Monday, June 18, 2012
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
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
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
Zeehi’s CueCast Technology Allows File Conversion Between Mixing Console Platforms
Zeehi, an entertainment technology company dedicated to developing solutions to improve the workflow of global entertainment production professionals, has announced the Beta Release of the CueCast Digital Mixing Console User File Conversion Service.
CueCast provides front-of-house and monitor engineers, sound reinforcement companies, broadcasters, performing arts facilities and production managers the ability to quickly and easily convert show files between different digital audio mixing console formats.
The current Beta Release of CueCast converts the most commonly used features and functions including bussing, sub-group assigns, control group assigns, routing, labeling, mutes and mute groups, EQ and dynamic in/out settings, aux. send on/off/and assigns, and effects and matrix on/off and assigns.
Future releases will support additional console platforms, and will provide conversion of variable settings, snapshots, and many other features.
The web-based service solves a fundamental challenge: how to take user settings from one mixing console to another without the time-consuming headache of entering those settings manually.
“Thanks to the console expertise and software savvy of our development team, converting the user data is reliable and takes just three easy steps,” notes Danny Abelson, co-founder of Zeehi. “Simply upload your file to our secure site – http://www.cuecast.com – specify the format you need, and download the converted file for installation in the new console.
“If you like, we’ll store your files on our secure cloud for safekeeping. CueCast technology supports files from 24 to 196 inputs, and the Beta release supports file conversion between the Avid Venue, DiGiCo SD8 and SD10 and Yamaha PM5D platforms. Soon you can expect us to add many other formats and features to the service.”
“We are very fortunate in our industry,” continued Abelson. “The manufacturers have designed mixing consoles with remarkable features and sound quality.
“Unfortunately the formats in which user data is stored in these different platforms are not compatible with one another. Our research with some of the world’s most prestigious sound companies, veteran sound engineers, broadcasters, and leading production managers indicates that this incompatibility results in a significant burden to users.
“For a variety of reasons engineers often need to transfer those settings to a different brand or model of mixer depending on touring conditions, a venue’s installed equipment, or the availability of a particular console.
“These users identified a number of different scenarios, including festivals, touring without a console, change of production truck, unexpected equipment loss or failure, or availability where switching consoles is simply a necessity.”
“The opportunities digital has given us have been remarkable, but with these improvements come greater expectations by performers and management for crews to do more in less time.
“On a surprising number of occasions, digital solutions unintentionally create obstacles for the production team. Our goal is to improve workflow for entertainment industry professionals by streamlining this process, and make changing to another desk fast and easy.”
Insights Learned From Doing Monitor Sound For True Singers
Working for that breed of performing artist who sings quite well and requires a refined approach in order to do their best on stage
Welcome to doing monitor sound for a singer.
There are many kinds of professional entertainers, songwriters and celebrities, but now you’re working for that breed of performing artist who sings quite well and requires a refined approach in order to do their best on stage.
If she is just starting out, microphone selection is something you could explore, but having used one particular microphone for vocals for her entire career, it’s simply a comfortable pair of shoes, its balance and weight feel familiar, much less its frequency response.
The microphone must be used with pad and high-pass engaged, as the proximity effect is too much, and she’ll blow up the capsule on the big notes.
If her microphone doesn’t seem that loud when you speak into it, that’s because she produces a lot of level that most performers can’t put out.
Please store the actual show microphone safely away until she arrives on stage. When you need to talk into her channel, use another microphone so that you’re not blowing germs into hers.
And I probably don’t need to tell you that there’s no smoking anywhere near the stage while she’s in the venue.
We’ve always used an XL-42 mic-pre and EQ for the “money channel.” There are a couple of peaks in this microphone’s response up high that we cut with narrow parametric filters so we don’t kill the “sparkle”.
Originally we tried just using a graphic EQ, but most problems don’t fall right on ISO centers, so parametric EQ is really helpful.
