Interconnect

Thursday, September 15, 2011

Real World Testing: Matching Amplifier Data And Specifications

Click here to view/download a printable pdf of this article.

I’ve done enough theorizing and musing about power and power ratings in part 1 and part 2 of this series.

Let’s convert some utility power into audio power. First, some details about the testing.

The Source, The Load
“Source” and “load” are usually associated with amplifier and loudspeaker. I had to zoom out another level and consider the AC source to the amplifier (these amplifiers need a lot of juice). The amplifier has to get its power from somewhere, whether it be the utility power company, generator or a large array of hamsters on treadmills.

I added a new power circuit to the lab for the testing. The line voltage can be 120 volts RMS or 240 volts RMS. The circuit breaker is 30 amps. The electrical panel for the lab is near the amplifier rack, so line losses are small and tripped breakers can be easily reset.

The reason for all of the fuss is that when you are attempting to draw power that is at the limits of what can be safely sourced from the utility company, details that one might normally take for granted suddenly matter. The amplifiers need a hardy supply. Remember that when you design your power distro.

So now all I have to do is plug-in the amplifier, right? At these higher current ratings, there are several choices of power cord plug for both the electrical outlet and the amplifier inlet.

For the outlet, I chose a versatile, 4-conductor connector that can be easily wired for any voltage/grounding needed. The amplifier inlet connector was different for all three amplifiers tested. After a trip to Home Depot, and some rummaging through the parts bin, I fabricated a power cord for each.

Figure 1: Inside the testing - four 1,800-watt water heater elements (purely resistive) mounted into a 25-gallon steel tank.

The audio power must flow to something. The loudspeaker is the obvious candidate, but they get really loud at these drive levels and are expensive to replace. I mentioned a somewhat crude dummy load in part 1, but it had some shortcomings.

As before, when you’re pushing the envelope, the details matter. I decided to build a new load. Figure 1 shows “The Mother-Load,” a 4-channel by 8-ohm dummy load. There are four 1,800-watt water heater elements (purely resistive) that are mounted into a 25-gallon steel tank filled with environmentally-friendly mineral oil.

All of the tests in the article were performed driving the Mother-Load. Yes, I know the name is sexist, but “Father-Load” just doesn’t feel right.

A feature of the Mother-Load is switchable reactive circuits that can be added to each 8-ohm element. I lifted the schematic from the Rane Professional Audio Reference. It is shown in Figure 2.

Figure 2: The “Mother-Load” includes switchable reactive circuits that can be added to each 8-ohm element. (Credit: Rane)

Figure 3 shows the overlaid impedances of the Mother-Load configured at 4 ohms with and without the reactive circuit. I have also included the impedance curve of a real-world 4-ohm load to demonstrate that the average impedance of a typical loudspeaker is higher than its rated value.

Figure 3: “4 ohm” loads compared.

Why do I (perhaps) bore you with these details about power circuits and loads? Because they matter. If someone lays out a bunch of cash for an amplifier that they expect to provide thousands of watts of power, they had better think about where the power is coming from and where it is going to.

If you can’t source or dissipate the power, save some money and buy a smaller amplifier. The voltage and current from these amplifiers will challenge the Mother-Load. What about a “mere mortal” loudspeaker voice coil?

The Test Signals
The power from an amplifier is completely waveform-dependent. I tested these amplifiers using three signal types. Here are the signals and a few relevant attributes:

1) Continuous Sine Waves - simple, unambiguous, and worst case with regard to current draw and power dissipation. Easy to generate, easy to measure. I mentioned my affinity for power ratings based on continuous sine waves in part 1.

2) Pink Noise - more “real-world.” The crest factor and bandwidth are more “music-like,” yet there is still enough stability to allow a “one number” rating to be estimated from an analog meter or bar graph.

3) Tone Burst - music contains transient events that can have very high amplitude and very short duration. I used Don Keele’s Peak Tone-burst test for this purpose. The spectral content and duty cycle can be sculpted to emulate a kick drum, staccato bass guitar, or whatever, but in a way that is completely repeatable and measurable.

I also used Don’s Tri-Tone Burst Generator, a program that runs under Igor, to produce the tone bursts. A center frequency of 80 Hz was selected to emulate the spectral content of a kick drum. The tone burst includes 6.5 cycles of three sine waves spaced at 1/3-octave and centered at this frequency.

The Amplifiers
I selected three amplifiers to test. They all claim output power in excess of 1 kW, but there are significant differences between them in terms of ratings, features and price. The measured data is mainly useful for seeing how each performs under various signal types and load conditions. Here is a little background on each.

The first is a QSC PL380 that I use for loudspeaker power testing on a regular basis (typically in bridged-mono). It is analog by design. DSP processing and control is available, but I don’t have the required module, etc.

The second is a Crown I-Tech 9000 HD that I use in SynAudCon training classes. It has extensive on-board DSP and I monitored its performance with the company’s System Architect software.

The third is a Lab.gruppen PLM 20000Q that is on loan for evaluation. I used two of the four channels. While these models are apples, oranges and pears, they are all multikilowatt rated amplifiers and are at or near the top of the product line for each manufacturer.

Amplifiers have many power ratings, based on load impedance, number of channels driven, signal type, etc. Most manufacturers provide a ratings matrix for a sine wave signal. See Figure 4 for an example.

Figure 4: An example of a ratings matrix for a sine wave signal.(Credit: Crown Audio)

It gets even more complicated when you perform the testing for multiple signal types for both resistive and reactive loads. Since this is an article, not a book, I have only presented some of the results in the data matrix (Figure 5).

Figure 5: Some of the results of the testing process.

I monitored the current from the electrical circuit during the testing. The 30 amp breaker didn’t trip until about 40 amps. I should again emphasize that these amplifiers need a lot of utility power, along with special cabling and connectors to produce their rated sine wave wattages at 4 ohms.

I tested each amplifier’s open circuit output voltage as the first test. This basically means “disconnect the load and see what the amplifier’s output voltage is looking at air.” This is a useful reference for considering what happens under load.

An Open-Circuit Sine Wave Reference
None of the amplifiers had any problem maintaining their open circuit sine wave voltage into a 8-ohm load, even with two channels driven. All of them could produce their open-circuit voltage into 4 ohms, but fans were on “high.” Amplifier ventilation would be a definite issue for 4-ohm loads and low crest factor signals.

Pink Noise & Metering Issues
Pink noise is the most real-world of the test signals. It’s also the most difficult to measure accurately and refine to a single number. Sine wave clipping is trivially easy to observe on an oscilloscope.

Pink noise clipping has to be pretty severe to see it on a scope, so I had to rely on the metering of each amplifier. That’s where things become gray.

