Friday, December 02, 2011
Accidental Electrician: Eliminating Dreaded Sound System Hum & Buzz
An epic tale of finding and eliminating a system's long-time and ongoing hum and buzz issues. Just who was the culprit?
There’s no better feeling than when you’ve setup your system and turned it on to find it lacking any noise.
This is no major feat when you have control of the electrical distribution, but when you don’t, things can be a bit more dicey.
The classic scenario when the system powers up with a hum is for the operator to declare “ground loop!” followed by muttering and cursing while digging in the accessories box to locate the power cord cheaters.
I find it funny how a device that’s supposed to be used to provide an electrical ground connection on older two-prong outlets is most often employed to lift the ground instead.
In such situations, you gotta do what you gotta do to safely eliminate these nuisances if possible. But I often see that the whole ethereal concept of a “ground loop” is a distraction from one big fact: not every hum comes from a ground loop.
What seems to be nearly universal is that a lot of folks don’t understand that the existence of a ground loop is not actually the cause of a ground loop hum. A ground loop is only a condition that is exploited by the true problem, which is an electrical current flowing through the loop.
Carefully designing a system to not have ground loops is a noble engineering endeavor. But in my book, there’s no real reason to do so if you take the time to eliminate the actual hum sources and potentials.
Treat the cause, not the symptom.
One job I worked in the past was on a sound system with a noise problem that historically couldn’t be solved.
This system was in a mid-sized church in a community known as the “Home of the Hardheaded Dutch.”
(You may accuse me of racial stereotyping, but believe me, this is how they referred to themselves. And they were proud of it.)
When we first walked into the sanctuary, the system was on and I immediately heard a significant buzz. The contractor turned to me and said, “Oh by the way, we’re going to solve their noise problem too.” Which was to say, “You need to fix this because I can’t.”
I hemmed and hawed about how the problem could be from the transformer on the pole, but was actually just making stuff up out of thin air, aggravated about having been surprised with this additional time-consuming task being added to my already conservatively budgeted schedule.
The church elder and de facto tech director we were working with was a retired Master Chief named Dave. He knew diddly-squat about sound systems, but possessed a good nose for BS. I could tell he wasn’t buying mine.
Pocketful Of Cash
I got on with acquainting myself with the system. Much of it had been obtained as surplus from the 1986 World’s Fair in Vancouver. I was surprised to find a rack filled with Bryston amps, still with a few years of warranty left and working just fine!
What disturbed me was that everything at the amp rack was plugged into cheaters and Radio Shack power-line RF filters. Someone wasted a pocketful of cash to buy those.
I removed them and the molesto mucho buzz was transformed into a simple hum. A step in the right direction…
After eliminating the lighting dimmers as the noise source, there were no other obvious conditions that I could immediately identify as contributing to the problem.
So I went home and whined to my wife about how I had been abused by having this additional issue dumped in my lap. She didn’t buy my BS either.
Three days later, we were back, doing the first thing these projects always require: taking everything apart. As we slowly updated and re-assembled the system, I got to a place where I could examine the power distribution.
I strung an extension cord from one of the sockets at the amp rack to the front of house position, and then used my trusty old Wiggins to check between the hot on the extension cord and the hot at the front of house outlets.
As feared, it measured 208 volts, meaning that the two different power circuits were on differing phases. It was time to root around in the breaker box.
The building had a modern electrical system, which is fortunate. I’ve run across some problems in systems that were on legacy electrical distribution, and short of violating code, there’s sometimes not a whole lot that can be done to fix a problem.
Suffice to say that a dedicated electrical ground and a modern distribution system is imperative for safe, noise-free sound.
Four circuits in the breaker panel were dedicated to the sound system, and sure enough, they were grouped together all in a column. This was a typical commercial 3-phase box with the phase alternating for every row.
The shame was that I could tell by reading the written-over labels that originally the sound system circuits had been on every third row (i.e., a common phase).
So the original system installer (OI) had specified that the circuits needed to be all on the same phase. My guess is that some well-meaning electrician had thought it would be smart to group the circuits together during one of the church expansion projects. I returned the circuits to their original spacing.
Having all the power circuits on an identical phase is important for minimizing the possibility of an inter-chassis current in a ground loop.