We started using a VariCurve, but switched to a Compact OmniDrive Plus, using the extra inputs and outputs for equalizing a reverb send, the band’s vocal send, and stereo side-fill inserts.
When forced to go back to simply using graphic EQ, I’m reminded of how clumsy most are for monitors.
I have a long menu of EQ filters that help, each for a specific characteristic of voice, microphone or room, but you can try some basics by inserting a graphic EQ into the vocal channel.
A model with minimal filter interaction and a high-pass that sweeps to 250 Hz helps the most. First, sweep that high-pass up all the way – it seems radical, but you’re hearing lots of lows from the mains already.
Now take 160 Hz and sink it. It’s an evil frequency that simply must be killed, and in combination with the high-pass, you’re almost there.
Another frequency that gets in the way of female vocals is 315 Hz, as it’s heard clearly in the head and in the mains. Sink it halfway.
Now with 400 and 500 Hz you must be careful, as this is where the natural singing voice turns into falsetto. It’s made more difficult because it’s really 450 Hz, as well as by the fact that some of these frequencies actually come out of the sides of both the other monitors and the mains.
If you take too much of this out of the singer’s mix, it still hits her from all the other loudspeakers which also have her voice in them, but aren’t pointed at her. Oh, and sink the crossover frequency in each mix.
You must split the vocal microphone into two channels: one for her, a second for band wedges.
She can adjust the EQ on her voice in the band’s mixes as well, but it won’t be the same as what she wants on her voice in speakers pointed at her.
The other half of a stereo graph should offer enough flexibility when combined with channel EQ. Programmable EQ helps, as room modes change nightly per venue, while voice and microphone fixes remain somewhat steady.
That’s enough EQ for now. You already owe me lunch.
Much as guitarists hate wireless companding, my singer doesn’t like the response of wireless microphones.
Because she takes it off the stand, the microphone must be taped to the cable so it can’t accidentally come unplugged. It’s a condenser, so it makes a loud sound if this happens.
A roll of PVC electrical tape is handy for this daily, and seemingly insignificant, chore. Insignificant, that is, until it comes unplugged in front of a full house at a sensitive moment in the show. Or on live television.
She won’t use in-ear monitors (IEM). We’ve tried them several times, and we’ve tried the best. When we last had background singers, we talked them into IEM, using John Hardy pre-amps for the vocal microphones, giving each their own reverb and individual stereo mixes with everything panned and tweaked, in hopes that they could convince her that it was the way to go.
And indeed, they said they’d never heard themselves better, but she still insisted she’d rather be deaf than sing with things stuck in her ears. She enjoys the intimacy of bantering with her audience between songs, and that would be lost without the ability to quickly answer comments from any direction they originate.
Most musicians like goofing around and jamming. It’s why they became musicians in the first place. It’s OK for them to have a little fun, if you’ve allowed time for it.
But when the singer arrives on stage for sound check, everything else must already have been checked, so there’s no reason to pay attention to anything but her and her microphone.
Band and crew understand that it’s entirely her stage when she’s on it, and all else is attended to only when she’s done out of courtesy and respect.
Compression & Reverb
Singers hate compression, but FOH engineers use it to create the studio sound that exemplifies pop music. Good singers learn to put up with it. Great singers learn how to cheat the compressor with microphone technique.
Monitor engineers naturally eschew the use of compression on the vocals. When the singer hits a big note, the stage monitors faithfully reproduce it, while the voice in the mains gets knocked back by 3 to 10 dB by the FOH comp.
The result is that the singer hears the sound jump out of the stage monitors on the big notes, while it collapses in the mains. When the microphone is pulled away on big notes, the singer avoids the compressor and the vocal collapses far less from her perspective.
Younger singers have difficultly understanding this, older ones do it instinctively.
The reverb plays an important role in the monitors. We’ve always used an M5000, because of its dense early reflections (ER), as well as a four-way crossover. If you can, turn your reverb effect down so that only ER is heard.
