Friday, July 29, 2011
Church Sound Recording: Start By Making The Right System Connections
A discussion of the correct procedures for connecting a recording device to your church sound system
At first glance, recording the audio of worship services might appear to be a relatively simple procedure.
Just connect a recording device to the sound system, set levels, press “record” and you’re ready to roll, right?
If only it were that simple.
Still, it’s true that capturing quality recordings is not rocket science. The most important thing is to get off to the right start.
Therefore, let’s start at the beginning and take a walk through the essentials.
Every sound system is a group of audio devices, and electronic signals flow through each device, from one device to the next. Each device is equipped with inputs and outputs.
Simply stated, inputs are connections that receive incoming signals, while outputs are connections that distribute outgoing signals.
A basic rule: the output of a device always feeds the input of another device.
Another basic rule: cables link outputs to inputs, and these cables need to be outfitted with correct connectors at each end.
It’s vital to understand the correct procedures for connecting a recording device to your sound system.
For our purposes here, we’re talking about two-channel recording devices such as CD recorders. However, note that the interconnection information presented here applies to any two-track recording device, such as memory recorders, PDA recorders, hard drive recorders, and others. (Even good ol’ cassette tape recorders!)
To add a two-track recording device to a sound system, connect the mixer (mixing console) output to the recorder input.
But which output?
Take a look at the back of the mixer, and you’ll see some connectors that are labeled with one of these names:
Master Output (Master Output)
Mix Out (Mix Output)
Stereo Bus Out (Stereo Bus Output)
Stereo Mix Bus
Line Out (Line Output)
Channel 1 Out, Channel 2 Out
Program Out (Program Bus)
Want to know a deep, dark secret? These terms all mean the same thing! (Audio people love to make life more complicated with extra jargon. Why agree on one name for something when several will do the trick?)
But for purposes of our discussion here, I will call these connectors Master Output, which is the most commonly used term.
The Master Output is a sum of all input signals feeding the mixer. And, the Master Output is already connected to the rest of the system - power amplifier (s), equalizer(s), etc.
Thus, because the Master Output is already in use, find a spare Master Output. What? There is no pare on your mixer?
Purchase a Y cable, available for a very reasonable price from many music retailers and electronics stores.
As shown in Figure 1, a Y cable is true to its name - a short cable assembly shaped like a “Y.”
It splits one signal into two, so that a mixer’s single output can feed both the sound system and a recording device. If your mixer is stereo (with two output channels), two Y cables are required - one per channel.
Figure 1: Two types of Y cables. (click to enlarge)
Some mixers offer dedicated “Recording Output” or “2 Track Out” connectors, and a recording device can be linked from there.
And note that some mixers include a built-in graphic equalizer. A recording devices should NOT be connected to the output of this equalizer. You don’t want your recording to be equalized!
Therefore, keep in mind to always use connector(s) in the signal path that come before the graphic equalizer.
Note that many mixers offer more than one type of output connectors. Again, why settle on one when more can serve to confuse and complicate things.
Possible connectors include:
RCA (also called Phono)
XLR (also called Three-Pin)
1/4-inch (also called Phone Jack, TS for “tip-sleeve” or TRS for “tip-ring-sleeve”)
All three types are shown in Figure 2. Each cable must have a connector that mates with your mixer (on one end of the cable) and a connector that makes with your recorder (on the other end).
Figure 2: Equipment connectors and mating cable connectors. (click to enlarge)
Suppose your mixer offers a Master Output already connected to the sound system, as well as a spare Master Output, which is an RCA connector. Your recording device input is also an RCA connector.
Thus the cable connecting the spare Master Output and the recording device should have RCA connector at both ends. More specifically, a cable with RCA “male connectors on both ends.
If the mixer has a stereo Master Output, and the recording device has a stereo input, the connection can be made with individual RCA cables (one per channel) or a stereo RCA cable.
Another example: your mixer offers Channel 1 Out and Channel 2 Out, and these are 1/4-inch output connections.
Your recording device offers RCA input connections.
The solution is two Y cables, one for each channel, and each with a 1/4-inch “male” plug on one end and a 1/4-inch “female” phone jack on the other.
Plug an adapter cable into one of the phone jacks as shown.
Figure 3 shows this configuration.
And for clarity, note that only one channel is illustrated.
What if your mixer has one mono Master Output, but your recording device offers two inputs for stereo recording?
Figure 3: Using a Y cable to connect a mixer with phone jacks to a recorder with RCA connectors. (click to enlarge)
Time again for a Y cable, with the single connector end plugged into the mixer output, and the double connector end plugged into both inputs of the recording device.
Where To Go
I highly recommend purchasing cables already outfitted with the correct connectors for your system, rather than trying to make them yourself. This saves time and hassle, in addition to insuring a rock-solid, reliable connection.
Another option is to use a cable with two identical connectors; say, 1/4-inch connectors at both ends. Then add adapters (i.e., a 1/4-inch male to RCA male adapter) to make the correct connection.
One other option that should not be overlooked is to purchase the correct cables from your system contractor.
Keep in mind that with cables, like everything else, you get what you pay for. Inexpensive cables usually live up to the negative definition of “cheap - they can add noise to a system and fail earlier. Best to spend a little more for quality.
A final note: always be sure to label both ends of each cable according to what they plug into, so you can easily tell what is plugged into what.
Making labels is easy - just use masking tape and a felt-tip marker.
Congratulations, you’re connected.
Next time we’ll talk about hooking up devices with XLR or TRS connectors.
AES and SynAudCon member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.
More Church Sound articles by Bruce Bartlett:
Microphone Strategies That Produce Great Results With Church Choirs
Preventing “Hollow” Sound With Microphone Techniques
Identifying & Solving Microphone Problems
Friday, July 22, 2011
PreSonus Unveils New AudioBox VSL Series Interfaces
A series of compact, rack-mountable USB 2.0 interfaces with integrated software that provides the dynamics processing and EQ of a StudioLive Series mixer
PreSonus has introduced the AudioBox 22VSL, AudioBox 44VSL, and AudioBox 1818VSL, a series of compact, rack-mountable USB 2.0 interfaces with integrated software that provides the dynamics processing and EQ of a StudioLive Series mixer.
All three AudioBox VSL Series interfaces incorporate Class A XMAX preamps with 48-volt phantom power; 24-bit, 96 kHz converters with 114 dB dynamic range; and PreSonus’ loud (150 mW), clean, clear headphone output. All three models also provide MIDI I/O and zero-latency monitor mixing.
As noted, the interfaces deliver reverb and delay effects with dedicated effects buses and the same Fat Channel compression, limiting, 3-band semi-parametric EQ, and high-pass filter found in the new StudioLive 16.0.2 mixer—except that the processing is done on a computer, using bundled PreSonus Virtual StudioLive (VSL) software, with extremely low latency. As a result, the user can monitor while recording with real-time effects.