Consider the following:

1) Is clipping even possible? Most modern amplifiers have algorithms that prevent clipping. How fast does it kick in? How much “over” does the amplitude have to be before clipping suppression activates?

2) How often does the clip light have to flash before I declare that I have reached clipping? Once? Fairly often? Continuous illumination? There is some interpretation required here.

3) Once I have decided that I’ve reached clipping, what is the measured voltage? Even with an expensive True- RMS voltmeter with both digital readout and bar graph, it is a judgment call to assign a single number to something that has a 12 dB difference between its peak and RMS value. It’s another judgment call.

The clipping indicators of three different brands are bound to use different criteria for each of these. And no, you can’t detect the onset of clipping by listening.

In part 1, I showed how the peak program meter (PPM) is designed to ignore clipping that is likely inaudible (6-8 dB). For these and other reasons, I wouldn’t use the pink noise voltage and power ratings for comparing various brands of amplifiers. Errors of 3 dB or more could be expected, and in this power range that could be hundreds of watts.

As proof that all of this matters, just look at the open circuit voltage comparison between the sine wave and pink noise for each amplifier. The sine wave has a 3 dB crest factor and the pink noise a 12 dB crest factor. That’s a 9 dB differential.

Yet when I calculate the dB difference between sine and noise output for each amplifier, I only get the expected 9 dB for the QSC. That tells me that there is a difference in the metering of the three amplifiers.

If you look at the ratings matrix it appears that the QSC output voltage and power is significantly lower than the other two with pink noise, but this is obviously due at least in part to differences in metering. I could have easily produced a higher voltage from the QSC using a different visual criteria for clip. That’s why I hate “one number” ratings when taken at face value.

Things Amplifiers Hate
There are two conditions that none of these amplifiers like - low crest factor signals and 2-ohm loads. The continuous sine wave testing was by far the most revealing with regard to how much power the amplifier (and the electrical outlet) could source. I had to reset the circuit breaker numerous times during the sine wave testing into lower impedances with two channels driven.

I should mention that all of the amplifiers recovered nicely from circuit breaker trips. Once they powered back up they resumed working, and the PCcontrolled models re-established communications. All of the amplifiers have internal breakers, and none of them tripped during any of the tests.

None of these amplifiers could maintain their output voltage into 2 ohms with a sine wave source. The results varied so dramatically that I removed that data from the matrix. They sort of work at 2 ohms with pink noise, although I believe that the voltage limitations were due more to the amplifier’s “smarts” than due to their electrical current limitations.

The designers are trying to protect their amplifier and loudspeaker. They are trying to keep you from “running with scissors” when you insist on overloading your amplifier to get more watts.

None of the amplifiers cared if the load was resistive or reactive. That is a very good thing for a commercial power amplifier, and something that hi-fi amplifiers often have a problem with.

Burst Testing Results
These amplifiers get their largest power ratings from tone burst testing. Ironically, that was the easiest signal for all of the amplifiers to pass.

None of them had a problem producing their maximum voltage for a few tens of milliseconds, even into 2 ohms with both channels driven. I did trip the circuit breaker a few times at 2 ohms, so this test was really more about the utility power source than it was the amplifier. It’s a shame that this number wields so much influence in the marketplace.

The Truth About 2-Ohm Loads
If you want the maximum output voltage and most linear performance from your amplifier, don’t load it to 2 ohms. I believed this before doing these tests, and I believe it even more now.

You can see how each amplifier’s voltage sags under different loads in the matrix. Voltage sag is bad. I would bet money that anyone who is conducting amplifier shoot-outs for driving their 2-ohm monster sub is mainly listening to each amplifier’s protection circuitry. If you want fidelity, design your subs to 4 ohms or higher (that’s RATED impedance - actual will be higher yet) and buy more amplifiers.

An Important Test That I Omitted
While the data matrix is useful for assessing the performance of each amplifier, it is not the whole story. If you change the program material to music, and drive both channels into 4 ohms or less, there will likely be significant differences in the sound, due to the philosophical underpinnings of the operation of the protection algorithms.

The best way to assess the amplifier’s performance under these conditions may be the FTLC6P test. It requires some Favorite Tunes, a Lawn Chair, and a 6-Pack of your favorite beverage. While the results are subjective, they may the be most revealing test of all for comparing overloaded amplifiers.

The Inevitable Comparisons
While I have stated that this was not an amplifier shoot-out, people will inevitably make comparisons. That is why I suggested the dBW scale in part 1, as opposed to “watts” and have included it in the matrix for the sine wave ratings.

Differences of less than 1 dB are negligible. A 3 dB difference may be audible under controlled conditions. The largest dB differences were for pink noise, which I showed above to be the most difficult signal to characterize with a single number. When we are looking at amplifiers rated at over 1 kW, the relative differences will be small.

Conclusion
The testing validated some important points from parts 1 and 2. Here’s a quick summary:

1) A 1 kHz sine wave rating into 8 ohms is the best measure of an amplifier’s performance for a sound system designer. The amplifier acts as a constant voltage source into 8 ohms. The voltage can be scaled to any crest factor by calculation. This makes accurate sound pressure level calculations at the drawing board possible.

2) Don’t load your amplifiers to 2 ohms to “get more watts.” When the voltage sags, you are losing output level and engaging protection algorithms.

3) Amplifiers don’t like low crest factor signals. If you excessively compress or limit the program material, and drive the amplifier to clipping, you are likely engaging the amplifier’s protection algorithms, with significantly audible ramifications.

4) Don’t compare amplifiers using power ratings derived from burst testing. These are vanity specs that look impressive but reveal very little about the amplifier’s performance.

5) Summing the instantaneous power ratings of all channels and thinking that you can get that much “power” is fantasy. Doing so is misleading at best and it promotes erroneous thinking and bad behavior in the marketplace. It’s putting a modern face on the peak power rating wars of the 1970s.

I was truly impressed in different ways by each amplifier that I tested. The output capability and efficiency given their size and weight is a true miracle of modern technology. It’s my hope that this series of articles sheds some light on how modern amplifiers operate and how their specifications are determined. This should lead to better correlation between the predicted and actual performance of each in real-world sound systems.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world in addition to providing web-based training at www.synaudcon.com.

Click here to view/download a printable pdf of this article.

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Posted by Keith Clark on 09/15 at 08:42 AM
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Line 6 Announces New Mobile In Digital Input Adaptor & Mobile POD App

Turns iOS devices into portable Line 6 POD multi-effect processors

The new Line 6 Mobile In digital input adaptor and Mobile POD guitar tone app allows guitarists to connect an electric guitar to their iPhone or iPad.

The new Mobile POD app brings Line 6 tone and technology to its most portable level. 

The free app includes generous collections of 64 models of celebrated modern, vintage and boutique guitar gear including amps, cabinets, stomp boxes and rack effects. 