First, EQ the reverb to sound natural.
Next, pre-delay is the critical adjustment, and it needs to vary from 10 to 30 milliseconds, depending on the room. Like any special sauce, a little bit goes a long way – don’t overdo it.
The four-way crossover allows the reverb to be tailored to complement the natural venue acoustics. Shortening the lower frequencies that already dominate the stage helps. Leaving the mid-highs the longest helps brighten the room for the singer.
There are lots of microphone cable choices. When I inherited this gig, I found the sound company’s tech taping spare cables into the drum loom.
After a week of constant failures with a mostly phantom-power input list, I bought the show a new set of 30 generic “quad” microphone cables for about the cost of a half-case of gaffe tape. Before sound check was even over the band was asking what had changed, and if I had a new console that day.
The case for carrying the main “show” cable for the lead vocal is clear on many levels. What has been lost, now that a generation has grown up with wireless microphones, isn’t just coiling a “figure-eight,” but also the ancient art of paging a vocalist’s microphone cable: one of those gigs handed down father-to-son a lifetime ago.
Cable pagers of yesteryear know that performers who travel with microphone in hand tend to do so in a consistent direction, often clockwise. And this happens several times each song, so that by show’s end, there can be quite a twist in the microphone cable.
This can be counteracted by simply taking microphone and cable before the show starts, and spinning it the other way a dozen or so times. This “buys” enough turns that by the encore, the cable is still relatively un-kinked.
There are many designs and approaches to wedge-based monitoring. I’ll skip the theory and just tell you what we spent 10 years finding out on tours and one-offs.
Her stereo wedges need to be a dozen feet apart and facing each other so that the sound comes from each side, helping the reverb’s stereo effect. The back of the wedge must be propped up with a two-by-four, so the horn is on-axis at the down-stage center position.
She travels to the sides of the stage so a second wedge, similarly angled, is needed about 12 feet past the first. This second pair can be a mono mix, as she’ll only hear one at a time except when she’s dead center.
You can high-pass this mix more than the others, as lots of low-frequency energy is coming off the mains at side-stage. A fourth mix that helps is the so-called upstage “butt-fill.”
The only thing that goes in her mix is her vocal and reverb. In extremely reverberant halls a very small amount of piano for pitch and maybe kick-drum for time will be needed, but only if she asks for it.
By the way, she has as many signals as a New York Mets third-base coach. Never take your eyes off her on stage and you’ll start learning them right away.
Good luck. She’s the best and deserves the best.
Mark Frink is an independent engineer and free-lance tech writer.
Proper Loudspeaker Placement: How To Avoid Lobes and Nulls
Let’s say the sound system in the house of worship you’re working on goes into feedback whenever microphones pass under the loudspeaker array.
Worse yet, there are “soft spots” in some sections of the audience area.
Choir mics “squeal” before they are loud enough and the podium mic “rings” annoyingly for some presenters. You know that the system should be equalized to eliminate these problems.
So you install an equalizer and the feedback is reduced, but the soft spots persist and the system just doesn’t sound good.
But that’s why you, the consumate audio profesiponal, are there.
After some careful listening tests, a “problem area” within the room is chosen for the measurement mic placement.
This is a place in the seating where people complain that they can’t hear, or a place where the mic consistently goes into feedback, such as directly under the loudspeaker array. The measurement looks something like that shown in Figure 1.
Figure 1: Comb filter caused by a time offset between two loudspeakers. The audibility of comb filters has always been the subject of heated debate. While humans may not be very sensitive to narrow notches in the spectrum, the spacial lobing implied by the comb filter can excessively excite rooms and dramatically reduce gain-before-feedback.
The response clearly shows an acoustic “comb filter” that results from a time offset between two sound arrivals at the measurement position.
The measurer first makes certain that the secondary arrival isn’t simply the result of a bad mic placement (floor bounce, etc.) or loudspeaker placement (ceiling or wall bounce, etc.).
