“The PhantomFocus System and all of its manufacturer partner elements made it possible to implement a world-class studio in a bedroom,” adds Tatz. “Bullock’s Village East is a tremendous showcase of what is possible nowadays. The room design is gorgeous, the monitoring is superior, and Bob and his clients can feel confident here knowing that it has the acoustics, monitoring, and vibe of a peerless mix room.”
Posted by Keith Clark on 06/18 at 10:16 AM
Saturday, June 16, 2012
Allen Products Debuts Updated Ceiling & Wall Mounts At InfoComm 2012
Allen Products rolled out several updated ceiling and wall mounts at InfoComm 2012, including the Steerables Wall Mount, Adjustable SocketMount, Adjustable U-Bracket and several MultiMount solutions.
Steerables Wall Mounts suspend and aim loudspeakers, equipped with suspension or mounting points, from walls and other vertical structures. Users can make or remake pan-and-tilt adjustments within seconds, offering unprecedented ease and precision.
Designed for use with permanently installed loudspeakers in churches, schools, gymnasiums, auditoriums, theme parks, nightclubs, and bars and restaurants, Steerables Wall Mounts feature a unique cross-arm steering mechanism providing up to 360 degrees of pan control. Their structural steel alloy assembly is coated with a baked-on black textured powder coat finish.
Steerables Wall Mounts can support loads of up to 100 pounds./45.5 kg. with a 5:1 safety factor. In addition, tilting can be achieved with Allen Products’ U-Series bracket or the Steerables Tilt Cable Kit.
Allen Products Adjustable SocketMount supports and aims loudspeakers that are equipped with pole sockets to walls and other vertical structures. Both pan and tilt rotations are provided for optimum aiming, and the support arm folds down when not in use.
The Adjustable SocketMount is designed as a temporary solution, as its easy-on, easy-off feature allows loudspeakers to be relocated to other SocketMounts and/or tripod stands, thus reducing the need for additional loudspeakers and saving floor space.
Adjustable U-Brackets mount loudspeakers quickly and reliably to walls and poles, and from under ceilings, balconies, overhangs and other structural mounting surfaces. These brackets give installers multiple options for mounting speakers to irregular or unpredictable mounting surfaces due to their unique mounting pattern.
Allen Products Steerables Wall Mount. (click to enlarge)
Adjustable U-Brackets support most of the mountable loudspeakers in the 60 lbs./22 kg. range, or less, and their open side-arm slots allow a single installer to easily mount loudspeakers in this weight range. Loudspeakers rotate within the arms of the U-Bracket kit to aim sound toward the audience. It then locks into position, with the included friction washers and compression hardware.
Allen Products also introduced several updated models of MultiMounts at this year’s show, including its MultiMount MM-120, MultiMount MM-024 and MultiMount MM-020-CM.
• The MultiMount MM-120 pan-and-tilt speaker mount is a sturdy and adjustable speaker mount that is safe and exceptionally easy to use for installing large-format loudspeakers. It accommodates up to 120 lbs. with greater than a 5:1 design factor. The speaker brackets’ mounting holes match two standard speaker mounting hole patterns, the 6.88- x 3.44-inch and the 2.75- x 5-inch, and two VESA mounting hole patterns, the 75- x 75-mm and the 100- x 100-mm.
The bracket’s pivot housing provides 65 degrees of tilt adjustment and the screw-drive thumb-wheel adjustment makes tilting speakers very stable. The pivot housing also provides a full 180 degrees of pan rotation. A clutch-style lock holds it in position. The wall bracket accommodates stud mounting with two centrally located mounting/alignment holes and a slot that allows for plumb adjustment. Four outer holes allow it to be secured to other structural surfaces.
• The MultiMount MM-024 supports loudspeakers weighing up to 60 lbs./27 kg, enabling the quick attachment of them to walls, ceilings, overhangs and other structural surfaces. Speakers can be aimed and locked in almost any direction or angle through the use of independent pan, tilt and clockwise adjustments. MultiMount-024 supports vertically oriented standard speaker mounting-hole patterns.
Tilt adjustments are made independently at the side of the speaker adapter plate. Panning adjustments are also made independently at the mounting plate and a third rotation is available between the speaker and the support arm. This rotation is especially useful in directing sound from the ceiling and under balcony applications, where mounting space is limited.
• MultiMount MM-020-CM is specially designed to suspend loudspeakers (weighing up to 60 lb./27 kg.) from overhead structures using a customer-supplied schedule 40, one-half inch NPT pipe. This allows the installer to bring the loudspeaker down from any height, positioning the speaker closer to the audience, further reducing the speaker’s size and power-rating needs. The MM-020-CM provides a very clean loudspeaker installation because signal wire and safety cables can be passed through the pipe from the ceiling to the speaker’s terminal block without being seen.
Also, this design adapts directly to standard loudspeaker mounting-hole patterns so no modifications are necessary. The oval-shaped ceiling attachment plate offers a sufficient mounting surface with four one-quarter inch mounting holes to secure to overhead structures including beams, beam clamps, etc. Pan-angle adjustments are achieved where the MultiMount and the vertical tube (pipe) connect, with down-tilt adjustments at the mount’s two-side clutch-lock points.
“We are thrilled to present an expanded line of ceiling and wall mounts at this year’s show,” says Paul Allen, president, Adaptive Technologies Group. “Allen Products provide a wide range of standard and custom mounting solutions to cater to nearly every possible configuration request. We will continue to develop new products based on our customers’ feedback.”
Adaptive Technologies Group combines the efforts of Allen Products, ATM Fly-ware and Adaptive Video Walls and Displays, and offers rigging and mounting solutions for a wide range of audio and video applications. Each brand offers its own standard and unique time-saving solutions, plus unlimited custom products for any venue or application. Based in Signal Hill, California, all overhead products and parts are made and assembled in the USA.
Adaptive Technologies Group
Posted by Keith Clark on 06/16 at 12:26 PM
Friday, June 15, 2012
Mixing Beyond Stereo: Delving Deeper Into Aspects Of Sound & Perception
The goal is to set up a mix that offers a wide stereo image, minimizes comb-filtering, and offers a quality mix regardless of listening position
As live mix engineers, the audio reinforcement systems we operate typically fall into two categories: mono or stereo.
Yes, there are the occasional opportunities to mix surround sound, and for many events, delay clusters or various fill loudspeakers are common, but for the most part it’s all about some version of mono or stereo.
While on the surface it may seem that stereo offers just a version of dual mono, there is a lot more to stereo than just two simple channels.
Stereo offers the significant advantage of allowing a perceptual horizontal source placement of the various signals. The primary challenge with mixing in stereo is avoiding situations where people on one side of the venue do not hear instruments that are panned to the other side.
To fully realize the benefits of stereo systems we need to look a bit deeper into some of the less obvious aspects of sound and human perception. Our ears not only recognize the volume and tonality of the energy that enters our ears, we also clearly perceive the direction from which the audio is radiating.