VSL also provides complete editor/librarian functions, and over 50 professionally programmed Fat Channel processor presets are included for enhancing instruments and vocals.
Like all PreSonus interfaces, the new AudioBox VSL-series units ship with PreSonus Studio One Artist DAW (Mac/Windows) and support Core Audio and ASIO, so they work with virtually any recording software.
The first of these interfaces to hit the market will be the compact, 2-in, 2-out AudioBox 22VSL. It is bus-powered, offers two combo mic/instrument inputs and two balanced L/R main outputs, and is housed in a steel case.
With included VSL software, it provides two stereo effects buses with reverb and delay and Fat Channel dynamics processors, EQ, and high pass filter on both inputs and both DAW returns.
The AudioBox 22VSL is expected to be available in August 2011 with an anticipated street price of $199.
The 4-in, 4-out AudioBox 44VSL offers the same audio quality and basic features as the AudioBox 22VSL and more. Using the included VSL software, the AudioBox 44VSL provide an 8 x 4 software mixer with low-latency, Fat Channel processing for each analog input and DAW return, plus two stereo effects buses with reverb and delay.
The half-rack-space (which is rack-mountable) unit offers two mic/line inputs, two mic/instrument inputs, two balanced main L/R outputs, and four additional balanced line outputs.
The AudioBox 44VSL is expected to ship in September 2011 with an anticipated street price of $299.
The AudioBox 1818VSL has all of the audio features, processing, and effects found in its two smaller siblings, but it goes much further. It sports 2 mic/instrument inputs and 6 mic/line inputs, all with XMAX preamps; 8-channel ADAT I/O (4 channels at 88.1 or 96 kHz); stereo S/PDIF I/O; MIDI I/O; and word-clock output.
With the included VSL software, the AudioBox 1818VSL provides a computer-based 26 x 18 (22x14 at 88.1 or 96 kHz), ultra-low latency mixer with two stereo effects buses for the reverb and delay plus the Fat Channel processing from the StudioLive 16.0.2.
The AudioBox 1818VSL is expected to ship in September 2011, with an anticipated street price of $499.
Note that these new AudioBox VSL-series interfaces do not replace the current AudioBox USB, which remains a part of the PreSonus line.
Friday, July 15, 2011
Kenton Introduces MIDI USB Host For Interfacing With Standard DIN MIDI Connections
New device can also provide up to 500mA of USB bus power to the USB device if required
The new Kenton MIDI USB Host provides convenient interfacing of USB-enabled devices directly to those with standard 5-pin DIN MIDI connections, either in or out, without the need for a computer…
This enables, for example, direct connection of a USB controller or keyboard to another MIDI instrument that has only a 5-pin DIN MIDI connection.
Mains powered, the compact and rugged MIDI USB Host features a USB port (USB A socket) and MIDI In and Out sockets (both on 5-pin DIN).
MIDI data received at the MIDI In socket will be sent to the USB device, while MIDI data received from the USB device will be sent to the MIDI Out socket.
Additionally, the MIDI USB Host can provide up to 500mA of USB bus power to the USB device if required.
Kenton MIDI USB Host Specs:
Power Input: 5V DC (regulated) – use only the supplied adapter
Power: 90mA, 2.1mm plug (centre positive) – 510mA available for attached USB device
MIDI ports: 1 x In, 1 x Out (both 5-pin DIN)
Weight: 100g (excluding power supply)
Dimensions: 110 x 55 x 32 mm
Power supply: A 5V power supply appropriate to the destination country is supplied with the unit.
Leads: No leads are supplied with the unit
The MIDI USB Host comes with a 12 month (from purchase date) back to base warranty, (i.e. customer must arrange and pay for carriage to and from Kenton Electronics Ltd).
Flexible, Scalable Digital Intercom Systems For A Wide Range Of Applications
Digital intercom systems provide a foundation for point-to-point and group multi-connections
As digital has taken over much of the audio signal path, so it goes as well with intercom systems.
Digital intercom systems provide a flexible and scalable foundation for point-to-point and group multi-connections, and they’re enjoying increased usage across a wide range of live productions and installed applications such as churches, theatre, performing arts centers, stadiums, broadcast facilities and more.
Analog intercom systems often incorporate a patchbay that facilitates connecting (usually via shielded twisted pair, a.k.a., microphone cable) individual stations with a base station. A common concern is a bad cable that causes a short, taking down a channel.
Further, many of these systems are “partyline” – whoever has a headset with a microphone or a station with a mic can join the conversation – whether they’re wanted or not.
The result, for example, is production staff that ends up talking over each other, hampering communication. I’ve even heard a story of a show director sending a stagehand up to a spotlight location with a roll of duct tape, presumably to tape that particular spotlight operator’s mouth shut.
Digital intercom systems eliminate those types of problems in addition to providing vastly increased functionality. The producer in our example above could have the stagehand mute the mic of that spotlight operator from the cozy comfort of the production booth simply using PC software to “click” the mute button for that headset.
Many digital intercoms provide the capability to talk station to station, create an ad-hoc group, reconfigure groups, connect over the Internet to a remote location, bring in a phone call in and route it to a specific station or channel.
The recently introduced ASL Digital Intercom System, which supplies several handy features in addition to intercom capabilities.
Systems usually include a base station with a matrix that offers a scalable foundation for a dozen channels, on up through thousands of channels.
Digital technology provides a lot of flexibility in setting up user-to-user as well as group-based multi-connection communications. Accompanying software suites makes it pretty simple to configure a system for particular applications.
Digital intercoms typically operate on networks that can be configured in daisy chain, line or star topologies, with the system professional (designer/installer) able to choose the approach that is most efficient and cost-effective for the given application. And flexibility continues to improve.
For example, the ASL Digital Intercom allows direct person-to-person communication (like a telephone call) from any user station (including beltpacks) to any other user station, without tying up a channel.
In addition, this system offers text messaging from a keypad loudspeaker station to any other user(s).
In working with digital intercom systems, it’s useful to have a solid understanding of audio and signal flow as well as good working knowledge of networking, telephony and VoIP (Voice over Internet Protocol).
The key to successfully setting up a system is to have a clear plan as to how the system is to be implemented (i.e., using existing wiring, using a facility’s structured cabling, establishing new cabling, and so on).
Further, there needs to be a thorough understanding of any additional connections (like connecting an existing legacy 2-channel partyline system or connecting in a remote location) that need to be made.
Really, though, many of the same principles for designing and implementing a legacy system apply to designing and setting up a digital one.
Here’s a glossary of terms it’s good to be familiar with when working with digital intercom systems:
DTMF – Dual-Tone Multi-Frequency signaling, used for telecommunication signaling over telephone lines in the voice-frequency band between telephone handsets and other communications devices and the switching center.
MADI – Multichannel Audio Digital Interface, an industry-standard electronic communications protocol that defines the data format and electrical characteristics of an interface carrying multiple channels of digital audio.