After ruling out these two possibilities, it becomes apparent that the multiple arrivals are due to the overlapping patterns of two loudspeakers being used to provide audience coverage.
Standing at the mic position and simply looking at the array, noting that you are clearly within the coverage pattern of two loudspeakers suspended over the stage, confirms the suspicion. Sound travels at a single constant speed.
Yet, in this case, there are two loudspeakers.
Therefore every location in the room that is receiving direct sound at equal level from both loudspeakers (except for the center line where the distance to each loudspeaker is exactly equal), will receive two signals arriving at different times.
This time offset causes the comb filtering.
Figure 2: Represents the lobing (a form of destructive interference) between two spaced loudspeakers at a single frequency.
An acoustic comb filter can produce undesirable coloration of the sound and loss of definition. It can even change where the sound seems to be coming from, ruining the “imaging” of the system.
The possible “options” are:
1. Set the analyzer resolution to smooth the comb filtering, and then adjust the equalizer for the desired response. This is not a solution. It just masks the problem.
2. Ignore the comb filtering and simply “notch” the frequencies that are prone to feedback. Even though this is a common approach, it is treating the symptom and not the problem.
Excessive frequency notching can ruin the sound of the system. Why filter out sound that needs to be there?
3. Conclude that humans aren’t all that sensitive to narrow notches in the spectrum, so the comb filters are just something that we can live with.
This is rationalizing the problem and is simply not true. It’s usually the explanation provided by someone who is responsible for the problem in the first place!
4. Get out the old one-third octave real-time analyzer. You can’t see the comb filters on it.
For many years, audio professionals did not have high-resolution analyzers that could identify arrival time problems. The system response looked fine on a one-third octave analyzer, but it still sounded bad.
Today’s analyzers are vastly more powerful and can reveal much more about the nature of a sound problem.
5. Inform the owner that the current loudspeaker placement has created some problems that cannot be “corrected” electronically. The only real solution is to relocate the existing loudspeakers or redesign the array.
Unfortunately, the sad reality is that only the last option is likely to fix the system.
Figure 4: Placing a greater physical distance between the loudspeakers.
An acoustic comb filter is a symptom of a more significant problem. When two loudspeakers are placed in close proximity, the resultant distance offset will cause “lobing” in the speaker’s radiation pattern.
Lobes can be described as “fingers” of sound pressure “maximums” in the three-dimensional space surrounding the array.
The fingers are separated by nulls or axis of minimal sound pressure level. The fingers typically cause problems with microphones, since a mic is likely to feedback when it is placed within a lobe.
The nulls cause problems for the audience, since parts of the audio spectrum that are critical for speech intelligibility (understanding the words) are cancelled at some listener’s seats.
Figure 5: Use of aggressive pattern control to reduce the overlap.
When a series of these lobes and nulls exist, the visual representation of the frequency response at one listener position will resemble the teeth of a comb, with a sequence of peaks and valleys.
This is a far cry from the “perfect” system response that would look more like a flat line. As such, a comb filter is the symptom of a spatial problem that has resulted from a loudspeaker selection and placement choice.
To illustrate, look at the simulations shown below (Figure 6), which show such a condition performed with the EASE sound system design software package.
Two loudspeakers with low directivity control have been separated by two feet.
The resultant does not represent accurate sound reproduction and can cause the afore-mentioned problems with acoustic gain and speech intelligibility.
Please note that it is certainly possible to build quality “arrayable” loudspeakers, and there are a number of good examples in the marketplace. However, all of them have several parameters in common:
1. Large physical size
2. Horn-loaded components
3. Aggressive pattern control to minimize interaction with adjacent loudspeakers
If these loudspeaker requirements present problems for a particular venue due to the required large physical size, then smaller loudspeakers can be used (usually in greater number) if they are placed sufficiently close to the listeners (i.e. exploded arrays or distributed systems).
Figure 2 (on page 38) outlines the options, and there aren’t many.