Users can mix and match the fully adjustable models to build a virtually unlimited variety of tones, which can then be saved as presets. 

Also included with the app, free of any in-app purchase, are over 10,000 hand-crafted presets created by artists, Line 6 and other guitar players.

Unlike analog adaptors that plug into an iOS device’s headphone jack, Mobile In digital input plugs into the 30-pin connector on the bottom of the iOS device. This digital connection enables the adaptor to offer pro-quality audio specs for guitar tones that are exceptionally rich and clear. 

Mobile In supports up to 24-bit/48 kHz digital audio. The guitar input offers 110 dB dynamic range. The stereo Llne input, which can be used for sending keyboard audio or any other standard mono or stereo line-level audio source to an iPhone or iPad, has 98 dB dynamic range. 

Mobile In can also serve as a digital input adaptor for use with any CoreAudio guitar apps, including GarageBand for iPad.

Mobile In comes with a quality 6-foot guitar cable, and the Mobile POD is a free download from the App Store. Mobile In and Mobile POD are made for iPhone 4, iPad 2 and iPad.Find out more about them here

“Mobile In and Mobile POD turn your iPhone or iPad into a fully functioning Line 6 POD multi-effect device complete with tweakable tones, simultaneous effects and legendary Line 6 guitar amp and effect modeling,” states Line 6 co-founder and SVP of new business development Marcus Ryle. “Line 6 pioneered digital modeling technology and, in 1998, released the first POD multi-effect processor. This introduced countless guitarists to digital modeling, a technology that gives them access to wide varieties of amp and effect tones within one device.

“Mobile In is designed to be the best possible guitar tone solution for mobile devices,” he continues. “Other iOS guitar adaptors are rife with issues like headphone feedback, latency and poor sound quality. Mobile In solves all of these issues to deliver the best iOS guitar playing experience.”

Mobile In and Mobile POD ship fall 2011. MSRP is $79.99.

Line 6

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Posted by Keith Clark on 09/15 at 06:22 AM
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Tuesday, September 13, 2011

Church Sound: Developing An “Easy Button” For A Student/Community Room PA

The whole system ran a little over $1,500, including controllers, and it took me about an hour to set up all the programming
This article is provided by ChurchTechArts.

 
Last week I was able to finally deploy the new EZ Mode (our term) for our student/community room PA.

The room is our most-used room every week, supporting everything from Jr. & Sr. High to men’s and women’s ministry to MOPS. There is something happening in that room every day of the week.

The system is relatively simple, but I’ve received more than one call at home on a Tuesday night from someone trying to make a mic work, or get sound out of the iMac.

I began looking for a solution and found it in the Symetrix Jupiter 8 processor and a set of associated wall controls, the ARC2 and the ARC-SWK. I’ll write up a review of the Jupiter in a separate post. For this one, I’ll focus on how I used the wall controls to create an EZ Mode.

It All Starts With Planning
My first step in the process was to sketch out what I wanted to accomplish. Based on the way the room is used, we really needed two major modes of operation, with two subsets each.

Big picture, it looks like this:

—Mix Mode, Delays On
—Mix Mode, Delays Off
—EZ Mode, Delays On
—EZ Mode, Delays Off

The room is a classic “two rooms in one with an air wall” layout; meaning sometimes we need the whole room, other times, just half. The main PA (hung over the stage) is the Electro-Voice LiveX 15s and subs I’ve written about before. We also hung some EAW JF80s (because we had them lying around) as delay loudspeakers to fill in the back half of the room on the other side of the air wall.

Student ministries runs a full band for their events, so they need to be able to mix a full compliment of inputs on the Yamaha MG32 we have in there. Most of the other ministries/events however, require one or two mics, audio for video (either DVD or the iMac) and an iPod input.

image

Since the Jupiter 8 has 8 inputs and 8 outputs, and the inputs can be either mic or line, I set about making up a plan.

The input side looks like this:

—1&2 Stereo In from the MG32
—3&4 Wireless Mics 1&2 (also double patched into the MG32)
—5&6 Audio from Video (will eventually be the audio output of an Extron IN1508, for now, it’s a double-patched ProAV2)
—7&8 A dedicated “EZ” iPod cable.

Outputs 1-4 feed the main speakers (L&R plus L&R Subs), while Output 5 feeds the delays. Output 6 feeds nothing, Output 7 feeds a CD recorder, while Output 8 is used to control the logic output that turns on and off the delay speaker’s amp. More on that later.

Basic Programming
The first thing we did was to get a baseline layout in all the DSP.

Inputs and Outputs were labeled, patched routed and gained.

We set up our crossovers for the main speakers and dialed in system EQ. We set up the delays and got their EQ where we wanted it.

That formed the basis of our programming.

The next step was to build presets that turn inputs and outputs on and off.

Preset 1 is Mix, Delay Off, so we muted inputs 3-8, the delay output and turned off the delay amp. Preset 2 (Mix, Delay On) was created by un-muting the delay output and turning the amp on.

Preset 3 is EZ, Delays Off. To create this preset, inputs 1&2 are muted, meaning the output of the MG32 is completely ignored by the Jupiter.

We unmute inputs 3-8, which enable the two mics, video and the iPod cable. As with Preset 1, the delays are off. Preset 4 is like Preset 3, only with the delays on.

Once that was all set up, tested and found to be working, we hooked up the two controllers.

Controlling

Each menu in the ARC2 can control volumes, a mute button or change presets.

The ARC2 is a menu driven controller. It’s extremely powerful and enables you to control quite a few parameters inside the Jupiter. I could have done everything I needed with this box, but figured the addition of an ARC-SWK would make the system easier to use.

In my setup, the ARC2 does one thing - enable users to switch between the four operating modes.

Anything that the user sees can be edited easily.

This is accomplished in the Jupiter software; simply add a controller, create a menu, and load the presets. You can edit the labels to make it easy to navigate.

Once it’s all assigned, you can simulate the controller to visually ensure it’s all working the way it’s supposed to.

You can quickly and easily create menus to control just about every volume and mute parameter in the Jupiter. Not to mention switch presets.

The next step is to add in the ARC-SWK controller. The SWK is a 4-button, single encoder remote with an A and B side. This means you can control up to 8 parameters very easily.

Here is how ours is set up:

—Button 1A: Wireless Mic 1
—Button 2A: Wireless Mic 2
—Button 3A: Stereo Audio for Video
—Button 4A: Stereo Audio for EZ iPod Cable

The ARC-SWK in software simulation mode.

The software makes it easy to control inputs and outputs as mono or stereo channels. At the moment, I don’t have any need to control anything else, though I have the capability to control four more parameters if need be.