SIP – Session Initiation Protocol, a signaling protocol widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP).
TCP/IP – Internet Protocol Suite, the set of communications protocols used for the Internet and other similar networks
AES3 – The digital audio standard frequently called AES/EBU used for carrying digital audio signals between various devices.
Star Topology – In its simplest form, a star network consists of one central switch, hub or computer, which acts as a conduit to transmit messages. Thus, the hub and leaf nodes, and the transmission lines between them, form a graph with the topology of a star.
Daisy Chain – Connecting each device in series to the next. If a message is intended for a device partway down the line, each system bounces it along in sequence until it reaches the destination.
Digital intercom systems are usually interconnected just like a computer network, commonly referred to as a LAN (local area network), and just like most audio networks, a digital intercom usually requires its own separate network.
It may run on the same kind of cable as a LAN, but still needs to be wired as its own network.
Note, however, that a couple of years ago, Clear-Com introduced a direct link from its IP-enabled V-Series panels to its Eclipse digital matrix intercom frames over an existing WAN or LAN Ethernet connection, allowing users to establish intercom communications in locations where direct cabling is lacking by employing the existing local IT infrastructure.
Most systems can be interconnected with CAT5, fiber optics and even good ol’ coaxial cable.
For control and configuration of the system, software is loaded on a computer, and the computer is connected to the network.
The computer may be on its own LAN and connected to many other computers, a connection can be made from the computers LAN to the digital intercom systems network giving control and configuration capabilities from anywhere that a computer is connected to that LAN network.
Clear-Com V-Series panel that enables a direct link of the company’s Eclipse digital
matrix intercom frames over an existing WAN or LAN Ethernet connection. (click to enlarge)
With the use of DSP (digital signal processing), very large digital intercom systems can be configured and reconfigured all from a software GUI (graphical user interface).
The bottom line is that digital intercoms, harnessing these technological advancements, can literally be configured into a networked system that spans the globe.
Gary Zandstra is a professional AV systems integrator with Parkway Electric in Holland, MI.
Posted by Keith Clark on 07/15 at 07:50 AM
APR Audio Brings Allen & Heath iLive Digital Consoles To Dual Stages At Glastonbury 2011
The systems were networked to provide input sharing and remote control via the iLive Editor PC software.
Rental company APR Audio provided systems for the West Holts and John Peel stages at Glastonbury 2011, selecting Allen & Heath iLive digital mixing systems to manage both front of house and monitor mixing duties, which were also networked for audio and control sharing.
On the John Peel stage, APR chose an iDR-64 MixRack connected to the largest control surface in the range, the iLive-176, for monitors. Twelve onstage mixes were provided, plus LR side-fills and up to eight stereo IEM feeds.
Engineer Fabrizio Piazzini additionally used the iLive MixPad app to make final tweaks. For front of house, an iLive-144 Control Surface was chosen, connected via CAT5 ACE to a modular iDR10 MixRack installed onstage.
The systems were networked to provide input sharing and remote control via the iLive Editor PC software.
“One of iLive’s great assets is that it is a networkable system, the benefits of which really come into their own in the fast paced festival environment. For instance, engineers can check FOH settings from the stage without battling through the crowds and mud,” explains Allen & Heath product manager Leon Phillips, who joined the APR team on site.
“Using iLive Editor software, we could drag appropriate channels from the generic patch directly onto the iLive surface layers in real time even while the engineer had already started working on the desk.”
A broadcast feed was also provided, split from the iDR-64 mic preamps and patched via the ACE network to 48 XLR outs on the iDR10 MixRack.
“Working with a generic festival layout is tricky, as you can end up with active channels spaced out all over the board and in different layers but this was not the case with iLive,” explains house engineer Stuart McKay. “The surface worked really well when it came to quickly putting bands’ channels next to each other, and hiding channels/outputs that I did not want visiting engineers to touch.
“Engineers who had never used iLive before soon got their heads around it and found the layout simple and intuitive. The way that ACE was implemented to feed the broadcast sends was really cool and the gain sharing that was in place between monitors and FOH worked really well.”
On The West Holts Stage, APR provided an iLive-144 surface at FOH, and an iDR-64 was built into the stage box input rack.
A compact iLive system, comprising an iDR-16 MixRack and iLive-R72 rackmount control surface, was used to manage compere mics, DJ, and BGM duties at front of house. Delayed near-field monitoring was also controlled by this set up at the front of house tower.
Allen & Heath Website
Thursday, July 14, 2011
Church Sound: The Promises (And Downsides) Of DSP
Do you really need DSP in every step of the signal chain?
The audio world is screaming down the digital highway with no end in sight. Manufacturers are introducing DSP (digital signal processing) products in record numbers, and why not? After all, digital makes everything better, right?
You don’t have to search hard to find DSP in all kinds of audio equipment, including amplifiers, consoles, powered loudspeakers, effects units, system processors and even microphones.
If you wanted really wanted to, you could string together a microphone with built-in DSP, send signal from it to a console with built-in DSP, then send the signal to a system processor (with DSP, of course), output the signal to a power amplifier with built-in DSP, and then send it out to the loudspeaker(s). Or, you could use powered loudspeakers with onboard DSP and skip the dedicated amplifier.
However, is there any point in that? Where should you have DSP in your system and how should you use it? The question may prove to be much simpler than the answer.
First, a point of clarification: when I talk about DSP, I’m referring to a digital processor built into a device. A device that has DSP built into it may or may not have a digital output, and the digital output might be a different protocol from other devices in the signal chain.
In my earlier (and extreme) example, it starts with the microphone with built-in DSP. The microphone capsule converts the analog signal to digital, goes through the DSP, and then a digital-to-analog (D/A) converter provides analog mic level output.
That analog signal then enters the console and hits an analog-to-digital (A/D) converter, then goes through the DSP and then another D/A converter. And then on to the DSP processor where once again we go A/D and D/A, finally into the amplifiers where, yes, you guessed it, once again the signal goes A/D and then D/A. Yikes!
An issue of major concern becomes the quality of the D/A and A/D converters. Just ask any studio guy how important the mic preamp is, and more importantly the A/D converter the preamp uses. At every conversion point there is the potential to as some “suck” to your sound quality.
Imagine the scenario above, we are talking suck x 8! I personally have never seen anyone with a set up like the suck x 8 above. Dave Mcnell has written a good overview of digital transports that will help explain the “suck” factor.
I have seen very real issues of choosing what digital transport to use to connect all of the devices together (assuming that the device has a digital output). With so many different digital transports to use it can become a bit daunting in trying to select which one to use.
Then the argument of is digital better enters into the conversation. Bruce Jackson and Steve Harvey have written an excellent article where the make the following statement:
Perhaps more significantly, the MP3 revolution may be leading to a generation that has no concept of distortion.
A few years ago, an engineering acquaintance related how he had to sit his son down and explain distortion to him.