I like to think of a mix in multiple dimensions. Left to right is the X axis, near to far is the Z axis, up to down is the Y axis, plus bright to dull and loud to soft, and also, there are timing factors that can be added to further modify or perception of what is being reproduced. All of these aspects give us a diverse palette from which to add more character and clarity to the sonic experience we offer.
One of the primary challenges faced when mixing together a multitude of sources is presenting the audio such that the listener can discern each individual source, yet the sources all blend together in a cohesive and complementary manner.
Purely turning up an input signal increases audibility while simultaneously masking the audibility of other signals. It becomes quite tempting and common to constantly cycle through turning things up until the system limitations are reached. But by using a careful and a well thought-out strategy, significant improvements in clarity can be realized.
Everything Piles Up
So let’s start with what not to do. If all instruments and vocals are sent at equal volumes to both the left and right loudspeakers, everything piles up in the perceptual center.
Additionally, if both left and right loudspeakers are reproducing identical signals, there is a maximum amount of comb-filtering issues occurring everywhere except dead center. Comb-filtering is frequency dependant cancellations and summations that are most pronounced when two identical signals combine with a relative time offset between them.
To reduce audible comb-filtering, we either need to run a single mono cluster, have everyone in the room stand exactly equidistant between the left and right loudspeakers, or minimize the tonal and timing similarities between the signals reproduced by the both the left and right side of the system.
Because running a single mono PA stack leaves us with some serious coverage limitations - and only selling seats for the exact middle of the venue will put a big dent in ticket sales - focusing on sending dissimilar signals to the left and right PA sides is the way to go.
The goal is to reduce the amount of audio occurring in the perceptual center without compromising the mix on one side or the other.
Put another way, we want both the left and right sides to reproduce as different a signal as possible while still having each side well mixed and tonally balanced.
Achieving that goal can result in a minimal of buildup in the center, a wide stereo image, and a great sounding mix regardless of listener position.
The most basic way to shift the perceptual center of a source is to slightly delay one side. The advantage is that left and right tonality remains matched while shifting the center to one side or the other.
The issue is that since the actual signal being reproduced by both sides is still identical, it does not help with comb-filtering, but rather just moves the comb-filtering issues horizontally off center to one side or the other.
Utilizing two different microphone types on a single instrument, equalizing them to sound similar, and then panning them hard left and right is a very effective way to introduce differentials in the signals being reproduced. By differing mic types, I’m referring to using a dynamic mic in combination with a condenser mic or a ribbon mic.
There are several interesting things that occur when pairing mismatched mics. In addition to the inherent tonal differences, there will also be differences in volume linearity that will occur. This means that as the instrument is played, the sound will have a volume dependant “pull” to one side or the other.
Also, dynamic mics tend to have relatively heavy diaphragms made of plastic which makes them a bit slow to start and stop moving. Condenser and ribbon mics tend to have very light weight, metalized ultra-thin diaphragms that are very quick.
This “speed” factor, due to differentials in the mass of the diaphragms, introduces a different type of “pull” where the lighter weight diaphragm tends to lead and the heavier dynamic diaphragm tends to lag. Keep in mind that both mic types chosen should sound great independently to the point where either one could easily be used alone.
Mic’ing distance is also a useful tool. When mic’ing a single mono instrument with two mics, using a mic positioned very close to the instrument for the signal sent to one side of the PA and a more distant position for the mic sent to the other side is a natural way to introduce some delay to one side.
Not only is there a slight shift in the perceptual center, the added ambience of the distant mic adds a “far away” feel that tilts the perceptional angle from which the instrument radiates.
If directional mics are used, the closer mic will have more low-end than the distant mic. The phase shifts introduced by the channel EQ used to match the sound of the two mics also introduces additional differentials between the left and right signals.
Pairing a cardioid mic with a figure-8 or omnidirectional mic when dual mic’ing a sound source is yet another way to create disparities between the signals sent to the left and right PA stacks.
Room ambiance variations between the mics, as well as the non-linearities introduced by proximity effect, can nicely enhance the “stereo-ness” of the sound.
Compressors can be used to create the illusion of motion.
Busing a guitar, backing vocal, or other instrument into a stereo pair of compressors - with the thresholds and ratios at unmatched settings - will cause a volume dependant shift of the sound to one side or the other.
The compressor reaching threshold first will hold back the volume of that side, causing a perceptual swing towards the opposite side.
However, if a higher ratio is used on the compressor that hits threshold later, the sound will swing back the other way as volume is further increased.
It’s very common that both a bass mic and bass DI are utilized, yet there is no rule that both of those inputs need to be panned dead center. If you have subwoofers on an aux, send the bass DI to the subs and then try panning the DI and mic outward to free up more of the overcrowded perceptual middle ground.
The same splitting can also be applied to dual kick mics or snare top and bottom as well.
Pushing things further, several of these techniques can be combined to create multidimensional mixes. Each instrument has a panned placement in the horizontal field, a left to right relative delay distance, an ambience level, a left to right push or pull, placements that can shift with volume or tone, and numerous other possibilities.
A mix can be set up such that a guitar leans to one side or the other, and when the lead pedal is stepped on, a pair of compressors forces it to the center. Instruments can drift or shimmer between left and right while each side of the PA can sound great independently. By clearing out the typically overcrowded center sonic position, there is now a wide open space to lay in the perfectly clear lead vocal.
All of the concepts discussed here involve methods of altering the perceptual placement of instruments and vocals in the stereo field. I’ve purposely avoided discussing EQ and effects as the purpose of these techniques is neither to alter the tonal qualities of your mix nor to add effects that distract from the music.
Rather, the goal is to set up a mix that offers a wide stereo image, minimizes comb-filtering, and offers a quality mix regardless of listening position. Just simple, clean, clear natural sounds grabbed with finesse from the stage and presented to the audience in a multidimensional manner.
Dave Rat (www.daverat.com) heads up Rat Sound Systems Inc., based in Southern California, and has also been a mix engineer for more than 25 years.
Consultants, Contactors, Retailers - And Your Church Sound Project
Many churches don't see the wisdom of paying for professional help with sound system needs, too often to their own detriment
When faced with a need for either a significant improvement to an existing sound system or an entirely new sound system, the most often-heard advise is “hire a qualified consultant.” Or at least it should be.
But many churches balk at the notion of paying a fee to a professional to help with sound system needs.
The thinking: this is not money well spent because there always seems to be “someone” in the congregation confident in his own abilities to choose appropriate equipment and put it all together. Or, the local music store will provide the required expertise - why pay anyone else?
These approaches have lousy track records, wasting buckets of money and making everyone involved with the church suffer through poor sound quality for years. And years…
Full disclosure at this point: I am a long-time electro-acoustic consultant, so there might be temptation to think I’m biased in dispensing advice.
But the reality is that I truly wish I didn’t have to address this topic, because I’ve spent my career trying to help churches pick up the pieces after they’ve suffered mightily by putting their trust in totally unqualified personnel.
The bottom line is that a qualified consultant who specializes in live sound reinforcement (because this is what a church sound system is designed to do) will end up saving the church money, time and a whole lot of heartache.