Power supplies leak small amounts of AC to ground. If there happens to be enough leakage from gear in the ground loop, and it’s of differing phases, an inter-chassis current will flow through the loop great enough to induce a hum.
By establishing a common system power phase I minimized the ground loop hum potential.
Elsewhere though, the OI committed a serious faux pas. The original installation included a remote power switch for the amp rack, which consisted of a key-switch on a panel at FOH that fed AC from a transformer to the coil on a relay at the amp rack. Turn the switch, the relay closes and the amps have power.
The problem was that the OI appropriated one of the shielded balanced lines running through the FOH to stage conduit for the remote power switch. For 90 feet, there was a line carrying 60 Hz AC in cramped proximity with all of the system audio wiring!
Removing the remote power switch eliminated this hum potential. We could have run an alternate line for it, but Dave didn’t care for the attitude exhibited by those who had keys (control) towards those who didn’t so he decided to just do away with it.
Next, the system was gotten to the point where the console was re-patched and turned on. There was a hum.
Examining the console, I began to see that there were some channels and sends marked “Bad” and “Don’t Use.” I began to grow suspicious. Dave said that one day they came in and the console was suddenly having these problems. My gut said the console had taken a surge during a lightning storm.
They had continued using it not realizing that there was more to the console problem than just non-working channels and sends.
Replacing faulty gear (and a lecture about surge protection) was the solution for this hum source. We snagged the console from the church’s youth room and patched it in. Ahhhh, schweet silence! I declared, “This house is clean!”
Too bad it wasn’t so.
It was the afternoon of the project’s last day and there was a sound check scheduled with the worship team that evening. We had patched in numerous mics for vocals, piano, drums, guitar cab and DIs for acoustic guitar, bass and keys.
When I opened the channels and pushed up the faders, there was a hum. My heart sank and Dave now began to give me “the eye.”
It wasn’t on just one channel, but on almost every channel from the stage. Pull the faders down or mute the channels and the hum would go away.
So the problem wasn’t ground related because it was controllable. Phantom on or off made no difference. Dynamic mic, condenser mic, or DI also made no difference.
I hit the stage and started taking a closer look at things. I picked up one of the mic cables and was disgusted to find that I recognized the writing on the jacket. I had seen it once before.
This cable had a certain respected pro audio name on it that had no business being there. I don’t know if this cable was a manufacturing screw-up or possibly a counterfeit, but it had no valid use anywhere in a sound system. It wasn’t even shielded twisted pair.
I looked across the stage and counted upwards of two-dozen of these demon-possessed cables from hell.
Luckily, there was a decent pro audio store about 20 miles away. We bought out their stock and I cut up the old cables so no one would be tempted to salvage them from the dumpster and put them back into service.
The final fix was replacing the bad cables with good ones. For the first time ever, the system was noise-free and there wasn’t a single lifted ground or shield anywhere.
I suspect that the original issue was with the remote power switch, and as the other issues occurred, they contributed to the complexity of solving the problem.
It’s a shame that out of hardheadedness they put up with it for as long as they did. But I can well understand the lack of trust when it came to parting with their money over a seemingly unfixable problem that shouldn’t have been there in the first place.
BS comes in many forms. Sometimes as talk, sometimes as a bogus feature in an install.
Not many weeks later, the church decided to expand the project by replacing the ailing console, adding a personal monitoring system and investing in some surge protection.
I guess we’d earned the respect of the Master Chief. Hooyah!
Since his start 30 years ago on a Shure Vocalmaster system, James Cadwallader remains in love with live sound. Based in the western U.S., he’s held a wide range of professional audio positions, performing mixing, recording, and technician duties.
Tuesday, November 22, 2011
Altinex Unveils Two New AV System Control Packages
Offerings address both large and small AV system designs
Altinex has announced the availability of two new AV system control packages, each offering provides the same functionality, with the primary difference being the port capacity of the Neutron Series controllers central to each system.
Altinex Control Package #1 consists of the CP500-110 Neutron controller, the AC301-109 EDID blaster / IR learner (for infrared control functionality such as power, play, stop, etc.), the CP450-007 MultiTouch control panel with a 7-inch screen (for touch screen control of the AV system’s functions), and the company’s acclaimed AVSnap programming software for system configuration. This systems’ Neutron controller provides two RS-232 bi-directional ports, two IR (Infrared) ports, two relay ports, and two sensor ports.