At the risk of sounding like an old fuddy-duddy, kids today are used to hearing “crunchy” audio.
Younger people who have only lived during this digital era are largely unaware of ‘good’ distortion, the acceptable harmonic distortion of analog audio.
To them, what we consider bad digital distortion is simply a part of the music that they listen to daily.
Trading poorly “ripped” MP3s (regardless of legality), they have become so used to the crunchy sound quality of the format that, as some readers who have discussed the matter with their own children may attest, they may find it not only acceptable but even preferable to CDs.
The crunch of digital distortion, which is not limited to MP3s, of course, but can just as easily find its way into the digital recording process and onto disc or into the live sound arena, is unpleasant.
If you want to argue the merits of digital please read their article, but for our discussion lets’s just accept that digital is here and “it ain’t going away”.
So the merits of digital, the protocol and transport aside what can DSP do for me?
I believe some of the most meaningful solutions that have come out of DSP are that speaker manufacturers are utilizing the power of DSP to correct some inherent mechanical problems that occur when a loudspeaker convert sound back to acoustical energy.
In my interview with Kenton Forsythe, founder of EAW, Kenton talks about how some new “gray box” settings which he refers to as Focusing, correct the inherent difficulties in transducer design and the ability to impact time smear.
The reality is much better sounding speakers at potentially a much lower price. These “focusing” DSP solutions are provided by the speaker manufacturer and not meant to be “user” set.
So what DSP does the end-user get to play with?
What I see as optimal DSP usage in a system would be a mic with built in manufacturer set DSP that maximizes the characteristics of that microphone to make it sound as transparent as possible. This would be followed by a digital mixing console that would allow the sound engineer maximum control over the audio signal.
This would include level, equalization and all sorts of processing (gates, compression, reverb…etc.) including the ability to mix multiple sources and output them to various devices and/or locations.
After the mixing console the signal would be transferred to a system DSP box that would be set up (most likely by a systems contractor) to maximize the sound for a given space. This would include overall system EQ, crossovers, delay etc.
The final part of this chain would be an amplifier with manufacturer set DSP tailored to the speakers that are connected to the amplifier. And of course, the entire signal path would be all under the same digital transport thus creating an “all digital” solution where the only A/D and D/A is at the first input and on the last output of the system.
So, what are the potential pitfalls besides all of the potential A/D and D/A conversion? Simply, the ability to mess it up! The control that is available with DSP is extremely powerful and without a great understanding of how to properly use the DSP and essentially know what to touch and what not to touch and why is critical to the person operating a system.
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.
DiGiCo SD10 Package Offers Compact Solution For Mix Engineer Jon Lemon On Tour With The Fray
Touring system pairs SD10 With DiGiRack and Waves/DiGiCo SoundGrid bundle
Jon Lemon has been utilizing DiGiCo consoles on A-list tours for nearly eight years, so when the mix engineer was asked to handle front of house on a short tour with The Fray, he took the opportunity to check out the latest DiGiCo SD10 console.
The Denver rocker’s mini tour, opening for U2 on a handful of make-up dates from last year’s cancelled tour, allowed Lemon to assemble and test-drive a small touring package utilizing the compact SD10 in tandem with an SD 192kHz DiGiRack and the Waves/DiGiCo SoundGrid bundle.
“I was really keen to use the new SD10 with a full Waves package to see if I could actually do a tour without carrying any of my normal, expensive outboard gear, which I’ve gotten quite used to using,” Lemon explains. “Last year, I took an SD8 out on a Smashing Pumpkins tour because we were flying so many places and I wanted something small, lightweight and powerful but I still had all the outboard etc. to ship I liked it, but preferring the SD7, the new SD10 seemed more SD7-like — from the meters and faders to general feel.
“Having 96 channels with full processing appealed to me, as did the SD10s macros, especially on a single screen DiGiCo product, because you can get around a lot quicker by having them programed. The ability to have more inputs is highly important, too. Bearing in mind that on most modern tours these days you seem to do more flying, it was my goal to get this powerful, small package together, and for a travel pack, I think it’s totally ideal — and probably the way I’m heading for the future.”
Working out of his home studio a few weeks prior to rehearsals, Lemon was able to set up the console, working off a hard drive from the band’s previous live shows. “I was impressed with SD10’s layout right off the bat and liked having 16 plug-in racks to work with. Going into rehearsals in Las Vegas, we set up, switched on, and it was all there ready to go. It felt like I’d done a couple of weeks of rehearsals, when in reality I’d only spent two weeks at home and two days with the band and I had a full show ready to go.
“The nicer bonus was the new SD192 rack, which showed up during rehearsals - my inputs actually sounded better. I’ve been using those other racks every working day of my life since 2002, and it’s a completely noticeable difference to me. It sounded cleaner, more analog sounding, and the high-end is different too. More airy.”
Lemon’s enthusiasm with his tour package is enhanced with the addition of the DiGiCo/Waves SoundGrid bundle. “Between the console’s compressors, gates, dynamic EQ and dynamic compression—not to mention the straight Waves plug-ins on top of that—you’ve got quite an arsenal there! The Renaissance plug-ins, especially the reverbs, are so good. Combined with some of the high-end valve compressor emulations, H-delays, H-compressors, and SSL buss compressors, there’s so much available to you.”
An interesting situation presented itself on the tour and one in which the benefits of the Waves bundle were extremely evident. “On the U2 shows, the main end of the PA had two lefts and two rights — flown side by side—so one system was doing all the vocals and all the guitars, and the outside system was doing all the drums, bass, keyboards, etcetera. Normally, on my left and right I would have a Waves hardware piece, the BCL — the one with the Renaissance Compressor, a Maxx Bass and an L2 Ultramaximizer. I
” added another stereo bus for all of the shows we were doing with them and then it was really easy. I was able to drag up another rack and then copied and pasted my original three plug-ins—the Renaissance Compressor, Maxx Bass and L2 Ultramaximizer — into another rack and then instantly I had my settings available on two masters. That was a very easy way to handle having another output in a complicated system.”
“I also found that, where I would normally run parallel drum compression live and have to use an outboard compressor such as a Smart C2 Compressor or an SSL Bus Compressor, I was able to do it internally on the console using the SSL Bus Compressor. Normally, when you do it with analog outboard, you’ve got to loop in and out of a straight drum bus to make up the latency of the compressor to stop it phasing. All that’s calculated and compensated for you within the Waves/DiGiCo package now, so stuff like that is useful and quick. You can build up your own presets like you can do with the regular channel EQ, dynamics etc.”
In the long run, scaling down his live rig has ultimately given Lemon more advantages and a new way of working moving forward.
“Having the ability to use what you use on your DAW at home out in the field has sort of changed things,” Lemon says. “There’s no doubt about it, and long-term, it will give me a lot more nuance in my mixes ultimately because there’s so much more flexibility—especially in terms of presets and snapshots.