Sound systems may be obtained in three basic ways:
1) All at once or piece-meal from a retail outlet (music store) or catalog vendor, usually installed by church members.
2) All at once or piece-meal from an AV contracting (“design-build”) firm that usually does at least part of the installation as well.
3) All at once with design by a qualified consultant and installation by a contracting firm that both work as colleagues in the process.
New let’s clarify these sources and what they do.
A retail supplier or catalog house that sells professional audio equipment does not design or engineer sound systems, and in most cases they’re not qualified to do either. All but a few of the largest do not employ seasoned live sound system experts. Very few, in fact, have staff with significant experience in live sound system design or operation.
Staff members are primarily part-time musicians or home studio owners who are there primarily to supplement their incomes. And musicians and home studio people are rarely qualified to provide sound reinforcement design advice, nor do they usually have a grasp of architectural accommodation, nor do they possess the necessary skills in installation and related safety issues, nor electrical systems and related issues, nor the physics of electroacoustics.
Further, they’re not familiar with the National Electrical Code (NEC) that must be adhered to, both for inspections and insurance purposes.
Usually, they can’t properly employ room and system modeling (computer-assisted prediction), and they’re not trained or invested in test and measurement skills/equipment that help truly optimize a sound system.
They’re usually not members of any pro audio and/or acoustics trade organization, and they’re also prone to dismiss many significant issues as “not that complicated.” (And if you hear someone say this, run away screaming immediately!)
Note that I am saying most - not all - fit these criteria. Anyone still bold enough to tread down this path should at least understand the right questions to ask.
A contracting firm usually specializes at installing systems, and many also offer system design services. However, be aware that this does not necessarily mean that they’re qualified to design systems.
Some have invested time and money in modeling and optimization training and equipment, and also have the necessary experience in using these tools and others to do successful design work.
But others, even with training, may be lacking the necessary design skills, whether it’s due to lack of experience and/or other factors. (Do you really want someone learning how to design when it’s your system?)
In other words, just because someone shows you a technical document on a computer screen - and goodness, it does look complicated - does not mean the document is providing any salient information that will translate into a better sound system design. Or, that the person doing the showing knows much more than you.
This is a common sales tool that is used to show off and “wow” the customer, when in reality, there might be little depth of knowledge and understanding behind the “dog and pony show.”
Further, contracting firms are increasingly focused on selling total audio-visual (AV) system packages, encompassing not just sound but also lighting and video. As a result, the emphasis is not just on sound, with staff members sometimes not likely to possess deep knowledge on the subject.
Like retail outlets, no single contracting firm can carry all lines of audio equipment, and they’re also bound by agreements with manufacturers to sell certain amounts of each brand they carry.
They’re also not prone to attend your project meetings without additional fees, and other than a complimentary site visit, they’re also not likely to budget time for programming and design development. They usually don’t make provisions for the focused, useful training of volunteer sound operators on aspects such as the “how and why” of live sound, as well as mixing, politics and so on.
Better contracting firms will have licensed engineers on staff, and they make it a practice to send staff members to seminars, workshops and other continuing educational endeavors. Further, these firms belong to respected trade organizations such as the National Systems Contractor Association (NSCA) and the Audio Engineering Society (AES).
There is one caveat to be pointed out. Some contracting firms offer true system design-build services, and they employ qualified designers, either on staff or via a business arrangement. This approach and structure has proven successful in some situations.
The bottom line is to understand what a contactor can and cannot do - again, if you simply ask the right questions.
A qualified electroacoustic consultant will offer ample experience working with and designing sound systems for churches, and often, other performance spaces. They will have attended advanced live sound related seminars and workshops, and they continue to do so because this is their specialty.
Consultants may or may not have licensed engineers on staff, but will have a firm grasp on safety issues, and are familiar with the terminology and methods employed by architects, other consultants (theater, acoustics), specialty engineers (electrical, structural, mechanical), and contractors (carpenters, electrical, HVAC, etc.).
Qualified consultants will also have worked at various times for a church or performance venue as staff sound system operators. Often they will have a background in contracting, and should have some background in music as well.
They will have membership with organizations such as NSCA and AES, as well as the ASA and USITT, and it is not uncommon for some staff to be trained as professional engineers (this is designated by a “PE” in their titles on business cards.)
They attend several trade shows each year because it provides insight into emerging technologies and new equipment, and it also allows them to attend technical papers and to sit in on technical committees. Participating in these conferences keeps the consultant “connected” and helps to prevent them from becoming either too proprietary or experimental.
To The Chase
One of the most significant issues: the consultant is employed directly by the client to provide the best possible system design at a reasonable cost. (It’s under the category of “who’s looking out for number one?”)
As a result, consultants specify equipment based upon its merits and value. Their affiliation with manufacturers is one of mutual respect and not unduly influenced by numbers. This independence is paramount.
However, many consultants are asked by manufacturers to participate in product development, or to provide valuable feedback so that minor design flaws may be corrected. (The better contractors and design build firms do this as well.) All must walk the lie between incorporating new products/technologies and undertaking unnecessary risk for every system project.
Consultants must also participate in your project meetings, and at the end of the design phase, they deliver a completely engineered audio system design in the form of a bid package, plus a list of pre-qualified contractors to bid on the project.
The specification should allow very few items that the bidding contractors are allowed to substitute. Specifying exact equipment for all but the most mundane components of a system is what the consultant is paid to provide, and the consultant should be able to clearly detail to the client why each device is required.
One other aspect of a complete system design is provision for future needs. This may take the form of recommending more mixer input channels than are initially needed, additional wire, cable and conduit (installed at the outset even if they’re not needed until later), and a digital processor that can easily be expanded in functions and input/output capabilities. This can save thousands and thousands of dollars down the line.
Finally, the design package should be complete and not require the contactor to “fill in the blanks.”
Another valuable ingredient is the bidding part of the process itself. If a church attempts to shop around for a sound system package, not only is this very likely to lead to multiples of package proposals (these are not designed systems in any way, shape or form), but prices will vary widely.
All of this leads to chaos since there is no way to qualitatively compare the multiple proposals.
Aside from confusion about what the church is really buying, this also opens the door to temptation to accept a lower quality system based solely on dollar value.
A consultant designs the appropriate system and estimates what it will cost. If this is too much money, then the consultant may be able to provide a lower cost option, and with a full explanation of the ramifications.
Note, however, that consultants must be willing to say “no” when being pressured to provide a compromised design, and if necessary, they will walk away from a client with unrealistic expectations.
Another option is to stage the system purchase and installation into several phases, allowing the client to obtain the best system over time. But note that this is still a “system” design and not a piece-meal approach.
As part of the bid package, the consultant provides a list of several pre-qualified contractors and they bid on the same system design. This results in no ambiguities, and the client can then accept the lowest (or one of the lowest) bids.
Programming is the most important first step in developing a design that provides what is needed. The consultant must also attend worship services and interview key personnel to establish what their needs and hopes are. This also helps the consultant to identify the technical capabilities of the crew.
Following programming and the various design phases (schematic design, design development), the entire sound system design must be presented in the form of contract documents, which include drawings and a written specification.