The Altinex Control Package #2 is intended for larger AV system designs. This offering includes Altinex’ flagship CP500-100 Neutron controller (with integrated IR functionality), the CP450-008 MultiTouch control panel with an 8-inch screen, and the company’s AVSnap programming software. The Neutron controller included with this package provides eight RS-232 bi-directional ports, eight IR (Infrared) ports, eight relay ports, and eight sensor ports.
Larry Drum, CTS, Altinex regional sales manager for the Central United States, notes, “These two control packages offer AV system designers and integrators a tremendous opportunity to incorporate cutting-edge control functionality into their projects at a highly competitive price. The Altinex Neutron controllers provide the I/O capability required to incorporate computers, Disc players, projection systems, displays, audio and more—with support for IR control of the equipment, enabling presenters to be fully in charge of their delivery.”
Each AV control package is now offered at a dramatic reduction compared to the regular system pricing. Control Package #1 carries a system price of $2,149 while Control Package #2 (the larger system) is available for $2,990.
Monday, November 21, 2011
Berry Center Utilizes Allen & Heath iLive Flexibility To Handle Heavy Event Schedule
“Over the calendar year we do between 2,600 and 3,200 events. The modular design of the iLive platform lets us design the right system for every venue and configuration." - Brent Buchanan, Berry Center
The Cypress-Fairbanks Independent School District in suburban Houston is home to the Berry Center, a unique multi-venue facility that includes a theater, football stadium, arena, and conference center.
To accommodate the busy schedule of diverse events it hosts, Berry Center technical systems manager Brent Buchanan recently asked the school district to invest in the Allen & Heath iLive mixing system, acquiring two T-112 mixing surfaces, one R-72 rackmount mixer, and a total of five iDR Series MixRacks, including three iDR-32s and two iDR-48s.
“We were looking for a premium product with the versatility to handle a wide variety of uses,” Buchanan states. “Ease of use, portability, reliability, great sound – all these things were of critical importance when we chose our digital mixing platform. I did a lot of research, and Allen & Heath was really miles ahead of everything else on the market.”
While the Berry Center is a school district facility, it is operated as a public facility that generates revenue by hosting outside events as well. With about 108,000 students including 10 high schools, the number and variety of events staged there is daunting. In addition the facility rents to outside clients hosting a steady stream of business conferences, concerts and other activities.
Buchanan directs a staff of 15 AV professionals managing a full-service production department that operates all sound, light, video and staging.
“Over the calendar year we do between 2,600 and 3,200 events,” Buchanan relates. “The modular design of the iLive platform lets us design the right system for every venue and configuration. For many events, we just use a laptop or iPad to control the system. That’s why we bought five iDR MixRacks, but only a few of the traditional console surfaces. In essence, we got the equivalent of five complete systems of 32 channels or more for the cost of maybe two or three other digital systems. It’s a huge upgrade for us, and incredibly economical.”
This architecture is the key to the flexibility of iLive. The iDR MixRack contains both the DSP “brains” of the system along with the primary inputs and outputs, and can be controlled either via a Cat-5 cable or through a wireless Wi-Fi connection.
By using an iPad tablet or a laptop, it becomes possible to control a fully operational digital mixing system with no physical footprint taking up valuable space.
“That can be really important for events where the client needs to squeeze every possible person into the room,” notes Buchanan. “It creates a huge benefit with no loss of functionality.”
Similarly, the Berry Center staff is impressed with the advanced DSP capabilities built into the iLive MixRack system. “The most obvious benefit, of course, is that we don’t have to deal with the racks of outboard gear we had with our analog systems,” says Buchanan. “But what really blew us away is the sheer amount of DSP effects we have available. You can literally put compressors, gates, EQ, and more on every channel, and you don’t have to worry about running out of resources. It allows us to handle any technical requirement that might get thrown at us.”