“Between what you can do with the Waves plug-ins and what’s available onboard the DiGiCo is fairly limitless. It’d be great with the SD7 as well, although I’m not sure whether I’d use that many plug-ins—32 racks is a lot—it would probably depend on the amount of inputs.
“But if you figure most bands I typically work with are between 50-90 inputs, I’d probably realistically only use 16 to 20 plug-ins or so.”
Another boom of the Waves SoundGrid came after the tour had ended when Lemon was asked to mix three songs from a live recording done at one of The Fray’s sideshows at the Fillmore Auditorium.
“The recording was done in Pro Tools by the venue as it was a fly date and one of the only shows I did not record myself. Funny enough, I was able to take many of the presets from the live console’s Waves plug-ins and import them into my Pro Tools session as I’d stored the files for the presets for each individual plug-in off the SD10.
” I found that intriguing; that I could get my drum and vocal reverbs and de-esser exactly where I’d honed it in live. It made the mixes I had to do for the band extremely quick and easy because I was able to access all my presets from the live show. And of course that will work the other way around as well, as you can hone reverbs and the like at home and import them into your DiGiCo/Waves session.”
What began as an experiment in streamlining a somewhat complex audio touring rig into a compact and features-packed system has perhaps become a new way of working for Lemon.
“I found this to be a viable system for many if not most of the bands that I’d be working with down the road. The trial proved most enlightening and will no doubt impact the day-to-day workflow for me from now on. And needless to say as a longtime DiGiCo user, I’m so pleased to be able to have an all-in-one that’s DiGiCo top to bottom and that I can flex for whatever band has got me on its payroll.”
Wednesday, July 13, 2011
Audio-Technica Offers Upgrades For ATCS-60 IR Conference System
System now features new voting module and updated software
Audio-Technica has announced upgrades for its ATCS-60 IR (infrared) Conference System from its contractor-exclusive Engineered Sound range of meeting/conference room offerings.
The ATCS-60 is a fully self-contained wireless distributed audio system using infrared communications technology, designed for corporate, government and educational applications.
The system offers many impressive and practical features, providing ease of setup and operation, and eliminating the need for microphone wiring and ceiling speakers. New upgrades include a voting module (model ATCS-V60) and updated software (ATCS-C60MAG-REG).
The new ATCS-V60 is a modular voting unit with five-button configuration that offers versatile voting options.
In addition to Yes/No and Yes/No/Abstain votes, the unit can be programmed to allow up to five options, useful when multiple candidates are standing for election. Up to 150 voters can be handled by an ATCS-60 system using ATCS-V60 modules.
The ATCS-60MAG-REG software has been newly enhanced. New features include support of the ATCS-V60 voting module, enhanced video and audio recording support, and the implementation of a simultaneous interpretation mode to support a total of four languages.
The ATCS-60 IR Conference System offers all the advantages of infrared communication – including security, high noise immunity and low power requirements, to name a few – and features wireless microphone units (ATCS-M60).
As a result, it offers more flexibility, with freedom to easily change microphone locations and system configurations. Even for a conference scenario of over 100 participants, the setup takes far less time than other conference systems.
The system also features sophisticated functions such as simultaneous interpretation for one native language and up to three foreign languages; automatic camera tracking to follow the switching of microphones; optional minutes recording; along with other features that help improve conference efficiency. The ATCS-60 system can also be fully portable.
The ATCS-60’s infrared technology offers secure communication. Since the infrared signal stays within the room, a meeting cannot be picked up or intercepted elsewhere, which can often be an issue with conventional RF systems.
The ATCS-60 system consists of the following components with U.S. MSRP as follows:
ATCS-M60 Delegate Unit (base unit)—$1,300
ATCS-60MIC Dedicated standard-length Gooseneck Microphone—$180
ATCS-L60MIC Dedicated Microphone (Long)—$200
ATCS-A60 IR Transmitter/Receiver—$1,050
ATCS-C60 Master Control Unit—$5,200
ATCS-D60 Distributor (splitter)—$260
ATCS-B60 Battery Charger—$1,360
ATCS-V60 Voting Module—$400
DMQ-60 Monaural Earphone—$50
LI-240 Lithium-ion Battery—$190.00
Setup of the ATCS-60 IR Conference System is simple and intuitive: the applicable number of ATCS-A60 IR Transmitter/Receiver Units (aka Transponders) are placed on the room’s ceiling space (or flown/elevated for full portability), and connected back to the master control unit with 75 ohm coaxial cable.
Each ATCS-M60 Microphone Unit, with dedicated ATCS-60MIC microphone, is set on the conference table (one for each participant) and assigned a discrete address; beyond that, it’s virtually plug-and-play.
Each ATCS-M60 Microphone Unit features a built-in dedicated loudspeaker, which will return the entire system’s audio (minus that conference member, therefore preventing feedback and echo).
The system operates in two modes, manual and automatic, allowing for a number of different meeting styles and levels of formality.
In manual mode, which works like a “discussion system” (commonly used with a more formal approach to corporate meetings), each meeting member pushes a button to request to speak, and each request is recognized by the meeting’s chairman.
The system’s ATCS-C60 Master Control Unit controls whether requests to speak are accepted in first-in-first-out or last-in-first-out order. This mode accepts up to 150 microphone units.
In automatic mode, which accepts up to 50 microphone units, all channels run through a “smart” mixer, which turns each microphone on and off depending on whether or not that meeting member is speaking. The mixer can be set (via the ATCS-C60 Master Control Unit) so that from one to five people can talk simultaneously.
The ATCS-A60 IR Transmitter/Receiver Unit has an oblong 120-degree coverage pattern. The ATCS-D60 Distributor acts as a splitter, expanding the Master Control Unit’s four IR Transmitter/Receiver Unit connections so that up to 16 ATCS-A60 units can be placed on a room’s ceiling space, depending on necessity presented by room size and shape.
The system’s included ATCS-C60MAG-REG software allows users to monitor and control the system using a computer screen with a graphical interface patterned after the room layout. An advanced version of the software can record meetings for electronic minute-taking.
The ATCS-M60 Microphone Units can run on battery power (LI240 Lithium ion battery, charged with ATCS-B60 Battery Charger) or via AC power with a standard adapter.
Mark Donovan, Audio-Technica sales engineer, states, “The upgrades to the ATCS-60 system in our extensive range of contractor and system integrator products represent Audio-Technica’s total commitment to providing install-sound solutions. The new features of the infrared conference system enhance its functionality, and with A-T’s hard-wired microphones, RF systems and the ultra wideband SpectraPulse wireless microphone system, Audio-Technica continues to offer engineers, sound contractors and installers a complete range of conference system options.”
The ATCS-60 system is currently available. As with all Audio-Technica Engineered Sound products, the system comes standard with a five-year warranty.
Tuesday, July 12, 2011
Lawo North America Appoints Philippe Guichard U.S. Product Support Manager
New York City presence brings increased support options for company’s customers
Lawo has announced the appointment of Philippe Guichard to the newly created position of U.S. product support manager.