The drawing set must include the functional aspects of each part of the system, as well as detailed drawings of key components such as loudspeakers (clusters, arrays, rigging and mounting methods), rack elevations, custom panels, and the isolated ground AC (electrical) power system.
The latter is provided as a concept (because consultants are not licensed to engineer them), and then it must be approved, detailed and stamped by a licensed electrical engineer in the local where the church is being built or the system is being installed. This is also the case for any suspended devices such as loudspeakers.
Once the contracting firm is selected, the consultant serves as the primary interface on all issues related to the system, and then interfaces with the client on any aspects of note.
Yet another important part of the services provided by a consultant is an ability to measure and optimize the system after it is installed, and then to train the users in its operation. This completes the picture.
Finally, the consultant will be available to answer questions later, as they crop up, which can inevitably happen with a new sound system.
Churches that invest the time and care in finding and vetting qualified candidates for design and install services are much more likely to achieve a sound system that serves their needs, and will do so for many years to come, while also receiving a very high return on their investment.
Audio-Technica Introduces New Components For SpectraPulse Wireless System At InfoComm 2012
Audio-Technica has introduced two new components for its SpectraPulse Ultra Wideband (UWB) wireless microphone system.
The new chg004 is a four-bay charger for SpectraPulse transmitters, while the new sei001 encryption interface is now included with the optional encryption software available for the SpectraPulse system.
The new chg004 offers four charging bays that hold either mtu101 microphone transmitters or mtu201 XLR desk stand transmitters; the unit is also designed to charge up to two mtu301 bodypack transmitters.
Supplied rechargeable AA Nickel Metal Hydride (NiMH) batteries are charged within the SpectraPulse transmitters. The chg004 offers built-in safety features that monitor cell voltage and automatically stop charging if problems are detected or if alkaline (non-rechargeable) or damaged batteries are installed.
Maintenance charging prevents battery self-discharge until the transmitter is removed from its charger. The package includes the chg004 charger, 12 NiMH AA batteries, two charging cables (for charging mtu301 units), a wall adapter power supply and an installation/operation manual.
SpectraPulse systems offer exceptional immunity to eavesdropping, and are available with optional sep128 encryption software that meets the AES 128-bit encryption standard developed by the U.S. government for securing sensitive material.
The sep128 software now comes equipped with the sei001 encryption interface, a convenient one-bay encryption station for SpectraPulse transmitters.
The first commercial sound implementation of Ultra Wideband technology, Audio-Technica’s SpectraPulse Ultra Wideband wireless microphone systems offer secure wireless operation for the installed sound market, free from RF competition, frequency coordination and “white space” issues.
These new components are part of Audio-Technica’s contractor-exclusive Engineered Sound line and will be available July 2012.
Posted by Keith Clark on 06/15 at 08:28 AM
Recording The Rolling Stones “Brown Sugar” Sessions
Reconstructing the night that gave us a rock song for the ages
If you’ve seen the Rolling Stones Gimme Shelter movie, you might recall Jimmy Johnson’s brief speaking role.
He was the one coaching Keith Richards on the proper Alabama pronunciation of “Y’all come back, y’hear.”
For three nights in December of 1969, the Stones cut basic tracks and live vocals for three songs: “You Gotta Move,” “Wild Horses” and “Brown Sugar.”
The sessions took place at Muscle Shoals Sound Studios-the “burlap palace” at 3614 Jackson Highway—a nondescript former casket factory which the four rhythm section members had purchased earlier that same year.
Prior to venturing out on their own, the foursome (Johnson, bassist David Hood, keyboardist Barry Beckett and drummer Roger Hawkins) had been the core players at Rick Hall’s Fame Studios, where their rhythm tracks laid the foundation for soul hits by Aretha Franklin, Wilson Pickett, Etta James, Arthur Conley and others.
Since early in his Fame days, Jimmy Johnson had switched roles back and forth, playing fatback rhythm guitar on some sessions, engineering others. His early engineering credits included “Sweet Soul Music” and “When a Man Loves a Woman.”
But when the Rolling Stones arrived-with little advance notice-Johnson was confronted with something quite other than the relatively low-volume, laid-back soul and pop sessions that were his usual fare.
On one hand, you could say the fledgling Muscle Shoals studio was ill-equipped for the task. On the other hand, you might say maybe this turned out to be a good thing. Let recording history be the judge.
In this interview, Johnson reconstructs (as best can be expected after 35-plus years) the night that gave us a rock song for the ages.
Let’s try to set the scene for those sessions, starting with the console you used.
When we did the Stones sessions, we had a Universal Audio console with tube modules, the one with the big rotary knobs, knobs as big as your hand. We had ten inputs.
There was some fixed EQ on it, a fixed low end at 100 Hz, and you could go two clicks of boost at two and four dB, and you could roll back to minus three.
But that’s all it was. It also had an echo send on it. Back then, we were using a live chamber. It wasn’t until a year after that we got an EMT plate. Of course, we were uptown then!
Did you get the Universal modules new, or from another studio?
We bought all the modules new, and put it in our own little console frame. We had a cabinetmaker build us a console, the same as Rick did over at Fame, this was the same thing he had.
At the time, it was one of the best things you could get, depending on your budget, of course. Our budget wasn’t too big at the time.
And what kind of tape machines did you have?
We had a Scully eight track, a one-inch, and it was great. We had no noise reduction, though back then we cut a lot of stuff at 15ips. We just packed a lot of it on!
And the tape was very forgiving, so as a result it turned out well. We got a lot of saturation, and that kind of became part of the sound.
Back then…I don’t even remember any noise reduction at the time. I know there wasn’t any when we went up to Atlantic in ‘66. But there might have been some around that I didn’t know about.
Were you the only engineer on hand for the Stones sessions?
Yes, I did all those myself, along with my assistant, Larry Hamby. It was supposed to be Jimmy Miller, from what I understand, but he didn’t show up.
It was my intention to assist him when the whole thing started, because I heard they would be bringing their own people. As it was, he never made it down. So I became the unofficial-official engineer for all those sessions.
Did you cut all the basic tracks here?
They did some overdubbing later, of backgrounds, saxophone and acoustic guitar. But electric guitars, lead vocals, piano and even the percussion was done right there, Jagger did that. Mick Taylor was on those sessions, of course, and during “Wild Horses” Jim Dickinson showed up, from Memphis.
What happened is that their touring piano player, who was also their road manger, Ian Stewart, he played on “Brown Sugar” some, but during “Wild Horses” Jim Dickenson was out behind the where we put the guitar amps “Do you remember Paul Simon’s ‘Kodachrome’ where we went to double time and the tack piano comes in, the piano kind of goes crazy?
That was our tack piano, an old upright piano; we put tacks on the hammers so it sounded like a honky tonk. Anyway, Jim was back there just tiddling on it, playing along with what they had settled on as the groove, and Keith walked by and said, “Hey you need to play that!”
Let’s try to reconstruct how “Brown Sugar” was tracked. First, what mics did you have set up, starting with the drums?