The Berry Center is planning to acquire even more Allen Heath iLive gear to take full advantage of the five iDR MixRacks they already have. Items like the PL-6 wall-mount mixer and the XDR-16 expansion mixer are on Buchanan’s radar, as he looks for ways to squeeze every bit of value out of the school district’s investment.
Allen & Heath
Tuesday, November 15, 2011
Wohler Announces New Software Interface For MADI-8 Audio Monitor
PC-based interface enables quick configuration of channel naming and preset assignments
Wohler Technologies has announced a new software interface for the company’s MADI-8 in-rack audio monitor that allows for control and quick reconfiguration of channel presets from any PC. With the new software, operators now have the ability to name any or all of the 64 available channels and then assign them up to eight available presets. Configurations can be saved as files on the set-up application, so operators can instantly recall channel names and presets from previous events and assign them to other MADI-8 units.
“MADI-8 was the world’s first in-rack MADI monitor to offer simultaneous monitoring of up to eight channels,” says Jeff McNall, product manager at Wohler. “Now, with this new channel-naming and preset configuration application, the unit provides even more convenience in high-volume monitoring applications.”
Giving users the option of creating presets via a PC and keyboard rather than on the MADI-8 unit itself, the firmware upgrade and software application download significantly reduces the time required to configure a MADI-8.
For installations with numerous MADI-8 units, using a simple device server adaptor for each of the serial ports can enable all of the MADI-8 units to be controlled via an Ethernet network. This capability allows users to switch MADI-8 units between sport- or event-specific presets over the network, taking very little time to shift channel names and presets from one broadcast to the next.
“We designed the MADI-8 to be a flexible, powerful, and easy-to-use solution for cost-effective audio confidence monitoring, and this new software interface makes the unit an even better match for demanding broadcast and production environments,” adds McNall. “Wohler is evolving to incorporate the use of Ethernet networks in configuration of our products. We’ve done it on our AMP2-16V-3G monitor, as well, and we continue to look for additional ways that our customers can take advantage of having Wohler products networked.”
The new software application and firmware upgrade are free to existing MADI-8 users.
Monday, November 14, 2011
Audinate Co-Sponsoring AES 44 Audio Networking Conference In San Diego
Largest event on audio networking coming up November 18-20
Audinate will be a co-sponsor at this year’s Audio Engineering Society’s 44th International Conference on Audio Networking.
The conference is being held from November 18-20 in San Diego, California and is anticipated to be the largest event to date on audio networking.
AES 44 features 17 papers on diverse topics related to networked audio, including five panel discussions industry veterans, plus three major technology demonstrations highlighting audio network control protocols, distributed performance, and a networked post-production master class.
Audinate’s Dante networking will be demonstrated during the event. Dante is recognized for delivering a tightly synchronized, low latency media network over standard TCP/IP network infrastructure.
In addition, Audinate CTO Aidan Williams will speak on topics such as product design and control protocols. “Dante shall remain the industry-leading media networking solution and that also includes support for AVB,” Williams says, “People choosing Dante as a ‘future proof’ solution are expressing confidence in Audinate’s ability to deliver an easy to use media networking system supporting AVB.”
Conference chair Nathan Brock notes, “The conference is the best chance to learn about audio networking from the engineers who are designing and building the devices currently in use, as well as the next generation of networking technologies.”
Social events include tours of the audio, visualization, and networking facilities of the University of California, San Diego, which is hosting the conference and banquet for all participants.
Click here to find out more about the AES 44 Conference.
And click here to register on-line for the conference.
Lake Processing Added To Yamaha Digital Audio Products
New MY8-LAKE card includes Lake Processing features such as Mesa EQ, Ideal Graphic EQ, Linear-Phase Crossover, and other intuitive, sonic tuning elements
Yamaha Commercial Audio Systems has announced the development of the MY8-LAKE processing card, designed exclusively for Yamaha digital products, which adds Lake Technology to PM5D, PM5D-RH, M7CL, LS9, DM2000, DM1000, 02R96, and 01V96 digital consoles, as well as DSP5D expander, DME24N/64N processors, and TXn power amplifiers.
The new MY8-LAKE includes Lake Processing features such as Mesa EQ, Ideal Graphic EQ, Linear-Phase Crossover, and other intuitive, sonic tuning elements found in the DSP expansion card.