Figure 6: Left - The balloon plot displays the 3-dimensional sound radiation from the two-device array described in the text. Right - The traditional horizontal polar plot views the equator of the balloon as viewed from above for one frequency.
Radio broadcast engineers have understood for years the importance of proper antenna array design to control lobing in RF radiation to steer their signal to certain areas within the listening range and away from others.
For instance, if a station is licensed to radiate 50 killowatts of power, they can use an antenna array to steer the radiated signal up and down an interstate highway rather than out across a sparsely inhabited area. In fact, if they do it wrong, they can be in violation of federal law and therefore subject to prosecution.
Loudspeaker array designers must work with the same physical laws and principles as antenna designers. The only difference is that they can’t be prosecuted for bad sound.
Balloon plots are useful because they show the three-dimensional radiation pattern from a loudspeaker or group of loudspeakers located at the center of the balloon.
The plot describes what is happening at a single frequency. The plots can be generated for multiple frequencies to more fully describe the performance of an array. The balloon plot of a “perfect” loudspeaker would be the same, regardless of frequency.
Comb filtering in the magnitude response (a measurement at a single point in space) is evidence of lobing in the spatial radiation of the array.
Pat Brown teaches the Syn-Aud-Con seminars and workshops. Synergetic Audio Concepts (Syn-Aud-Con) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, Syn-Aud-Con is dedicated to teaching the basics of audio and acoustics. For more information visit their website.
The Analog Tape Recorder: An Introduction
What every engineer should know about analog recorders, excerpted from Huber & Runstein's "Modern Recording."
This article is the first half in our series on the analog tape recorder, excerpted from Huber & Runstein’s book Modern Recording Techniques, Seventh Edition. For the second half, click here.
From its inception in Germany in the late 1920s and its American introduction by Jack Mullin in 1945 (Figure 1), the analog tape recorder (or ATR) had steadily increased in quality and universal acceptance to the point that professional and personal studios had totally relied upon magnetic media for the storage of analog sound onto reels of tape.
With the dawning of the project studio and computer-based DAWs, the use of two-channel and multitrack ATRs has steadily dwindled to the point where no new analog tape machine models are currently being manufactured.
In short, recording to analog tape has steadily become a high-cost, future-retro, “specialty” process for getting a certain sound.
This being said, the analog recording process is still highly regarded and even sought after by many studios as a special sonic tool … and by others as a raised fist against the onslaught of the “evil digital empire.”
Without delving into the ongoing debate of the merits of analog versus digital, I think it’s fair to say that each has its own distinct type of sound and application in audio and music production.
Although professional analog recorders are usually much more expensive than their digital counterparts, as a general rule, a properly aligned, professional analog deck will have a particular sound that’s often described as being full, punchy, gutsy and “raw” (when used on drums, vocals, entire mixes or anything that you want to throw at it).
Fig. 1: John T. (Jack) Mullin (on the left) proudly displaying his two WWII vintage German Magnetophones, which were the first two tape-based recorders in the United States. (Courtesy of John T. Mullin.).
In fact, the limitations of tape are often used as a form of “artistic expression.” From this, it’s easy to see and hear why the analog tape recorder isn’t dead yet … and probably won’t be for some time.
To 2-Inch Or Not To 2-Inch?
Before we delve into the inner workings of the analog tape recorder, let’s take a moment to discuss ways in which the analog tape sound can be taken advantage of in the digital and project studio environment.
Before you go out and buy your own deck, however, there are other cost-effective ways to get “that sound” on your own projects.
- Make use of plug-ins that can emulate (or approximate) the overdriven sound of an analog tape track.
- Rent a studio that has an analog multitrack for a few hours or days. You could record specific tracks to tape, transfer existing digital tracks to tape or dump an entire final mixdown to tape.
For the cost of studio time and a reel of tape, you could inject your project with an entirely new type of sound (you might consider buying a single reel of multitrack tape that can be erased and reused once the takes have been transferred to disk).