When controlling volume, you can specify minimum and maximum values - initial values are stored in the preset. It’s all very easy to do, and took less than 5 minutes to assign everything.

The ARC series of remotes connect daisy-chain style to the Jupiter over Cat5. You can also send audio through certain wall panels, either in or out, depending on the model.

Final Programming
What sold me on the Jupiter is the calendar feature. Once all the presets are built, you can create events (single or repeating) that will automatically switch modes.

So in our case, on Tuesday morning, the system goes into EZ, Delays On at 8:45 AM for the Women’s Bible Study. At 3:00 PM on Wednesday, it switches to Mix, Delay Off for Jr. High. On Thursday at 8:30, it switches back to EZ, Delay On for MOPS.

Eventually, I will add the Jupiter to the network so I can access it from anywhere and create custom events (like next Friday when the Boy Scouts use the room).

After all that, I dove into the logic outputs.

The Jupiter has 4 dual-mode logic outputs. Each logic output can deliver 5 VDC for connecting an LED indicator, or act as a simple contact closure (alternately, you can just use the +5VDC to close an externally powered relay).

The logic outputs are assigned to parameters anywhere in the system.

In our case, to make programming easy, I assigned the control to Output 7, which we weren’t using anyway.

When Out 7 is muted, the delay amp turns on (using a Furman Relay). When it’s unmuted, the amp turns off. The two mute states get saved into presets, and just like that, the amp turns on and off as if by magic.

The Cost
Doing something like this used to require a Crestron or AMX system, and programming could easily run into the thousands of dollars, not to mention the additional costs every time you wanted to make a change or the equipment cost. In this case, the whole system ran a little over $1,500, including controllers.

It took me about an hour to set up all the programming (and another few hours to tune the system).

It’s easy enough to use that any TD will be able to get the system doing whatever they want in no time. I was able to train our entire staff on the EZ Mode of operation in about 10 minutes, and the documentation takes just two pages (and half of each page is a large picture of each controller).

Best of all, I won’t have to take any more calls like this one…

Caller: Mike, the mics aren’t working.

Me: OK, so go to the fader labeled RF-A. Make sure the button above that fader is turned on and lit up.

Caller: [Long pause, obviously frustrated] I…I…don’t even know what a fader is.

Mike Sessler is the technical director at Coast Hills Community Church in Aliso Viejo, CA and serves as the Church Sound Editor for Live Sound International. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts. Mike also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

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Posted by Keith Clark on 09/13 at 04:53 PM
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Crown Audio Introduces PIP-BLU Programmable Input Processor Card For CTs Series Amplifiers

Adds BSS Audio Soundweb London digital audio bus functionality to CTs amplifiers

At PLASA 2011, Crown Audio announced the introduction of its PIP-BLU Programmable Input Processor (PIP) card for the Crown CTs Series power amplifiers.

The PIP-BLU adds BSS Audio Soundweb London digital audio bus functionality to CTs Series amplifiers, offering a simple, cost-effective way to get digital audio all the way to the amplifiers.

“The PIP-BLU adds BSS Audio’s Soundweb London digital audio bus to Crown’s popular CTs Series amplifiers, without having to use additional network switches. This makes installation much easier, saves time and money, greatly simplifies the cabling within an amplifier rack and provides advanced control and monitoring capabilities,” said Daniel Saenz, install business segment manager, Crown Audio. “With the PIP-BLU, contractors are now able to leverage the functionality and simplicity of the Soundweb London digital audio bus all the way to the amplifiers.”

The PIP-BLU is compatible with Crown CTs Series 2-channel amplifiers built after April 2011. It provides BSS Audio Soundweb London digital audio bus functionality along with analog input and output connectors.

It also offers two digital audio bus channels to the amplifier, an RJ-45 Ethernet jack for remote configuration, control and monitoring and two digital audio bus channels back from the amp (for the listen bus), along with two analog input channels and two analog output channels.

The analog channels provide analog backup and also facilitate using the CTs amplifier as a 2-channel Break-In Box (BIB). The analog outputs provide a buffered version of the inputs, allowing daisy-chaining down a rack.

The PIP-BLU card allows the amplifier to be connected into the Soundweb London digital audio bus ring, providing the sharing of digital audio, fault tolerance and greatly simplifies cabling. The PIP-BLU uses Harman HiQnet protocol and is configured, controlled and monitored using HiQnet London Architect.

Crown Audio

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Posted by Keith Clark on 09/13 at 09:50 AM
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Monday, September 12, 2011

Engineer Jim Warren Utilizing Avid VENUE For Current Arcade Fire Tour

“Everything I’ve done with this console, I’ve been able to do with plug-ins. It keeps the workflow very focused." - Jim Warren

Jim Warren, a veteran front of house engineer has been at the helm for some of rock’s most renowned artists, including Peter Gabriel, Radiohead, Nine Inch Nails, Crowded House and Duran Duran, to name but a few, is presently on tour with recent Grammy winners Arcade Fire, manning the faders of an Avid VENUE system with D-Show console, FOH and Stage Racks.

“I’ve dabbled with a number of digital consoles over the years,” says Warren, “and they all had things about them that I found difficult to deal with in one way or another. Even the VENUE did when it first came out – there were certainly a few teething problems.

“But I was very impressed with how willing the Avid folks were to listen to people’s comments, and how quickly they responded to them,” he adds. “In many cases, they were already working on issues before people had discovered and reported them.”

Working with a number of artists whose arrangements can be lush and complex, Warren is no stranger to the demands of large channel counts.

“With many tours it had gotten to the point where, even with larger analog consoles, I’d have to have a sidecar or two to handle the number of channels I was using,” he reports. “When it gets to the point where you have to walk from one end of the console to the other, having layers with multiple channels becomes far more convenient. I think one of the most important things for me about the VENUE system was the ability to handle a large number of channels in a relatively small footprint.”

VENUE’s processing power also contributes to its small footprint, says Warren, replacing racks of outboard gear with a wide range of plug-in processing. “VENUE is the first digital console I’ve used where I don’t need to use any outboard gear,” he says. “Everything I’ve done with this console, I’ve been able to do with plug-ins. It keeps the workflow very focused. I like being able to turn up at the gig, turn the desk on and have everything there at my fingertips.”

“The other point about onboard processing is that it keeps me from compromising,” he adds. “When you start getting into higher channel counts, you get to the point where you decide, ‘well, do I really want to carry a whole rack of compressors if I’m only going to use them on one or two songs?’ But with the VENUE, you’ve got them there when you need them, without adding any additional weight or complexity.”

Snapshot automation is one of the features of VENUE that has made a tremendous difference in the way Warren works. “Having lots of channels and lots of songs, snapshots became more than a luxury,” he observes. “With an analog desk, I found myself getting to the point where I was still making changes 30 seconds or so into the next song. When you’re working with a band whose playlist is six feet long, and the order’s never the same from one show to the next, it becomes next to impossible to remember every little tweak and change. Recalling as many changes as possible with the push of a button means you only have to worry about the fine adjustments, rather than every detail of each song’s mix.”