We only had three mics on the drums. We ran a U47 up over the top up over the top, about nose high to the drummer. We had a high stand out in front, with the mic facing downward at the kit, from the bass drum in with a little boom that came over the snare.
So it gave a good overview of the whole kit, so you could play with a lot of dynamics and you could get an incredible sound. In fact, Charlie Watts wanted to buy that microphone! But of course, I wouldn’t sell it. He couldn’t get over the sound we were getting.
On the bass drum we used the E-V 666, a fantastic dynamic mic for the time. It was on a little stand looking to the backside of the drum. Then I had a hi-hat mic, which I think was another (E-V) RE-15, though it could have been a little (E-V) 635A, that remains in question.
The RE-15 was a better mic, had more response. We avoided using the 635A unless we had to. Actually, if not the RE-15 it might have been an SM57, more likely than the 635A.
And Charlie brought all his own drums?
Yes, he brought all of this own kit.
What guitar was Keith playing?
It was a Gibson, but not a Les Paul. Do you know that model that was right under the Les Paul, the solid body double cutaway-what is that?
Oh yeah, the SG. I think it was an SG, and as I recall it was black. I remember it had those sharp horns on the cutaways. That’s what he played most of the time he was here.
And Mick Taylor?
Taylor, to my recollection, was playing a Strat. And guess what we came up with for Bill Wyman? Do you remember those Plexiglas body basses that were around then?
I checked with David Hood later and he says it was a Dan Armstrong. So to the best of our recollection, that’s what it was. He played through David’s Fender Bassman setup, the tube head and separate box.
And the guitar amps?
Keith played a Fender Twin, and so did Mick Taylor, and they brought those in with them. The loudness on those tracks really came from Keith. I had it put in that back booth and shut the door on it.
So Mick’s was out in the room?
Yeah, it was out, set where I normally played. If you looked from the control room it was on the left side, about the middle, facing toward the front. You see, we had all these wonderful baffles, covered with burlap, with that pink insulation underneath.
We would corner off the sound with a couple of baffles up against each other. It would just knock the directness off, it took a lot of top end off.
So you could balance it out, but not stop leaking altogether.
Exactly, you couldn’t really snuff the sound out. It wasn’t as evident in the other mics, but it was there.
How did you mic the guitars?
On the guitar amplifiers, let’s see there were two different ones, on Mick’s I had a Shure SM-57, and then on the other I was using.. I might have been using an RE-15 on Keith. But I had a real problem with Keith because he was running a Fender Twin amp wide open, I mean that sucker was getting it.
I had a real problem with distortion going on, but I happened to remember that my maintenance guy, about a month before that, had left me a 20dB pad that he had made, a homemade pad, so I just stuck it in between. So I dropped that level before it hit the front of the Universal Audio and it saved the day.
Otherwise, I would have been hosed. I still thank God for that. I would have just been screwed. So on Keith’s amp, ‘oh no, I remember what was on his amp, an RCA 77DX, because I was having to get that level down any way I could, it was a ribbon mic.
With the pad and that RCA, I made it, just barely. A lot of that had to do with how it sounded, and I was always real pleased with that guitar sound.
I assume you close-miked the amps.
Yes, they were miked about two or three inches from the grille cloth, and with the Twins, we would get right in front of one of the two speakers. I’d make sure that both were working all right, and that one didn’t have a hole.
How about the piano mics?
On the piano I was using only one mic, not two, so I had to move it around to find the hot spot. I’m going to have to think on that one.
I think it was a U47, that was the other one, because three was all we had. And we used them all on every session. Jagger sang on a U47.
So the U47 on Jagger, that was a live vocal track? Or was it overdubbed?
I don’t think so, not unless he had to fix something in London. The only overdub I remember was the percussion that he did. He had mono earphones of course, and they were hearing what the board was hearing, they couldn’t get a separate mix.
Did you have a mic on the bass amp?
Yes, the bass guitar mic was an RCA 44. We didn’t have direct back in those days.
How much separation could you get in that studio?
Well, Keith’s guitar amp was in a booth, and Jagger was in the back of the room with baffles around him. There was some leakage going on, but you couldn’t tell because he was so close to the mic.
It was part of the sound. The drums did not have a booth, they were open, but with baffles. But there was a lot of leakage on the drums, cymbals and stuff, even though he didn’t play real hard.
Really? But there’s a lot of impact in the drums on that song, more than on most Stones tunes.
Yeah, it’s that mic and the way we set it. Even today, that would be a good way for a rock band to mic their drums, if they like some great live drumming sound. They would be surprised to find that sometimes less is more. I think it would blow them away.
And the sound of Keith’s guitar is so good, and I really attribute it to that RCA DX77 with the pad, going into that Universal Audio tube console which warmed it up, too. Pretty wild, huh.
Did you use any compression on those tracks?
None. At the time, I did not have a compressor in the building. It was a couple more years before we had compressors. The only outboard gear was that 20dB pad, that’s all I remember.
What about board EQ? Did you use much of that?
Mostly, on all sessions, I would use one click or two on the highs to air it out. It was set at around 3 or 4K, with two dB steps; you could go to two or four. We had 100Hz for the low end, and I guess around 3500 for the high.
I remember Barry Beckett saying he was sitting outside on the steps and could feel the building shaking.
Yeah, when they stated grooving around one in the morning, when I started the machines, it was an unbelievable thing; I have not experienced anything quite like that since.
If you compare Brown Sugar to other cuts done by Glyn Johns at about same time, most came out on Let It Bleed, you don’t get that kind of room sound. They have that clean separation, you don’t get the feeling of the whole room being pumped up by the music.
Right. You get the same thing from those old Motown records, cut at Hitsville USA in Detroit. When they moved to LA it all changed, they never had that wonderful sound again. I don’t understand how they could divorce it that way. But I love that sound, and the old Philly Studio, Sigma Sound and all the great records done by Gamble and Huff, God they were great!
Yeah, but up in Philly, they were probably saying, “How can we get that funky sound they have down there in Muscle Shoals?”
Oh, we didn’t even think about that. I suppose the grass is always greener somewhere else.
Straightforward Approach To Setting An Optimized Audio System Gain Structure
Over the years I’ve seen many haphazard approaches in establishing gain structure through a sound reinforcement system.
Often rough adjustments can be made to make the problem less apparent, because gain is easily and cheaply available in today’s industry.
Years ago, when a 100-watt power amplifier was used to power the main loudspeaker system, gain structure was a critical issue. Today, with the advent of amplifiers that can output levels of 1,000 watts or more per channel, proper gain structure can be easily overlooked as a critical element in the performance of a system.
Powerful amplifiers, however, are not an excuse for an individual to lack a firm understanding of proper gain structure. Many of today’s signal processors and amplifiers have jumpers, switches, or knobs that, if adjusted properly, will maximize the systems signal to noise ratio while also ensuring the system will safely operate at the levels that are required.
First we’ll determine how much gain is required throughout the system, from the console to the listener. Then, once overall gain requirements are known, we can discuss the approach to setting the system’s gain structure.
How much gain is enough?