- Rent an analog machine from a local studio equipment service. For a rental fee and basic cartage charges, you could reap the benefits of having an analog ATR for the duration of a project, without any undue financial and maintenance overhead.
A few guidelines should also be kept in mind when recording and/or transferring tracks to or from a multitrack recorder:
- Obviously, high recording levels add to that sought-after “overdriven” analog sound; however, driving a track too hard (hot) can actually kill a track’s definition or “air.” The trick is often to find a center balance between the right amount of saturation and downright distortion.
- Noise reduction can be a good thing, but it can also diminish what is thought of as that “classic analog sound.” Newer, wide tape width record- ers (such as ATR Services’ ATR-102 1-inch, two-track and the 108C 2-inch, eight-track recorder), as well as older 2-inch, 16-track recorders, can provide improved definition without the need for noise reduction.
Magnetic Recording And Its Media
At a basic level, an analog audio tape recorder can be thought of as a sound recording device that has the capacity to store audio information onto a magnetizable tape-based medium and then play this information back at a later time.
By definition, analog refers to something that’s “analogous,” similar to or comparable to something else.
An ATR is able to transform an electrical input signal directly into a corresponding magnetic energy that can be stored onto tape in the form of magnetic remnants.
Upon playback, this magnetic energy is then reconverted back into a corresponding electrical signal that can be amplified, mixed, processed and heard.
The recording media itself is composed of several layers of material, each serving a specific function (Figure 2).
The base material that makes up most of a tape’s thickness is often composed of polyester or polyvinyl chloride (PVC), which is a durable polymer that’s physically strong and can withstand a great deal of abuse before being damaged.
Fig. 2: Structural layers of magnetic tape.
Bonded to the PVC base is the all-important layer of magnetic oxide. The molecules of this oxide combine to create some of the smallest known permanent magnets, which are called domains (Figure 3a).
On an unmagnetized tape, the polarities of these domains are randomly oriented over the entire surface of the tape.
The resulting energy force of this random magnetization at the reproduce head is a general cancellation of the combined domain energies, resulting in no signal at the recorder’s output (except for the tape noise that occurs due to the residual domain energy output).
When a signal is recorded, the magnetization from the record head polarizes the individual domains (at varying degrees in positive and negative angular directions) in such a way that their average magnetism produces a much larger combined magnetic flux (Figure 3b).
Fig. 3a: Orientation of magnetic domains on unmagnetized and magnetized recording tape - The random orientation of an unmagnetized tape results in no output.
When the tape is pulled across the play- back head at the same, constant speed at which it was recorded, this alternating magnetic output is then converted back into an alternating signal that can then be amplified and further processed for reproduction.
The Professional Analog ATR
Professional analog ATRs can be found in 2-, 4-, 8-, 16- and 24-track formats. Each configuration is generally best suited to a specific production and postproduction task.
For example, a 2-track ATR is generally used to record the final stereo mix of a project (Figures 4 and 5), whereas 8-, 16- and 24-track machines are obviously used for multitrack recording (Figures 6 and 7).
Fig. 3b: Orientation of magnetic domains on unmagnetized and magnetized recording tape - Magnetized domains result in an average flux output at the magnetic head.
Although no professional analog machines are currently being manufactured, quite a few decks can be found on the used market in varying degrees of working condition.
Certain recorders (such as the ATR-108C 2-inch, multitrack/mastering recorder) can be switched between tape width and track formats, allowing the machine to be converted to handle a range of multitrack, mixdown and mastering tasks.
The Tape Transport
The process of recording audio onto magnetic tape depends on the transport’s capability to pass the tape across a head path at a constant speed and with a uniform tension.
In simpler words, a recorder must uniformly pass a precise length of tape over the record head within a specific time period (Figure 8).
During playback, this relationship is maintained by again moving the tape across the heads at the same speed, thereby preserving the program’s original pitch, rhythm and duration.