Warren also cites VENUE’s Virtual Soundcheck as a key feature. “We’ve just gone through a whole summer of doing festivals where often it’s impractical or even impossible for the band to do a sound check,” he says. “Even when there is time, if you’ve ever tried to get a band to play the same 15 seconds of a song so you can nail the change from one section to another, it’s very difficult to do. But with Virtual Soundcheck, I can put a Pro Tools recording on a loop and accomplish in ten minutes what might be a month’s worth of trial and error during a performance. It tends to make you a little bolder about trying new stuff out, because you’re not running the risk of potentially wrecking someone’s show.”

Avid

 

 

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Posted by Keith Clark on 09/12 at 03:21 PM
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Finding The Optimum Playback Level Of Your Sound Card

A common question is “Where do I set the output level of my PC for optimum results?”

The personal computer is frequently used as a program source for sound reinforcement systems.

Internal sound cards can be pretty good and software wave file players abound.

A common question is “Where do I set the output level of my PC for optimum results?”

The answer can vary per PC, so here are some steps to help figure it out for yours.

I will assume that WAV files are being used, but this applies to any audio file format.

Sound on a PC can be broken down into two parts:

  1. The WAV file itself.
  2. The digital-to-analog converter DAC and output circuitry.

To find your optimum output level setting, you first need a test waveform that represents the maximum level that a WAV file can have. A full-scale sine wave will do nicely.

The best way to come up with one is to simply create it from scratch. Any WAV editor should be able to do it, but I will describe the steps used in the free ware Audacity editor:

  1. Open Audacity
  2. Select Generate/Tone
  3. Select “Sine” as the waveform type and enter the desired attributes. I suggest Frequency = 440Hz, Amplitude = 1.0, Time = 10sec.
  4. Hit “Okay.” See Figure 1 for a zoomed-in look

Figure 1 - A full-scale sine wave.

If you do this in a different editor, an additional step may be necessary. The WAV must be normalized, an option usually found under the Process or Effects menu.

Normalization makes the waveform full-scale – the highest possible undistorted amplitude.

The objective is to determine the volume control setting that produces the highest undistorted playback level of this file. This will also be the correct setting for any full-scale WAV file.

Following are the steps to determine the maximum setting of your PC’s level control. I will assume that there is a speaker icon in your system tray as shown below (Figure 2).

Figure 2

If not, you can enable this under Control Panel/Sound and Audio Devices (Figure 3).

Figure 3- Audio Properties Dialog

Here we go:

  1. Turn the volume all the way down.
  2. Play the WAV file. Holding Shift while clicking Play will loop the file in Audacity.
  3. Advance the level control until you hear the tone.
  4. Keep turning it up until you hear the onset of harmonic distortion (a distinct buzzing sound).
  5. Back it off until the distortion disappears.

Note the setting of the control. This is the maximum level for undistorted playback of a full-scale WAV file. On most PCs it is about three-fourths of the slider travel (Figure 4).

Figure 4 - Typical optimum setting.

If you are using an outboard USB audio interface, the steps are the same. Note that headphones can be substituted for the PCs speakers if necessary (Careful, it’s going to be LOUD!).

So, why do they make it (the level control) go further? Some WAV files are not full-scale, either intentionally or due to poor recording practices.

In these cases the slider may need to be a maximum to get sufficient playback level to the next device in the chain.

The last step is to make sure that whatever you are driving with the signal is okay with your PCs maximum output level. Most mixers have input level indicators for this purpose.

Make sure that nothing is “in the red” on the input channel you are driving.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world in addition to providing web-based training at http://www.synaudcon.com.

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Posted by Keith Clark on 09/12 at 05:36 AM
AVFeatureAVInterconnectNetworkingProcessorSignalSoftwareAudioPermalink

Friday, September 09, 2011

Allen & Heath Launches 64-Channel Waves SoundGrid Interface For iLive Series

New M-Waves module opens up the opportunity to use Waves SoundGrid plug-in processing and concurrent multi-track playback and recording

Allen & Heath has launched a Waves SoundGrid option module for the iLive digital mixing series, which provides a low latency 64-channel bi-directional interface to SoundGrid.

The new M-Waves module opens up the opportunity to use Waves SoundGrid plug-in processing and concurrent multi-track playback and recording.

Developed by Waves, SoundGrid is an Audio-over-Ethernet networking and processing technology for live applications. It provides an extremely low latency environment for high precision Waves audio processing using standard Gigabit Ethernet networks and an Intel-based server running a customised version of Linux.

A separate standard Windows or Mac computer runs the MultiRack SoundGrid control application and GUI functions. An integrated Ethernet switch with two EtherCon and one RJ45 SoundGrid port is provided on the M-Waves module, allowing direct connection of the SoundGrid Server and a PC.

Signals from the mixer can be routed to the M-Waves card for processing and recording, including any bus or Mix Channel (pre or post fade), Input Direct Out, Wedge and IEM monitors, MixRack Inputs, and Input or Mix Insert Sends. Signals from SoundGrid can be routed to Input Channels, Mix External Inputs, Insert Returns, and Output sockets.

M-Waves will be incorporated into Allen & Heath’s range of audio networking cards, which allow iLive to interface with a variety of common audio networking standards - such as MADI, Dante, EtherSound and Aviom.

The cards fit in the expansion slot of iLive MixRacks, comprising the modular iDR0 and iDR10, and fixed format iDR-16, iDR-32, iDR-48, iDR-64 and xDR-16 expander.

The Allen & Heath iLive digital mixing series comprises several Control Surface and MixRack variants, which can be mixed and matched in any combination and share the same firmware, so that show files are transferable between systems via a USB key. They connect together with CAT5 cable and use the Ethernet protocol for control.

All MixRacks feature the same 64x32 RackExtra DSP mix engine architecture, providing processing for 64 channels, 32 mixes, and 8 stereo FX processors.

Allen & Heath

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Posted by Keith Clark on 09/09 at 06:21 AM
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Thursday, September 08, 2011

Studer Unveils Range Of New Remote Stageboxes For Vista Consoles

Both offer a more fixed architecture

Responding to customer requests for lower-budget or simpler connectivity options on the range of Vista mixing consoles, Studer has supplemented its flexible and expandable D21m remote stagebox system with two new models, both with a more fixed architecture.

The new Vi Stagebox provides a 6U solution including 64 analog mic/line input XLRs and 32 analogue line output XLRs, with the option to replace any of the 8-channel cards with an alternative format such as AES, CobraNet, Aviom or EtherSound.