A good designer will always have an established sound pressure level (SPL) criteria for each system in which he/she is working on. Without this pre-determined resultant SPL, you may often find yourself over or under specifying the total gain required in a system.
So, let’s establish a criteria for the purpose of discussion. We will assume that we are designing a sound system for a church that has a contemporary music program. During the music portion of the program, it is anticipated that peak levels in the room need to reach nominal levels of 95 dB SPL, with peaks of 101 dB. It is also the intent to provide 10 dB of headroom. Our design criteria is now determined, and we can begin our discussion of gain structure.
The first thing to determine in the system is the amount of loss due to distance. For the purpose of our example, let’s assume that the furthest distance a listener will be from the speaker is 80 feet. Because sound radiates spherically, the attenuation is proportional to the square of the distance from the source, and thus there is a 6 dB reduction for each doubling of distance.
Assuming that the sensitivity of the loudspeaker is given in reference to 1 meter, this can be represented mathematically by the equation:
(Equation 1) SPLdist-loss = 20 log (distance in feet / 3.3)
The 3.3 factor is used to convert feet to meters. Using this equation, we determine there will be a total loss of 28 dB as a result of distance. We can now calculate the maximum output level of the loudspeaker that we will require in order to achieve our design criteria.
We have already determined that we need a maximum SPL level of 111dB at the listener position (101 dB peaks with 10 dB of headroom). At the loudspeaker we will need a maximum SPL level of 139 dB (111 dB at the listener position + 28 dB of loss due to distance.)
The selection of the loudspeaker is the next step in the process. Any loudspeaker that is specified will have a sensitivity and a maximum power rating. The sensitivity is normally given in dB SPL at 1 meter when a 1-watt signal is applied to the input of the loudspeaker, and is usually given in AES watts.
This AES measurement is a clearly defined standard in which a band of pink noise from 125Hz to 8Khz, with +6 dB peaks, is applied to the input of the loudspeaker for a period of two hours. Any loudspeaker that has its power rating in AES watts can very easily handle short-term peaks of +6 dB above the AES rating.
The maximum output level at 1 meter away from a loudspeaker will be derived from the formula:
(Equation 2) SPLmax-AES = sensitivity + 10 log (AES power rating)
(Equation 3) SPLmax = sensitivity + 10 log (AES power rating) + 6
Any loudspeaker we select must have an SPLmax of at least 139 dB. We will take one particular manufacturer’s loudspeaker that has a sensitivity of 112 dB at 1 watt/1 meter. The high frequency component can handle 200 watts, AES. Using the equation above, we find that the SPLmax equals 141 dB. This loudspeaker will have the ability of achieving our design criteria.
To complete the design, we must choose the correct amplifier size for the application. Amplifier power ratings are given in watts, but unlike loudspeaker AES power ratings, amplifier power ratings are the upper limits and do not include any crest factors.
For the purpose of discussion, let’s assume that we have a 3-way loudspeaker system (loudspeakers with high, mid and low sub-sections), with the following AES power ratings and sensitivity ratings:
Loudspeaker Sensitivity & Power Ratings
AES power rating————————200————————-400———————-1000
(Using Equation 3)
The high and mid sub-sections of a single loudspeaker can handle the minimum SPL requirements of 139 dB at 1 meter. However, the low frequency sub-section will require two loudspeakers.
And then, by doubling the number of loudspeakers, we will obtain a +6 dB gain, which results in a low frequency peak SPL of 143 dB. We can now go directly to our amplifier selection. In order to calculate the amount of power required, we need to use the following equation:
(Equation 4a) PWR(dB) = SPL Criteria peak - sensitivity + SPLdist-loss
(Equation 4b) PWR (watts) = 10 PWR(dB)/10
The peak SPL criteria was established earlier at 111 dB SPL (96 dB nominal + 6 dB peaks + 10 dB headroom). The loss due to distance is 28 dB. By plugging these numbers into equations 4a and 4b (above), we obtain the following results:
Amplifier Power Requirements
Calculated Minimum Power———27 dB——————30 dB———————-32 dB
Power in watts**——————-500 watts—————1000 watts—————1585 watts
* One loudspeaker will be required to provide an SPL criteria peak of 105dB SPL since two loudspeakers will give us our required SPL criteria peak of 111 dB SPL.
** This is peak power, not AES. The AES power handling would -6 dB lower than this (divide by 4).
Now that the amplifier size has been determined, the next thing to look at is processing level inputs and outputs.
Most sound consoles can comfortably handle an output level between +18 dBu and +24 dBu.
This, in turn, will feed the processing equipment. Analog processors can usually handle +18 dBu input and output signals. This is the first place in line where attenuation or a pad may be required.
If you are using a console that can output +24 dBu, you will want 6 dB of attenuation at the input of the audio processor. This can usually be achieved by the input attenuators on the signal processor.
The outputs of the signal processors require a bit more discussion. Many DSP devices have either output switch settings or output jumper settings that can select between 0, +6 dB, or +12 dB, so the obvious questions are “Why are there different options?” and “When do you use them?”
To answer this, we must first continue our discussion about amplifiers.
AMPLIFIER INPUT LEVELS
Many manufacturers have input selection settings than can choose between 0.775V, 1.4V, X20 (or 26 dB), or X40 (or 32 dB). For the purpose of discussion, the table below indicates the input level that 200-, 400-, and 800-watt amplifiers will accept before the amp clips.
The 0.775V and the 1.4V input level settings indicate that all amplifiers will clip at the same input level. For the X20 (26dB) or the X40 (32dB) selection settings, the size of the amplifier and the load on the amplifier will determine the level at which the amp will clip. It is very important to be able to understand the clip levels and gains of the amplifiers in both dB and in voltage.
For 0.775V or 1.4V input sensitivity
(Equation 5a) Gain (volts) = sqrt [Max power rating * load (ohms)] / input sensitivity
(Equation 5b) (dB) = 20 log[Gain (voltage)]
(Equation 5c) Clip Level (volts) = input sensitivity (0.775V or 1.4V)
(Equation 5d) Clip Level (dB) = 20 log [clip level (volts)]
For X20 (26 dB) or X40 (32 dB) gain
(Equation 6a) Clip level (volts) = sqrt [Max power rating * load (ohms)] / gain (20 or 40)
(Equation 6b) Clip level (dB) = 20 log[Clip level (volts) / 0.775V]
(Equation 6c) Gain (volts) = gain (20 or 40)
(Equation 6d) Gain (dB) = 20 log[gain(volts)]
Amplifier Input Clip Levels
———————-200 Watts————————400 Watts————————800 Watts
X20 (26 dB)———-8.2 dB———————————11.2 dB—————————-14.2 dB
X40 (32 dB)———-2.2 dB———————————5.2 dB——————————-8.2 dB
0.75V——————0 dB————————————-0 dB——————————-0 dB
1.4V——————-+5 dB———————————-+5 dB——————————+5 dB
Now that we have thrown all of these numbers out there for you to ponder over, we now need to know when we would want to use these different input settings.
The primary factor in determining which settings to use is determined by the designer’s requirement for the system’s noise floor.