The processing power of this compact card offers up to 8 inputs/8 outputs in Mesa mode (system EQ), 4 inputs/12 outputs in Contour mode (crossover), or combinations of the above to suit the application.
Flexible I/O configuration via the console’s insert points and the card’s AES/EBU connectors along with the ability to run at 96, 88.2, 48, or 44.1 kHz, makes it easy to integrate Lake Processing into any live sound system.
Multiple cards can be utilized in Yamaha products that support multiple card slots.
“The addition of the MY8-LAKE processing card adds unprecedented sonic potential and versatility to Yamaha digital products,” states Marc Lopez, marketing manager, Yamaha Commercial Audio Systems. “The new card’s leading edge Lake Processing technology will further expand upon our direction to offer complete systems solutions to our customers.”
Lake Controller software installed on a compatible Windows PC allows precision control of the MY8-LAKE as well as other Lake devices in the system, while close compatibility with Rational Acoustics Smaart sound system measurement software contributes to smooth, efficient speaker system tuning. Over 1,000 loudspeaker presets are accessible for optimum performance in any environment.
“We are delighted to be able to provide Lake Processing as an integrated part of Yamaha digital product performance, and excited by the potential of this partnership with Yamaha to deliver the unique benefits of Lake processing technology to a wider customer base,” says Jon Alkhagen, managing director of Lab.gruppen and Lake. “The development of the MY8-LAKE further reinforces our credentials as a leader in digital audio processing and provides significant flexibility and benefits to both our existing users and those looking to integrate Lake into their systems.”
The MY8-LAKE will be available Spring 2012 with a target MSRP of $3,200.
Yamaha Commercial Audio Systems
Friday, November 11, 2011
AES X192 Audio Network Interoperability Standard Connects With Manufacturers
Will provide manufacturers with the means to remain with the network technology they are invested in, while also seamlessly interfacing with products that support other networks
An AES standards task group chaired by Kevin Gross is currently developing an interoperability standard for audio networking. The focus of the project, which is called “X192,” is addressing the need for interoperability between products of many different manufacturers.
Once implemented, X192 will provide manufacturers with the means to remain with the network technology they are invested in, while also seamlessly interfacing with products that support other networks.
There are currently a number of existing and work-in-progress protocols, each with a distinct heritage. These include the IEEE 1733 variant of AVB, Dante, Livewire, Q-LAN and RAVENNA.
With an “interoperability mode” built from existing protocols and compatible with existing network equipment, system integrators and end-users will be able to select and interface the products that best meet their design goals with confidence that the X192-enabled devices work and play well together. And by “interoperability” Gross means the ability for devices operating under various proprietary Layer-3 protocols to easily exchange audio data.
Alternately, manufacturers may find that X192, once fully developed will meet all their criteria and may decide to implement it as their only networking protocol. QSC Audio Products, LLC and Telos Systems’ Axia Audio division have become sponsoring members supporting Gross’s work.
Manufacturers and users of networked audio products are recognizing the benefits of using Layer 3 network technology and applying existing IP protocols such as IEEE 1588, RTP and DiffServ to the challenge of distributing high channel-count, low-latency, uncompressed digital audio. The benefits of such an approach are numerous and include compatibility with off-the-shelf network hardware, scalability, manageability and acceptance by IT professionals.
Gross, who conceived and developed the CobraNet system for transport of real-time, high-quality audio over Ethernet networks, describes the genesis of the effort. “When you’re working in the IP environment there are a limited number of ways to mix and match existing pieces to implement an audio network,” he explains. “So it’s inevitable that IP-based solutions will have similarities. As I surveyed various implementations it became apparent that these similarities provided an opportunity for interoperability.”
Gross is also active contributor to the AVB standards efforts, has helped QSC deploy Q-LAN, holds several patents, and has written papers and articles and presented on numerous AV networking topics. In 2006 he was awarded an AES fellowship for his contributions to digital audio networking.
The task group membership is comprised of representatives from prominent audio manufacturers including ALC NetworX and members of the RAVENNA consortium, network equipment and component manufacturers and key end users.