The Vi Stagebox also includes eight channels of GPIO with relay-controlled outputs and features redundant power supplies as standard. A 64 x 64 MADI HD link is fitted as standard to connect to the console’s SCore DSP rack, and multiple Vi stageboxes can be connected to provide a larger pool of I/O.

The new Compact Stagebox provides a very cost-effective solution in just 4U, offering as standard 32 analog inputs and 16 analogue outputs (but can be ordered in other configurations), together with eight channels of GPIO and two standard single D21m option card slots to add SDI, ADAT connectivity, etc.

The Compact Stagebox also links to the console via a MADI optical connection. The addition of the D21m slots provides an excellent method of deriving a multichannel recording feed via MADI.

The existing Studer D21m I/O remote stagebox system is based on card frames, each having the capability of accepting a wide range of different audio format cards, and multiple stageboxes can be easily distributed around large facilities

Studer

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Posted by Keith Clark on 09/08 at 11:23 AM
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Hosa Technology Unveils New Pro Speaker Cable Line

Both Loudspeaker and 1/4-inch TS connectors are available in the line

Hosa Technology has announced the introduction of the new Pro Speaker Cables line, combining high-quality cable for enhanced signal transmission and audio quality with REAN connectors by Neutrik AG, available at a midline price.

All cables feature 14 AWG Oxygen-Free Copper (OFC) conductors for enhanced signal clarity and a black PVC jacket for durability, flexibility, and low visibility on stage.

Both Loudspeaker and 1/4-inch TS connectors are available in the product line.

REAN Loudspeaker connectors by Neutrik incorporate silver-plated contacts for superior signal transfer, a glass-reinforced housing for reliability, a robust, twist-lock mating system for secure connectivity, and chuck-type strain relief for maximum cable retention.

The 1/4-inch TS connectors include nickel-plated contacts for efficient signal transfer and rugged durability, a zinc die-cast housing to enhance reliability, crimp-type strain relief for larger-diameter cable, and rubber boot kink protection to ensure long cable life.

“Hosa Pro Speaker Cables are engineered to deliver years of rugged, dependable performance,” notes Jonathan Pusey, Hosa Technology director of sales and marketing. “By combining REAN connectors by Neutrik AG with world-class manufacturing techniques, these cables deliver unsurpassed performance and value. I’m absolutely confident these cables will be very well received by all who audition them.”

Hosa Pro Speaker Cables are available in 3-, 5-, 10-, 25-, 50-, and 100-foot lengths and will be available in Q4, 2011. Configurations include Loudspeaker to Loudspeaker, Loudspeaker to 1/4-inch TS, and 1/4-inch TS to 1/4-inch TS.

MSRP pricing ranges from $14.40 to $169.20.

Hosa Technology

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Posted by Keith Clark on 09/08 at 06:27 AM
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Tuesday, September 06, 2011

Gepco Launches New Low-Smoke, Zero-Halogen Cabling Solutions For European Market

Deliver exceptional quality while complying with IEC and RoHS standards

Gepco International has announced that it has re-engineered a line of its audio and video cables in response to the increasing demand from the European marketplace for low-smoke, zero-halogen (LSZH) cables where safety is critical in the event of a fire.(The company is exhibiting at IBC 2011, stand 9.B02)

Gepco LSZH cables deliver exceptional quality while complying with IEC and RoHS standards.

In the event of a fire, a building’s electrical wiring can act as a vehicle to propagate the fire hazard from area to area. The fire damages account for a high level of replacement costs each and every year, associated with both structural replacements and the provision of damaged equipment and cabling, particularly since conventional cable manufacturing materials, such as PVC, exhibit burning characteristics that produce dense black smoke and harmful halogens. These halogens—chlorine, fluorine, bromine and iodine—result in corrosive acid when they come in contact with water. 

“Gepco LSZH cables are the best choice for any audio or video application where smoke might build and come into contact with people or equipment,” says Joe Zajac, General Cable market development manager of Gepco brand products. “This product line is a great example of turning end-user feedback into reality, and we will work to continue to develop other LSZH cable options.”

The materials used by Gepco in their range of cables are halogen-free, and when subjected to flames, will emit low levels of smoke. These features help ensure that a safe evacuation of people may be undertaken, and that exit routes remain visible at all times during a fire.

Maintaining visibility also allows firefighters to reach the source of the flames quickly and efficiently, therefore, ensuring that systems and equipment can be brought under control and shutdown with minimal delay.

“Our Gepco Brand LSZH cables are jacketed using advanced compounds that offer flame resistance, low-smoke production and reduced toxicity,” notes Brad Pope, General Cable director of technical services, Gepco brand products. “With fewer toxic chemicals, our cables offer a reduced environmental impact and are easier to dispose of than other forms of cabling.”

The redesigned LSZH cables can be used with the same connectors as other Gepco brand cables and meet the following flame performance standards:
—IEC 60332-3-24 Flame Propagation
—IEC 61034-1, -2, 2005-4 Smoke Emission
—IEC 60754-2 Corrosivity and Acid Gas Emission

Currently, the Gepco LSZH offering is comprised of seven solutions, including:

Four cables for video applications – Low Loss RG6 High-Definition SDI Coax (VSD2001LS), Standard RG59 High-Definition SDI Coax (VPM2000LS), Miniature High-Definition SDI Coax (VDM230LS), and Extended-Distance RG11 High-Definition SDI Coax (VHD1100LS)

Three cables for audio applications – Standard Easy-Strip Single-Pair (61801EZLS), Thin Profile 110Ω Single-Pair (DS601LS), and Wide Bandwidth 110Ω Single-Pair (DS401LS)

For more information about our Gepco LSZH solutions and other Gepco® Brand, SheerWire™ and General Cable products, please visit us at IBC2011 September 9-13 in Amsterdam at Stand 9.B02 or click here for our detailed Product Bulletin.

Gepco International

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Posted by Keith Clark on 09/06 at 01:50 PM
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Monday, September 05, 2011

The Old Soundman: Club Restrictions

Too many boneheads running the board?

Here’s one of those situations that make you wonder about your career choice or where you are in your life path.

Pay close attention, our buddy Brian is showing us how to keep the disgustedness in check and not resort to a brick through the front window of this fine establishment…

Dear OSM:

O.K., try this one out…

Hit me with it, Bri! Let me have it!

You just found out the band you regularly mix for has a gig at a “new” or “never played there before” club…

Surely this is not an unknown experience for you.

So you lock out the night for the gig, then the band calls back and says “uh, the club guy says ‘no outside soundman touches the board’ but you can stand next to him and assist.”

Ah, that’s brutal, Brian! I can see why you’re ticked off. But don’t freak out if I tell you that this is exactly what happens if you and your band go on Conan or Letterman or “The Tonight Show” or any of the 99,000 awards shows.