If noise floor is not absolutely critical (NC-25 or higher spaces), then the amplifiers can safely be set on 0.775 (or preferably 1.4V if available).
Because the actual gain of the amplifiers is quite high (~X40 for a 100 watt amp to ~X130 for a 2500 watt amp), the noise floor will be higher. The clear advantage, however, is that you do not need to calculate the attenuation needed for every channel of every amplifier.
If noise levels are a critical concern, then constant gain settings should be used, but you will need to calculate the attenuation for each amp channel.
To conclude our discussion on signal processing and the output level switches on DSP devices, if you are using an amplifier that has its input sensitivity set on 0.775V, then the output of the DSP should be set at 0 dB. This will provide 18 dB of attenuation between the console and the amplifiers.
If the input sensitivity of the amps are set at 1.4V, then the output of the DSP should be set at 6 dB. If you are using constant gain, then each output needs to be addressed on an individual basis.
One final note on gain structure worth mentioning is to always have a good sense for what is occurring with the system equalization. Let’s assume that there is a large +10 dB boost in the EQ at 8K.
During system tuning it may make the speakers sound very well and provide extended high end frequency response, but 8K signals will clip the amplifiers 10 dB sooner than the rest of the system.
Similar problems may arise from very large EQ cuts, but if at all possible, for gain structure purposes, it is better to cut than to boost, and it will always be best to keep your cuts and boosts to an absolute minimum.
Brian Elwell is senior consultant with Acoustic Dimensions and has contributed to system designs at major stadiums, houses of worship, theme parks and many other venues.
Thursday, June 14, 2012
Aphex Introduces New Audio Xciter App
Aphex (Booth C11142) is proud to announce the Audio Xciter App, the first audio app that dynamically enhances and improves the quality of digital music (and audio books, podcasts etc.) on smartphones and other personal computing devices.
It has been a goal of Aphex Chairman/CEO David Wiener to help musicians and music lovers make the most of every music experience. Audio Xciter is a testament of Wiener’s drive to end the debate about poor audio quality in digital devices.
Designed for phones and tablets that use Apple iOS and Android operating systems, Audio Xciter uses patented Aphex technology to improve the quality of digital music by analyzing and enhancing the audio signal in real time by applying the same kind of processing power developed by Aphex.
“Heavily compressed digital music file formats – MP3, for example – have taken a lot of sonic performance out of music recordings,” stated David Wiener, “So Aphex has taken its professional studio technology and created this app in order to restore the detail, richness and spaciousness to your audio experience.
“The result is a more immersive sound that more accurately reflects the original intentions of the artists, producers and engineers.”
The Audio Xciter App analyzes a music file while it’s playing and in real time restores missing harmonics lost through data compression and production inefficiencies.
The sonic benefits of the Aphex Audio Xciter App apply to any kind of playback monitoring system or device, from headphones and earbuds to home and auto speaker systems and device docks – all will deliver dramatically improved audio quality via Aphex’s Audio Xciter app.
Aphex will be giving free copies of the app to industry attendees and media who stop by the Aphex booth at InfoComm. Just come by and ask for a free copy of the app, and Aphex will arrange for a gift app to be emailed directly to you.
In the recording studio, leading artists once paid as much as $30 per minute to license the Aphex Exciter audio enhancement system for mixing and mastering.
Now, music lovers will be able to try out a fully functional demo version of Audio Xciter as a FREE download that gives them a 15-minute per day test drive. Once users have tried the App and choose to purchase and download, there are two options: basic and Studio versions.
The Audio Xciter basic version ($4.99) allows a fully automated version of the Aphex technology to be engaged on their device, giving three options for settings. With Audio Xciter Studio ($4.99 in-app upgrade), users will have access to the exact same array of parameter controls found on Aphex’s legendary pro audio Exciter rack products, allowing them to adjust the App’s settings to customize the enhancement details and sound to their own tastes.
Both versions of the Audio Xciter App will be available through the Apple iTunes App Store for iOS users, and the Android Store and Amazon.com for Android.
Audio Xciter automatically plays all your existing music and playlists. It does not copy your audio files – preserving your valuable memory – and the processing is done in real time. The Aphex process uses a patented method of dynamically enhancing the harmonics and low frequencies present in audio that get truncated by bit-rate reduced formats such as MP3.
Audio Xciter represents the first time that the Aphex technology has been brought from the recording or broadcast environment and made available for personal mobile devices.
The Aphex Audio Xciter App’s ability to let you hear instruments and vocal details you never realized were there is summed up succinctly by GRAMMY Award-winning record producer and engineer Alan Parsons, known for his work on classic Beatles and Pink Floyd recordings, among others: “The Audio Xciter will transform your listening!”
The Aphex Audio Xciter will indeed transform your listening experience by revealing nuances in music files that have been obscured by the digital process. The Aphex Audio Xciter brings you closer to the music you know and will help you discover new music and artists by making the experience of listening to digital music files much more rewarding.
RFvenue Debuts New Antenna Products At InfoComm 2012
Antenna and wireless products manufacturer RFvenue (InfoComm Booth #C11542) unveiled three new products on Wednesday, marking the one-year anniversary of its first antenna product, the Diversity Fin.
CEO Chris Regan announced two new extensions of the popular dual element antenna system, including an EU version covering 470-790 MHz and also matte white and matte black architectural versions.
RFvenue also showed its new four-channel antenna distribution system which can interface with any brand wireless microphone receiver, and provide DC power distribution functions from a high-quality internal power supply, eliminating extra power modules.
Other additions to the RFvenue product line include high-quality, American-made, mounting equipment, and a 2.4 GHz long-range helical antenna for extended range performance for new digital wireless microphones and production communication systems.
Also shown at RFvenue’s booth is its newly designed UHF ceiling antenna for wireless microphones, intercoms, and IEM systems which mounts easily to any ceiling tile to optimize system performance in conference rooms, classrooms, or lecture halls. The ceiling antenna, dubbed the CX22 permits efficient low-noise passive receive and transmit coverage of 440-888 MHz over conventional plenum rated cables.
Posted by Keith Clark on 06/14 at 10:12 AM
JBL Professional Introduces AWC82 And AWC129 Compact All-Weather Loudspeakers At InfoComm 2012
At InfoComm 2012, JBL Professional introduces the AWC Series of compact, all-weather loudspeaker systems, bringing sound quality and high sound level capability to today’s outdoor applications.
The wide-range, smooth frequency response and high sensitivity ensure high-fidelity music reproduction and superb projection of clear, intelligible speech.
The AWC82 is a very compact, 8-inch, 2-way coaxial speaker with compression-driver high frequency, wide 120 x 120-degree coverage and 250 Watts power handling.
The AWC129 is a 12-inch, 2-way coaxial speaker, also with compression-driver high frequency, focused 90 x 90-degree coverage, and 400 Watts of power handling.
Both speakers can be operated as direct low-impedance (8 ohm) or on a 70V or 100V distributed line via built-in 200 Watt low-saturation, multi-tap transformers.
Components feature Kevlar reinforced low-frequency drivers and high-frequency compression drivers with a unique patented design and high temperature polymer diaphragm for high output levels with low distortion.