Guichard’s appointment coincides with the opening of a new Lawo support/demonstration facility in New York City that Guichard will head.
As Lawo’s new support manager for the United States, Guichard oversees a number of crucial operations for the company and his presence is expected to dramatically enhance Lawo’s support services.
Guichard’s responsibilities include technical support for the company’s console and audio networking products as well as product training and demonstration.
In addition to telephone support services—which are now available through a U.S.-based telephone number—the new office in Gramercy Park, Manhattan will serve as a training/demonstration facility where existing and potential clients can meet to discuss and receive hands-on experience with Lawo’s various products.
Lawo’s new U.S. Support Operations center can be reached at 347-903-2965.
Guichard’s background covers a wide range of experiences that make him ideally suited for his new position with Lawo. Prior to joining Lawo, Guichard served as sales/product support manager North America for Fairlight U.S. In this capacity he handled training, technical service, sales support, and demonstration of the company’s audio and video products.
Prior to this, he served four years as vice president and partner for Pasadena, CA-based MediaGear, Inc., a corporation created to distribute, sell, and support Fairlight products throughout North America.
He also held a sales engineer position with Sonic Solutions of Novato, CA and a product specialist position with Audio Intervisual Design of Los Angeles.
“Lawo is a very highly respected brand that is recognized by audio professionals everywhere as a leader in both console and audio networking technologies,” says Guichard. “I believe my background in these areas is a good fit that makes me well suited for the responsibilities of this new position. I’m eager to set up shop and dig in and I look forward to contributing to the company’s continuing growth as we move forward.”
Herbert Lemcke, president of Lawo North America, states, “All of us here at Lawo North America are very pleased to have Philippe joining us. Philippe brings a wealth of experience and technical expertise to our company that, I’m certain, will be put to very good use. He has a thorough understanding of our company’s business and he takes the initiative at every opportunity. I’m confident he will be an invaluable asset to our company.”
Friday, July 08, 2011
Avid VENUE Digital Consoles At The Heart Of House, Monitor Systems Serving Maryland’s Reid Temple
“From a workflow perspective, the transition was absolutely painless. It really feels like you’re working on an analog console." - Teddy Davis, Reid Temple
Located in Glenn Dale, Maryland, Reid Temple’s sprawling, ultra-modern campus belies the church’s humble roots.
Originally located in nearby Bladensburg, the church’s original home, Dent Chapel, was purchased in 1900 from the conference of the newly formed Methodist Episcopal Church for the princely sum of $500.
Weathering more than a century of hardships and relocations, Reid Temple A.M.E. Church held its first worship service at its current location in the fall of 2004. Today, Reid Temple is one of the area’s largest houses of worship, with more than 10,000 members and two locations.
It’s also one of the most modern and technically advanced, offering a powerful worship experience both live and on the web. The church’s main 3,000-seat sanctuary is outfitted with a full-featured video production suite and high-end sound systems designed by Acoustic Dimensions, based around EAW KF Series line arrays and Avid VENUE systems at front of house and monitor positions.
A VENUE D-Show System with dual Stage Racks and an FOH Rack is the centerpiece of the church’s front-of-house mix position, handling its 56 inputs. As Teddy Davis, the Temple’s manager of audio and recording studio operations explains, the decision to replace their aging analog console with VENUE was sparked by the temple’s pastor, Dr. Lee P. Washington.
“After a theatrical play one Sunday, our pastor asked me if there was something we could purchase that would allow us to just press a button after an event and have all the levels and settings just revert back to their original settings,” says Davis, a veteran musician and engineer who has worked with artists such as Kirk Whalum, Yolanda Adams, CeCe Winans, and many others. “I did the research, and we auditioned pretty much every digital console on the market, and VENUE was really the ideal fit for us.
“From a workflow perspective, the transition was absolutely painless,” Davis continues. “It really feels like you’re working on an analog console. On the other hand, all of a sudden we had access to total recall, so we were able to create snapshots for every situation. That alone was a tremendous move forward, not only in terms of running the service, but for training purposes.”
Another major plus in moving to VENUE was its tight integration with the temple’s Pro Tools|HD 9 studio setup. “We’ve got a VENUE Profile System at monitors, with Stage and FOH Racks, and both systems are fully integrated with our Pro Tools setup,” says Davis. “We’re able to pull up Pro Tools from either position and record every service and every sermon. We can stream live over the web, as well as put together podcasts and clips for our website.”
“We were able to remove seven 96-point patch bays and several racks of outboard gear when we installed the VENUE system,” Davis concludes. “And ultimately we’ve ended up with far more processing power too. On so many levels, VENUE has really helped us to accomplish things we would have never been able to do before.”
Thursday, July 07, 2011
EQ As The Miracle Solution? What Equalization Can - And Cannot - Do
Some view the use of equalization as an art form, while others see it as distinctly technical in application
Almost every sound system has an equalizer. It can be as simple as a channel strip tone control or as sophisticated as a multi-band parametric.
Some view the use of equalization as an art form, while others see it as distinctly technical in application. The prevailing myth is that an equalizer can correct all of the shortcomings of a sound system.
Of course this is nonsense, but used properly, equalizers can improve the sound quality and general performance of most sound systems.
Rather than covering every aspect of equalizers and their use, let’s have a look at ways of achieving the best overall results in the shortest period of time. The approach is technical, not artistic.
Let’s work under the assumption that we have a means of measuring the magnitude of the frequency response of the direct field of the loudspeaker using one of the mainstream tools.
The procedure often followed for equalizing a loudspeaker is to place the measurement microphone on-axis and adjust for the flattest frequency response. This involves cutting and boosting some filters on the equalizer.
Those that are opposed to the use of boost filters may choose to arrive at the same resultant response by reducing (cutting) parts of the response to the lowest common denominator. This results in the same electrical curve, but without compromising headroom in the signal chain.
None of this is rocket science, and the process could be relegated to a computer algorithm.
As useful as this exercise is in producing a flat on-axis response curve, it ignores what is going on at other vantage points around the loudspeaker and may actually reduce the overall sound quality from the system/room for many of the listeners.
A modification to this approach: considering the off-axis as well as the on-axis response (not a new idea but this may speed the process). The goal of the equalization process is to produce a better sounding system for all of the audience. Yet a relatively small percentage of the audience sits in the on-axis position.
Figure 1: Axial frequency response of a multi-way loudspeaker, with the dip in response at 2 kHz due to phase cancellation between multiple drivers.
It would therefore seem ill advised to consider only the axial position when equalizing a system.
A possible solution is to average the response of a number of seating positions to arrive at the best “common denominator” curve for the equalizer.
This is called a spatial average, and while useful, there are some major drawbacks, including:
- The measurement microphone is in a different acoustic environment each time you move it. This makes it difficult to isolate the direct field without a lot of setup work.