So in a weird way, what you’re faced with is good training for the big time! Although those broadcast mixers usually have a conscience and spend a little time studying the record.

I’m actually going to have my own awards show next year! It’s going to be called “The People’s Radio Scene Superstar Vibe-A-Thon For Players and Soundpeople.”

All of the servile tools-of-the-manufacturers audio mags are going to cover it, and my co-hosts will be Ann Wilson of Heart, Martha Davis of the Motels and the new chick from Evanescence.

I’m pretty sure she has a “thing” for me! (But don’t tell the Old Soundwoman.)

And I reply, “Did you mention to the club I’m a ‘professional’ and do this for a living, know the band’s material backward and forward, and have special cues for each song?”

Of course your pals did! Didn’t they?

They reply, “Sorry, we get too many boneheads running the board and screwing things up.” (Gee thanks, boneheads.)

Yeah, thanks a lot, boneheads!

So at the gig, I’m supposed to tell the house guy, “O.K., on this next chorus, hit the lead vocal with a 360 ms delay to trail off on his last note, then a big snare hit, followed by a guitar solo… ?”

May I make a suggestion, Brian? Go to this club as a customer one night, and strike up a conversation with the soundman.

Tell him exactly who you are. Have a couple beers with the guy, and tell each other some tales of the soundman life.

Of course, if the club is far from your home, this may not be practical. But if it’s nearby, go ahead and do your best to make friends with this individual who you’re busy demonizing, just as he is demonizing you.

Because, really, we all know he has a point – there are so many boneheads out there running around ruining sonic life for everyone within earshot of their ham-handed hijinks.

But – he is taking it pretty far. After all, he’s not controlling a major network program going out to millions of people every night.

Ahh, forget it – I’d rather stay home and watch reruns of “The Twilight Zone.”

Brian W

Can I come over and watch with you? How about the one with William Shatner as the nut who sees the ape out on the wing of the old airliner?

Yeah, you know exactly what I’m talkin’ about! You’ve now established yourself as a soundman of great taste and discernment.

I’m sure this is only a tiny, momentary stumbling block in your rampage to greatness!

Luv –

The Old Soundman

There’s simply no denying the love. Read more from the Old Soundman here.

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Posted by admin on 09/05 at 10:37 AM
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Friday, September 02, 2011

A Look At Handy, Inexpensive Mini Stage Snakes & Related Tools

The design goals were thus: create a 4-channel audio snake with an Ethercon jack to carry the M-48 signal as well
This article is provided by ChurchTechArts.

 
This isn’t the first time I’ve built a snake like this, nor is it the first time I’ve written about it.

But we’ve made a few updates, so I thought I’d write a new article on them.

This particular snake in question is what we use for our percussion or winds player, though sometimes it gets dragged over to the guitar position depending on the weekend.

The design goals were thus: create a 4-channel audio snake with an Ethercon jack to carry the M-48 signal as well. We wanted it all loomed together neatly so it would be easy to deploy and pick up. Here’s how we did it.

Middle Atlantic UCP Series
The box is built around the Middle Atlantic UCP Univ6 plate. The UCP Series is an incredibly useful modular series of plates designed to go in a rack mount rail system. You can put 5 plates between the rails and build pretty much any kind of patch panel you want.

The UCP Series has everything from Neutrik D-style to DB-style knockouts to various sized holes to fans and power switches.

They also make a UCP Box Adapter, which cleverly lets you mount a UCP plate into a 4-inch square box. With this simple adapter, you can quickly make up a 4-inch snake end with up to 6 connectors of your choice. Speaking of connectors…

Standard Neutrik connectors.

Neutrik D-Series
I use the Neutrik NC3FB-1-B XLR panel mount connectors for most of these boxes because they’re black and blend in well.

We either rivet or screw the connectors to the plate, depending on whether we have rivets or screws on hand. If you use rivets, make sure to put a washer on the backside of them.

We also dropped in a NE8FDV-Y110-B Ethercon connector. This is a standard panel mount connector with 110 contacts on the back for the Cat5 cable. Finally, we put a blank panel on the sixth port because we didn’t have anything to put there.

Cable
We used a 4-channel snake that we had lying around, cutting off the old Switchcraft ends to go into the box. For the Cat5, we used Gepco Tactical Cat5, which is super tough (and super-tough to strip—you’ve been warned). All of our stage Cat5 is Gepco Tactical, and we’ve yet to have an issue with it.

Standard Neutrik connectors.

Boxes
The boxes are really hard to come by. We just go to Home Depot and buy 4-inch square boxes and 3/4-inch cable clamps. Actually, we would have used 3/4-inch clamps if had some, but we used 1/2-inch instead. Either is fine.

About 6-8 coats of black spray paint gets them nice and black; however if you know of someone who can powder coat, that would be better. The paint comes off after a while.

Loom
I’ve used various types of loom in the past, but I’ve found my new favorite. It’s made by Techflex and we buy ours through CableOrganizer.com.

What’s great about F6 is that it’s side-entry, meaning it’s split down the length so all you have to do is open it up, slip the cables in and you’re done. If you’ve ever tried to shove 50 feet of cable through regular braided sleeving, you’ll be amazed at how easy this is. Pro Tip: Buy the $4 insertion tool. You can thank me later. A little bit of heat shrink tubing (also from CableOrganizer) cleans up the ends.

That’s about it. One thing we had to keep in mind is how much non-loomed cable to have at the fan end.

For us, we needed about 3-4 feet of separate cable to get the XLR ends of the snake into our stage snake and the Ethercon into IEM rack. Your mileage may vary, depending on how you need to configure it.

Total cost for this set up will vary depending on length and what type of cable you use, but it will be less than what you’d pay from a custom cable shop. Unless you don’t like to solder. In which case, find someone who does.


Mike Sessler is the technical director at Coast Hills Community Church in Aliso Viejo, CA and serves as the Church Sound Editor for Live Sound International. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts. Mike also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

 

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Posted by Keith Clark on 09/02 at 01:57 PM
Church SoundFeaturePollInterconnectMonitoringStage • (3) CommentsPermalink

Thursday, September 01, 2011

Complete Line Of Extron MTPX Plus 6400 Twisted Pair Matrix Switchers Now Available

Incorporate the existing features of the MTPX Plus Series with new advanced features

Extron Electronics has announced that the entire line of MTPX Plus 6400 Series twisted pair matrix switchers for RGBHV, HD component video, standard definition video, audio, and RS-232, is now shipping.

These switchers are available in five different I/O sizes, from 48 x 48 to 64 x 64. They incorporate the existing features of the MTPX Plus Series with new advanced features, including EDID Minder for local inputs and outputs, RS-232 insertion from the Ethernet port to all MTP outputs, and switchable video pre-peaking on all outputs.