This constant speed and tension movement of the tape across a head’s path is initiated by simply pressing the Play button.
The drive can be disengaged at any time by pressing the Stop button, which applies a simultaneous breaking force to both the left and right reels.
Fig. 4: Otari MX-5050 B3 two-channel recorder. (Courtesy of Otari Corporation, http://www.otari.com
The Fast Forward and Rewind buttons cause the tape to rapidly shuttle in the respective directions in order to locate a specific point.
Initiating either of these modes engages the tape lifters, which raise the tape away from the heads (definitely an ear-saving feature).
Once the play mode has been engaged, pressing the Record button allows audio to be recorded onto any selected track or tracks.
Beyond these basic controls, you might expect to run into several differences between transports (often depending on the machine’s age). For example, older recorders might require that both the Record and Play buttons be simultaneously pressed in order to go into record mode; while others may begin record- ing when the Record button is pressed while already in the Play mode.
On certain older professional transports (particularly those wonderful Ampex decks from the 1950s and 1960s), stopping a fast-moving tape by simply press- ing the Stop button can stretch or destroy a master tape, because the inertia is simply too much for the mechanical brake to deal with.
In such a situation, a procedure known as “rocking” the tape is used to prevent tape damage.
The deck can be rocked to its stop position by engaging the fast-wind mode in the direction opposite the current travel direction until the tape slows down to a reasonable speed … at which point it’s safe to press the Stop button.
In recent decades, tape transport designs have incorporated total transport logic (TTL), which places transport and monitor functions under microprocessor control.
This has a number of distinct advantages in that you can push the Play or Stop buttons while the tape is in fast-wind mode without fear of tape damage.
With TTL, the recorder can sense the tape speed and direction and then automatically rock the transport until the tape can safely be stopped or can slow the tape to a point where the deck can seamlessly slip into play or record mode.
Most modern ATRs are equipped with a shuttle control that enables the tape to be shuttled at various wind speeds in either direction.
This allows a specific cue point to be located by listening to the tape at varying play speeds, or the control can be used to gently and evenly wind the tape onto its reel at a slower speed for long-term storage.
The Edit button (which can be found on certain proffessional machines) often has two operating modes: stop-edit and dump-edit.
If the Edit button is pressed while the transport is in the stop mode, the left and right tape reel brakes are released and the tape sensor is bypassed.
This makes it possible for the tape to be manually rocked back and forth until the edit point is found.
Often, if the Edit button is pressed while in the play mode, the take-up turntable is disengaged and the tape sensor is bypassed.
This allows unwanted sections of tape to be spooled off the machine (and into the trash can) while listening to the material as it’s being dumped during playback.
Fig. 7: Studer A827 analog multitrack recorder with autolocator. (Courtesy of Studer North America, http://www.studer.ch
A safety switch, which is incorporated into all professional transports, initiates the stop mode when it senses the absence of tape along its guide path; thus, the recorder stops automatically at the end of a reel or should the tape accidentally break.
This switch can be built into the tape-tension sensor, or it might exist in the form of a light beam that’s interrupted when tape is present.
Most professional ATRs are equipped with automatic tape counters that accurately read out time in hours, minutes, seconds and sometimes frames (00:00:00:00).
Fig. 8: Relationship of time to the physical length of recording tape.
Many of these recorders have digital readout displays that double as tape-speed indicators when in the “varispeed” mode.
This function incorporates a control that lets you vary the tape speed from fixed industry standards.
On many tape transports, this control can be continuously varied over a ±20% range from the 7 1 2 , 15 or 30 ips (inches per second) standard.
Stay tuned for the next part of the series where we’ll discuss cleaning, alignment, archiving, and equalazation.
Click to enlarge book cover
This article is the first part in a series on the analog tape recorder, excerpted from Huber & Runstein’s book Modern Recording Techniques, Seventh Edition For the second half, click here.
Posted by admin on 06/13 at 08:26 AM