- Loudspeaker interactions can produce huge swings in the frequency response from seat-to-seat. A spatial average can’t correct this. This means that equalization should initially be conducted on one loudspeaker at a time.
- While the symptoms of acoustic problems can be observed with this method (i.e. uneven coverage) the cause cannot.
- It’s not practical or possible to measure at all positions around the loudspeaker, so the equalization curve is influenced by relatively few measurements.
Because any adjustments with an equalizer affect the total radiated energy, it is wise to give consideration to all of the radiated energy. The spatial average is not a bad idea, it’s just hard to implement.
Different Way To Same
Another way to consider off-axis listener positions is to determine the base equalization curve for the loudspeaker by observing its three-dimensional radiation balloon.
A properly gathered balloon will reveal the anechoic response of the loudspeaker at all listener angles.
Because the direct field of a loudspeaker is considered to be largely independent of the acoustic environment (at least at short wavelengths), direct field equalization based on balloon data has a strong theoretical basis.
Figure 1 (page 1) shows the axial frequency response of a multi-way loudspeaker. The dip in response at 2 kHz is due to phase cancellation between multiple drivers in the box.
A boost filter at this frequency center will restore the axial response to flat. Observation of the polars and the entire radiation balloon at 2 kHz (Figure 2 and 3, page 2) shows that even though they cancel on-axis, the devices come into phase at two off axis positions. At these angles there is most likely a significant peak in the response for many of the audience members.
The “correction” made to the on axis response will likely worsen the off-axis response where a greater number of listeners are located. Worse case is that one of the off-axis energy lobes covers a microphone position, so a boost filter will likely worsen the gain-before-feedback.
Devices that have a destructive phase offset at one listener position are likely to have an in-phase relationship at another. The on-axis notch might better be addressed by the use of precision signal delay between the elements to bring them into phase rather than feed them more energy.
What They Do
In general, the use of equalization will inflate or deflate the radiation balloon at a specified octave band. The same effect occurs at all angles around the device.
But what is often needed is a reshaping of the balloon. This can be accomplished by using multiple elements and varying their physical spacing and relative delay. This is the heart and soul of loudspeaker design.
Boost and dip filters applied based on the axial response only have a minimal affect on the balloon shape for those frequencies. A good loudspeaker design will direct the radiation lobes toward the audience.
For frequency bands where this is not happening due to phase interactions, precision signal delay can be used to steer the lobes to thedesired position.
Figure 2 and 3: The polars (above) and entire radiation balloon at 2 kHz. Even though they cancel on-axis, the devices come into phase at two off-axis positions.
It’s counterproductive to just pump more energy into the loudspeaker and increase the level for all listeners, which is exactly what an equalizer does. It should be noted that psychoacoustics also plays a role in this.
Peaks in a frequency response are much more audible (and bothersome) to humans than dips in the response. We’re more aware of “too much” than “too little.”
A safe approach to equalization that embodies the theories described here is to avoid the use of boost filters when calibrating a sound system since it’s better to have “too little” than to have “too much.”
By observing the axial response for equalization, and using cut-only filters to smooth the resonant peaks, acceptable results can likely be attained for most listeners, because off-axis lobes will not be “inflated” by the process. The balloon data of a loudspeaker can reveal what will happen off-axis if a boost filter is implemented to smooth the on-axis response.
Getting The Balloons
The good news is that much of the work has already been done for you. It is not practical or even possible for end users to measure loudspeaker directivity balloons.
The burden is on the manufacturer to provide this data. Data balloons are available from many manufacturers in the data that they distribute for acoustic modeling programs.
Here is an orderly approach to using information from balloons plots to optimize the equalization process:
1) Measure the magnitude of the frequency response for the on-axis listener position. It is very important to use a mic placement and/or time window to exclude the room reflections (especially the floor bounce). Also the mic should be in the far field of the loudspeaker (at least 3x its longest dimension away).
2) Decide what correction is needed to flatten the direct field. Start with frequency bands that have too much energy and reduce them with the equalizer.
3) Next consider frequency bands where there is too little energy. Don’t boost them, but look at the radiation balloons and see if any off-axis lobes exist that are higher in level than the on-axis response. If so (and it is likely to be so) do not boost this region with the equalizer.
4) If the radiation balloon is well behaved off-axis then boost with care. This will usually only be true at the frequency extremes, where only a single device is producing sound and the band-limits of the transducer are being approached.
An exception could be the compensation need by constant directivity horns, but again, a look at the balloon data will tell if the high frequency roll-off is happening at other listener angles.
Figure 4 and 5: If the microphone were placed slightly off-axis when equalizing this loudspeaker, an attempt to achieve a flat response will likely overpower listeners at other angles.
5) Multiple device interactions should not be “corrected” with equalization. These will be clearly observable in the polars and radiation balloons (Figure 4 and 5, page 3).
If the microphone were placed slightly off-axis when equalizing this loudspeaker, an attempt to achieve a flat response will likely overpower listeners at other angles.
Inflate Or Deflate
It’s important to differentiate between inflating a radiation balloon and reshaping it. Equalizers inflate (or deflate), whereas delay or physical movement between loudspeaker elements reshapes the balloon.
Many “corrections” that are made by looking only at the axial response cause worse problems than they are solving. Manufacturer-supplied radiation balloon data is the most accurate and simplest way to look at the off axis response of a loudspeaker.
It’s plausible that the complete equalization of the direct field can be determined from the axial frequency response and the spherical balloons.
The remaining room-dependent equalization chores can be left to the installer or commissioner of the system. This can include compensation for boundary loading, device coupling, etc.
The possibility also exists based on the above for a manufacturer to provide an appropriate equalization curve in the form of a simple magnitude vs. frequency plot or as a setting that can be imported into popular digital signal processors.
Pat and Brenda Brown own and operate Syn-Aud-Con, conducting training seminars around the world. For more info go to www.synaudcon.com.
Gand Sound Supplies Nexo GEO Tangent Arrays Fantabuloso “Dos” In Chicago
GCS provided 21 GEO T4805s and three GEO T 2815s per side, seven 4805s and two 2815s per side for out fills, along with 28 CD18 subwoofers
The recent Fantabuloso “Dos” 2011 at the Allstate Arena just outside of Chicago had Gand Concert Sound (GCS) once again providing audio system support that included a Nexo GEO T tangent array rig.
Artists appearing at the concert included Pitbull, Mike Posner, Lupe Fiasco, Taio Cruz, T-Pain, Jesse J, Far East Movement, David Gueta, and Tinie Tempah.
GCS supplied 21 GEO T4805s and three GEO T 2815s per side for mains, seven 4805s and two 2815s per side for out fills, supported by 28 CD18 subs.
Six NEXO PS10s were used for front fills, with eight bi-amped NEXO PS15s down stage with Yamaha T5n amps and eight NEXO PS15s with Nexo 4x4 NXAMPs. Nexo Alphas were used for side fills.