“With the key technology advancements offered in these larger matrix switchers, AV integrators can take full advantage of the benefits offered by twisted pair signal distribution,” says Casey Hall, vice president of sales and marketing for Extron. “The MTPX Plus 6400 Series makes system integration easier, eliminating many of the headaches normally encountered with other twisted pair solutions.”

The MTPX Plus 6400 Series is capable of switching local and remote AV signals to multiple destinations, and is fully compatible with the entire Extron MTP product line.

The MTPX Plus 6400 Series provides additional features to dramatically reduce costs associated with rack space, cabling, and installation. These features include dynamic skew equalization to maintain RGB color alignment at all times, video level and peaking compensation for brighter and sharper images, and local RS-232 insertion ports to eliminate the need for control system wiring to remote displays.

Local high resolution video inputs and outputs eliminate the need for additional transmitters and receivers, and local audio output volume adjustment and muting eliminate the need for preamplifiers in many AV systems.

Extron

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Posted by Keith Clark on 09/01 at 12:23 PM
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JoeCo Shipping New BlackBox BBR-DANTE Recorder

New live audio recorder version marks successful collaboration between JoeCo and Audinate

JoeCo Limited is now shipping the BBR-DANTE version of the BlackBox Recorder, the latest addition to the multi-channel live audio recorder family that is the result of a collaboration, announced during 2010, between JoeCo and Audinate, creators of the Dante digital media networking solution.

The BBR-DANTE is designed to record/replay up to 32 channels of audio data from a Dante network. Developed to solve the inherent problems of working with computer-based systems in a live performance environment, it provides engineers with a computer-free, high-quality solution for multi-channel live audio capture.

Able to record up to 32 channels of audio at 44.1 kHz and 48 kHz directly to Broadcast WAV files on an external USB2 (FAT32 formatted) drive, the BBR-DANTE also accommodates higher sample rate recording (88.2 kHZ and 96 kHz) at a reduced track count.

The recorder connects to any Dante-enabled network device from a range of console and converter manufacturers. Designed to support standard network components and switches, it can also record 8 channels of analogue (balanced line in) alongside 24 channels of Dante for capturing audience and ambience.

Individual channels, or pairs of channels can be monitored on an internal PFL bus providing both hi-resolution metering and headphone output. A number of standard BlackBox Recorder safety features are also included, such as the Safe’n'Sound Record Recovery, which will recover files even if the power fails in the middle of a recording.

Dante delivers a no-hassle, self configuring, true plug-and-play digital media networking that uses standard internet protocols, in addition to providing a migration path to upgrade to new emerging A/V standards such as the IEEE Audio Video Bridging (“AVB”).

JoeCo managing director Joe Bull states: “The Dante enabled system is a great addition to the BlackBox range. We are delighted to have been able to combine this exciting Dante technology with the robust, computer free solution that the BlackBox Recorder provides.”

“JoeCo’s ability to design this product while maintaining their attention to detail, and rapidly getting the new BBR-DANTE to market has been extraordinary,” adds John McMahon, VP of sales, marketing and support at Audinate. “Based on early market feedback, we are confident the BBR-DANTE Recorder is going to be an extremely popular product.”

JoeCo
Audinate

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Posted by Keith Clark on 09/01 at 08:54 AM
AVLive SoundRecordingChurch SoundNewsPollProductDigitalEthernetInterconnectNetworkingAudio • (0) CommentsPermalink

Tuesday, August 30, 2011

Church Sound: When Technology Detracts Rather Than Enhances

Some advice for all of the IEM users out there...

If you read my writing with any regularity, you’ve probably noticed that in ear monitors (IEM) have been a recent hot topic. 

I like them - no, I love them! But only when they’re used on the right person, in the right venue, and in the proper way.

This past weekend I was at a conference that hit almost every one of these points, yet I found the IEMs to actually be a nuissance.

The event featured a very good worship band (think Chris Tomlin and David Crowder combined, strange but it worked well), solid keynote speakers and 1,100 teens and adults. 

The room was a nicely retrofitted sanctuary built in the 1980s, and new technology had been added to give it a nice flair. The video screen was well placed and looked good, the lighting and stage backdrop also looked good. 

The PA sounded good - it was well designed and tuned. I was sitting in the second row and the stage lip loudspeakers and the house loudspeakers worked together really well. 
.
On stage, the drummer was in an enclosed drum “cave” - Plexiglas front, padded rear, lid on the top and a door that sealed him in. (Ventilation must have been lacking as he was sweating profusely!)

There were no amps (guitar or otherwise) on stage, but I could see an ISO box for the lead guitar just off stage. All of the musicians (keys, acoustic, lead, bass, drums and the lead singer also played acoustic) had personal monitor mixers and wore in-ears - everyone hard-wired except the lead singer.

During the first session of the day, the lead singer would put his ear buds in and then pull one of them out, put it back in, pull it out - repeat, repeat, repeat. After a while, this really started to bother me. I was uncomfortable for (and with) him. 

I thought perhaps he didn’t have an ambient mic in his mix, or his mix was bad (it would be his own fault with the Aviom control and wireless transmitter on stage right next to him), or perhaps the ear buds weren’t comfortable (not fitted properly?).

During the next two sessions of music, he followed the same process, taking one bud in and out. It was downright distracting!

Also throughout the day, the lead guitar player constantly adjusted the clip on the wire of his ear buds.  It was on the back of his collar and held the wire in place. 

I don’t know if the clip was poking him in the back of the neck or if he was just nervous - and his playing with it was a result of that - but it was also very distracting.

During the second set, the keyboard player grabbed an acoustic guitar for one song. As he was playing, now free from the restraints of having to stay at the keyboard, he proceeded to walk to the “end of his rope” and have the ear buds pulled right out of his ears. Again, very distracting.

Musically the band was quite, and minus the IEM issues, their stage presence was excellent.

One final distraction - earlier I mentioned the drum cave, and frankly, I’m not a huge fan of shields.  In this room, with this style of music, and particularly with this heavy-handed drummer, the cave might have been necessary.

However, from where I was sitting, the snare drum sounded like it was going THUNK, THUNK.  I was hearing way more low-mids than anything else. 

I’m not sure how it sounded farther away - hopefully much better because you’d hear more PA and less of the sound that was leaking out of the cave.

My advice to all IEM users out there is to make sure that you’re comfortable with them before using them on stage. Because if you’re uncomfortable, you’re very likely to make your audience uncomfortable - or distracted at the very least.

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.

 

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Posted by Keith Clark on 08/30 at 03:10 PM
Church SoundFeaturePollInterconnectMonitoringStageSystem • (2) CommentsPermalink
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