“We strongly support Kevin’s efforts,” states QSC VP of marketing Gerry Tschetter. “Two years of field experience with the Q-LAN protocol used by Q-Sys networked audio products have proven to us that an IP based approach to networked audio is the right solution. We are looking forward to working with Kevin on an interoperability definition that expands options for the industry.”
Clark Novak, marketing manager for Telos Systems and Axia Audio states, “Telos and Axia have advocated standards-based audio networking since we pioneered Livewire in 2003. The development of a networking standard whose benefits all broadcasters can enjoy is the logical next step for the industry. We’re delighted to be a charter member of the X192 group.”
Find out more about project X192 here.
Thursday, November 10, 2011
Radial Engineering Introduces MC3 Studio Monitor Controller
Acts as a monitor, sub and headphone management system
Radial Engineering has introduced the MC3 studio monitor controller, a “straight wire” passive monitor switcher and headphone amplifier designed to provide smaller production studios with a cost-effective yet high-performance monitor, sub and headphone management system.
Designed for active nearfield monitors, the MC3 is equipped with two stereo outputs and a separate send to feed a sub-woofer. Each may be precisely set using a trim control thus allowing seamless switching between loudspeakers.
Separate on-off switches allow any or all to be selected at one time and an adjustable dim switch may be activated to temporarily reduce the monitor level. A single control acts as the master for quickly setting the desired listening level.
All monitor switching and level controls are passive, thus ensuring no extra amplifiers are inserted in between the source recording system and playback monitors.
Unlike active switchers that color the tone, the MC3 is completely passive. In other words, it employs ‘straight wires’ between the input and output with only gold sealed relays and a variable resistor in between. This ensures the signal coming from the recorder to the monitors is not altered in any way.
The MC3 is equipped with a high output headphone amp that is capable of driving two sets of headphones at the same time. A separate ear-bud output makes for easy connection to iPod type environments.
All outputs may be summed mono for quick phase check and AM radio listening simulation.
As the MC3 will likely find its way into live production, it is manufactured with very high durability, including a 14-gauge steel enclosure with a unique protective zone to keep switches and controls out of harms way.
All steel cased switches and steel potentiometers provide maximum lifespan while the 100 percent discrete circuit topology is supported with a dual sided mil-spec circuit board and full ground plane to further reduce noise.
“Smart studios make money by being able to work in all mediums. For instance the ultimate playback can occur on ear buds, headphones, TVs, computers, gaming, car stereos and of course on a sophisticated home entertainment system,” says Radial president Peter Janis. “Effective mixing requires listening to various types of speakers and headphones to ensure the ultimate mix will translate well and stand up. The Radial MC3 is a cost effective device that enables today’s smaller production studios and post production suites to properly address the situation.
“And although there are all kinds of monitor controllers on the market,” Janis continues,“in our view, many are either overly complex or introduce active circuitry and artifact in between the recording system and the playback monitors. We wanted to bring an affordable solution to the market that would enable the recording engineer to quickly switch between monitors, listen via headphones and be able to turn on or off a subwoofer.”
Tuesday, November 08, 2011
Audinate & Brit Row Team Up To Deliver Pristine Audio For NFL Game At Wembley Stadium
The use of fiber optic cabling to deliver both Dante audio and control remotely gave real benefit
Last week, for the fifth consecutive year, the NFL (National Football League) held a football game in London, and once again Britannia Row Productions provided live audio for the event to the 86,000 crowd inside Wembley Stadium, utilizing Audinate Dante technology distributed audio to the sound reinforcement system.
An event of this magnitude incorporates large amounts of media, ranging from the pre-game show to video feeds, replays, referee microphones, PA announcements, and live satellite linkups to other games in the league.
The home stadiums of NFL teams are designed with this in mind, but as a soccer venue, Wembley Stadium is equipped to meet different demands. As a result, additional infrastructure is required to deliver the production needs of a regular season football game, incorporated with the fixed installation already in existence.
System engineer Sergiy Zhytnikov explains,“Signal distribution around the field was achieved using Dante-linked Lab.gruppen LM 26 and LM 44 processors. A total of 14 of these units were deployed for the game.”
“The use of fiber optic cabling to deliver both Dante audio and control remotely gave real benefit instead of using 2000 meters of analog cabling, which due to necessity was placed in cabling conduits at field level, along with every other kind of electrical cable imaginable,” he adds.