Enclosures come in light gray or black and are paintable to match the requirements of the application.
AWC components are highly weather-treated and a 3-layer grille with vapor-barrier backing minimizes direct rain contact with the system transducers. The recessed terminal compartment is protected by a panel cover and a water-tight gland nut fitting.
A U-bracket is included. The system’s IP-56 outdoor rating and wide, smooth frequency response makes JBL’s AWC speakers ideal for projects such as racetracks, school stadiums, theme parks, fairgrounds, skating rinks and swimming facilities, as well as for arenas, general public address and a variety of other indoor and outdoor applications.
“The high power 8-inch and 12-inch low-frequency drivers, derived from JBL’s Control 328 and Control 322 high power, large format in-ceiling systems, combined with a new generation of compression driver, utilize the internally contoured pole piece and the entire driver cone to form a large diameter waveguide for the high frequencies,” said Rick Kamlet, Senior Marketing Manager, Commercial Installed Sound, JBL Professional. “Those factors, along with high-slope crossovers, provide exceptionally consistent pattern control from a very compact system. These coaxial drivers allow these speakers to be extremely compact while really packing a punch in both sound level and audio quality.”
Posted by Keith Clark on 06/14 at 09:55 AM
PreSonus Adds Powerful New Free Features To StudioLive Mixers
Integrated Smaart frequency analysis, new equalization and talkback features, and even more extensive iPad control are included in PreSonus’ free Universal Control 1.6 and StudioLive Remote 1.3 for iPad updates.
“All current owners of our StudioLive 24.4.2, 16.4.2 and 16.02 can immediately use these enhancements,” noted PreSonus Executive Vice President of Product Development John Bastianelli. “It’s further proof of our commitment to providing complete problem-solving systems, not just hunks of hardware.”
With Universal Control 1.6, Rational Acoustics’ Smaart real-time analysis features have been integrated into Virtual StudioLive (VSL), which is part of Universal Control. Virtual StudioLive now offers both Smaart Spectra time-frequency spectrograph and real-time analyzer (RTA) views, which are superimposed over each graphic EQ.
The Smaart Spectra spectrograph display can help to precisely identify nasty feedback frequencies, enabling even less-experienced users to easily tune P.A. speakers to the room. Smaart Spectra graphs a continuous series of spectrum measurements, showing frequency on one axis, time on another, and amplitude indicated by colors. This makes it very useful for quickly identifying feedback frequencies, which can be easily addressed using StudioLive graphic EQs.
Universal Control 1.6 adds other useful features, as well. You can now control Talkback on/off and routing with VSL. Version 1.6 also adds support for the new StudioLive Remote 1.3 update (discussed shortly).
Finally, Universal Control 1.6 updates the Windows and Mac drivers for all and StudioLive-series mixers products and for all FireStudio-series interfaces except for the original FireStudio (26x26). This update greatly improves performance, especially on Windows systems. Note that Universal Control 1.6 requires a firmware update to StudioLive mixers.
PreSonus has simultaneously released StudioLive Remote 1.3 for iPad. This version of the company’s wireless mixer-control app adds Talkback on/off and routing assignments and enables Scene recall for hard-disk-based Scenes (but not for Scenes held in mixer memory).
Universal Control 1.6 is a free update and can be downloaded from the Technical Support section of the PreSonus Web site. StudioLive Remote 1.3 is a free download from the Apple App Store.
Posted by Keith Clark on 06/14 at 09:36 AM
Wednesday, June 13, 2012
Yamaha Launches First Models In New Commercial Install Series
Yamaha Commercial Audio Systems, Inc. has announced the launch of the new CIS Series at InfoComm 2012.
The Yamaha CIS (Commercial Installed Sound) product offerings have been created specifically for installed sound market applications where there may not be an experienced audio operator running the system; for example, restaurants, retail outlets, public address systems within the transportation industry, convention centers, hotel ballrooms.
The first Yamaha CIS product offerings consist of matrix processors, multi-channel amplifiers, ceiling speakers, and surface mount speakers providing renowned Yamaha digital technology, quality, and reliability for cost-sensitive projects within the commercial installation market.
“The Yamaha CIS products have been specifically designed with ease of installation and ease of use in mind,” states Marc Lopez, Marketing Manager, Yamaha Commercial Audio Systems, Inc. “And, by using Dante technology and our YDIF cascade bus, customers will be able to conveniently share audio and connect with all Yamaha CIS products as well as other professional audio products on the network.”
The Yamaha YDIF Cascade Bus provides a simple intra-rack connection utilizing inexpensive Cat5 cabling for an easy to use 16-channel bus, while Dante technology will allow CIS systems to share audio with up to 40 subnets (using the Audinate Netspander product) enabling Yamaha CIS systems to merge with other complex professional audio systems simply and efficiently.
There are two models in the programmable DSP matrix processors. The MTX3 is an 8x8 analog unit with a YDIF cascade bus and SD Playback for a 26 x 8 matrix. The MTX5-D is also an 8x8 analog unit with YDIF cascade bus, 16 channels of built-in Dante, and includes SD Playback and an MY card slot for a 36x16 matrix.
Both units feature PEQ, Dynamics, Feedback Suppression, Auto Gain Control, and Priority Ducker. Outputs feature Delay, PEQ, and Speaker Processing. Optional YDIF I/O units can expand analog inputs and outputs. Control and programming software will be available for both units. Optional wall mount controllers are available as well as an iPhone App for remote control.
The MTX3 and MTX5-D will be available at a target MSRP of $1,599 SRP and $2,799, respectively.
Four multi-channel amplifiers have been designed for the Yamaha CIS product line: XMV4280, XMV4140, XMV4280-D, XMV4140-D.
The four-channel amps are rated at 140W and 280W and are available in either YDIF or Dante versions. The four channels can be used in bridged mode, are high impedance and selectable in pairs, and can be controlled remotely by the MTX software.
Target MSRPs: $1,299 for the XMV4140, $1,999 for the XMV4280, $1,899 for the XMV4140-D, and $2,599 for the XMV4280-D amplifier.
The new Yamaha VXS range of surface mount speakers consists of four models: the 5” VXS5, 8” VXS8, VXS10S 10” sub, and VXS10ST 10” sub with transformer—all weather resistant (IP55 rating).
The speakers contain built-in transformers and include a quick mounting bracket for easy installation. Available in both black and white, the Yamaha CIS surface mount speakers will carry a targeted MSRP of. $179 for the VXS5, $299 for the VXS8, $399 for the VXS10S sub, and $499 for the VXS10ST sub with transformer.
The new VXC ceiling speakers are available in 4”, 6” and 8” models. The Yamaha VXC speakers are easy to install with Patent-Pending anti-drop tabs for temporary installation and secondary safety, as well as double threaded screws for faster installation. Transformers and back can are included.
The ceiling speakers have a target MSRP of $146, $189, and $269, respectively.
Selected units in the Yamaha CIS Series have a targeted availability date beginning fall 2012.
Yamaha Commercial Audio Systems