“We are in our ninth season using the GEO T, being the first company in the U.S. to invest in an arena-size rig in 2003,” states GCS president Gary Gand. “Constant software upgrades have allowed NEXO to stay at the forefront of line arrays.
“Many other companies adopted ‘marketing improvements’ with cute names and smaller boxes. The GEO T was always small but produced SPLs comparable to the bigger boxes. Coupled with their cardioid design and cardioid CD18 subs, we still feel they are the best in their class, and so do the artists.”
GCS chose a Yamaha DME64n digital mixing engine to sum the three mixing consoles at front of house, one of which was a Yamaha PM5D-RH.
Each of the front of house consoles had control of left and right front fills, and aux to feed subs. A second Yamaha PM5D-RH was part of a trio of consoles used for monitors.
“The DME was used to sum left and right to feed out fills for the seating on the sides, and also for room EQ and time alignment,” says Gand. “Interfacing multiple console platforms has always been a challenge, however, the DME64n made using artist-specific consoles a snap, allowing each engineer to tailor various zone options to their own taste.”
Scottish Band GlasVegas Tours With Allen & Heath iLive-R72 Digital Mixing System
FOH engineer Steve Pattison uses an iDR-48 MixRack with the rackmount iLive-R72 and iLive MixPad app
Scottish band GlasVegas is currently on a global tour with a compact Allen & Heath iLive digital mixing system managing its FOH audio.
The band has already visited an assortment of venues across Europe and the USA, and is playing at many of the international summer festivals, including a headline spot on the John Peel stage at Glastonbury.
FOH engineer Steve Pattison has previously used various iLive systems for bands such as Alphabeat, Ellie Goulding and Royksopp. For this tour, he opted for an iDR-48 MixRack with the rackmount iLive-R72 supplemented by the iLive MixPad app, which provides remote and instant control of key faders.
“The R72 fits anywhere and I can carry it under my arm but that’s only part of the story,” Pattison explains. “To me, using only the basic functions of a desk is like having a gym membership and never going. There are a lot of tricks you can use to create a groove and a pulse. For instance, I am using a gate at 3dB across the bass, with a trigger placed on the kick drum so I get a 3dB boost when it is hit.
Pattison makes use of iLive’s numerous onboard FX for the drums, guitars, and backing vocals to create the band’s signature sound.
“I am using loads of reverb and the whole mix is quite ambient,” he adds. “The drummer plays standing up so I use the subharmonic synthesizer, which has a really nice low frequency tone, to get some good bottom end and a solid back beat for the song. I also have three guitar channels with FX: One is a pitch-shifted echo, a second has a panned delay, and the main guitar is set to jump to stereo so it punches through the middle. The FX returns are fully-featured so you can gate stuff and tweak channels as you wish.”
Pattison can recall each show file so there’s only minor tweaking from venue to venue, with the exception of the lead singer’s vocals.
“James Allen has chosen to use a low gain mic designed for snare drums in a studio environment, so I have to dial-in the gain higher than usual and pick up loads of other things onstage, like the snare drum. I use lots of EQ and compression and then feed it to a subgroup with a graphic and a parametric EQ. I even have an EQ for when he talks between songs because it gets a bit muddy. I am also using a big plate reverb and a stereo delay on his vocals. We have actually been accused of miming because it sounds so much like the album,” he concludes.
Allen & Heath Website
Wednesday, July 06, 2011
Crown Audio Calculators: Amplifier Power Required
A useful calculator for designing audio systems with Crown amplifiers.
This calculator provides the required electrical power (power output from the amplifier) to produce a desired Sound Pressure Level (SPL) at a given distance, along with an amount of headroom to keep the amplifier(s) out of clip.
Example: You are designing a system where the farthest listening position from the loudspeaker is 100 meters, and the desired Sound Pressure Level is 85 dB SPL.
The loudspeaker chosen for the job has a sensitivity rating of 95 dB. With the minimum recommended amplifier headroom of 3 dB, the you need to choose an amplifier that can supply at least 1,995 watts to the loudspeaker.
Equations used to calculate the data:
dBW = Lreq - Lsens + 20 * Log (D2/Dref) + HR
W = 10 to the power of (dBW / 10)
Lreq = required SPL at listener
Lsens = loudspeaker sensitivity (1W/1M)
D2 = loudspeaker-to-listener distance
Dref = reference distance
HR = desired amplifier headroom
dBW = ratio of power referenced to 1 watt
W = power required
Thursday, June 30, 2011
Creston Debuts New Docking Station For Original Apple iPad Or iPad2
The new charging docks turn the Apple mobile tablets into tabletop or wall mount Crestron touch screens for simple finger touch control of the home, classroom and building
Crestron today began shipping a stylish new line of docking stations for both the original iPad and the new, thinner iPad 2.
The new charging docks turn the wildly popular Apple mobile tablets into tabletop or wall mount Crestron touch screens for simple finger touch control of the home, classroom and building.
IDOC-PAD and IDOC-PAD2 enables any iPad or iPad 2 to be stationed and used while charging in a sleek, minimalist in-wall design, or in a slim ergonomic tabletop model.
Available in a gloss white or black finish, both models provide a clean, contemporary appearance for mounted use of iPad and iPad 2 in any environment. Simply press the slim frame on the wall mount model and the dock extends, inviting you to slide the amazing tablet in or out.
The curvy and lightweight tabletop model is designed for quick docking – and then swiping it off again – so you can move freely from room to room or across the office, keeping you as mobile as your iPad. There are no clumsy latches or annoying locks to slow you down.
“Whether you own the original iPad or the new iPad 2, there’s a Crestron mobile control solution to accommodate you,” said Crestron VP of Technology, Fred Bargetzi. “We’re committed to keeping our customers on the edge of cutting edge by offering the solutions that bring the latest, most popular technology into their lives. Our new IDOCs for iPad and iPad 2 are just the latest examples of our commitment.”
IDOC-PAD docking stations for iPad and iPad 2 are the latest additions to Crestron’s complete line of mobile solutions for Apple mobile devices. The best-selling Crestron Mobile Pro G app turns any iPad into a powerful Crestron or Prodigy touch screen, enabling reliable whole home and building control from anywhere in the world.
From any iPad or iPad 2, users can control lighting, thermostats, and audio settings and view streaming video in one or multiple locations such as a primary residence, vacation home, office, or across a campus.
The intuitive graphical interface also provides convenient real-time feedback so users can adjust settings as they deem necessary, no matter where you are. Room temperatures, shade positions and lighting levels, plus audio volumes and metadata including album, song and artist, are displayed right on the iPad screen.
Describing how Crestron consistently stays ahead of the curve when it comes to the latest new control technologies, Crestron Executive Vice President, Randy Klein said. “We deliver the products you want when you want them. We make it easy.”
Crestron Mobile solutions
Posted by Pro Sound Web on 06/30 at 11:45 AM