Dante also solved the problem of signal degradation over long distances. “Using Dante on the Lab.gruppen platform is reliable, presents an easier control system is faster to connect and remove, and most importantly, delivers higher quality audio,” Zhytnikov says.
These sentiments were echoed by veteran front of house engineer Roger Lindsay: “This was the cleanest audio distribution system we’ve ever used for an NFL game weekend.”
Lindsay went on to comment that he had received some very positive feedback from the visiting NFL production team on the smooth running of the event and the continual improvements in delivery of this complex task.
Friday, November 04, 2011
Church Sound: How To Resolve A Simple Line Check Problem
A three-step approach to resolving line check problems with vocal microphones
Resolving line check problems can be quick and easy when you use this three-step approach. It’s simple, it follows a logical flow, and it can be done in a very short period of time.
A line check is the process of checking that all instruments and microphones on the stage are sending signals to the sound board. This process happens before a sound check or might be considered part of the sound check. During a line check, you are apt to discover problems like bad cables, bad connections, dead batteries, and (gasp!) dead mixer channels.
I think of a line check as more than just getting the signal but also getting a clear signal as well. In some cases of line check problems, you will neither get volume nor a signal light on the channel. While the lack of the signal light can determine what you check, using the below process list, you’ll find out everything that should be checked when either case arises.
Resolving a simple issue with a vocal microphone
Let’s say the singer is singing into the microphone and you don’t hear anything coming through the main speakers.
Step One: Check for the obvious
These are all things you can do from the sound booth and/or have the person on stage easily check for you.
1) Channel fader. Make sure it’s at the 0 position as a good starting point.
2) Channel gain. Check the channel gain is turned up. If you aren’t seeing the signal light on the channel glowing on and off, then increase the gain to see if that’s the source of the problem.
3) Channel padding. Make sure you haven’t engaged the signal padding where it’s not needed. That could cut your signal so low you don’t hear anything.
4) Sound board volume. Make sure you have the master volume turned up on your sound board. Hey, I wouldn’t mention it if I hadn’t done it myself.
5) Subgroup usage. Make sure the channel isn’t routed to a subgroup. If it is, take it out of the subgroup and listen for the sound.
6) Wireless microphones – receiver power and signal. Make sure the receiver is turned on and the receiver shows a wireless signal. If you don’t see a wireless signal, ask the person on stage to make sure the wireless pack (or wireless handheld) is on. If it’s on but the pack/handheld doesn’t show any associated power light, it might be as simple as battery replacement. See your wireless microphone manual for how it displays a battery-strength light indicator.
7) Wired microphones. Make sure that if it has an on/off switch that it’s turned on. Ask the person on stage.
8) Channel/stage jack pairing. Sometimes, a problem can be as simple as the cable being plugged into the wrong stage jack or you have it marked as the wrong channel. Ask the person on stage to check the jack number.
Step Two: Follow the signal flow
1) Connection into microphone. Re-seat the cable into the microphone. Make sure the channel is off / muted when you do this.
2) Connection at stage jack. Re-seat the cable into the stage jack. Make sure the channel is off / muted when you do this.
3) Connection at mixer. It’s unlikely but it’s *possible* that the connection into the mixer was pulled out for that channel. Make sure it’s properly connected.
Step Three: Time to swap
1) Swap microphones. Swap the microphone for known good one and try again. You might have a microphone that’s gone bad.
2) Swap cables. Swap the microphone cable with a cable that’s known to work.
3) Swap stage jacks. Still not getting a signal to the sound booth? Might be something from the stage jack to the mixer itself. Connect to a different jack/channel and see if the mixer gets that signal. If that does get a signal, also try swapping cables on the back of the mixer from the good channel to the channel that wasn’t receiving the signal. If you still don’t have a signal, you have a bad channel on your board.
Follow this three-step approach to resolving line check problems with vocal microphones and you’ll be moving onto your sound check faster than a vocalist can say “testing 1, 2, 3.“
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
DiGiCo Launches Expanded I/O Distribution With New MINI & NANO Racks
Instead of all the I/O connections having to be in one place, they can be distributed throughout a venue at the most convenient points
The new DiGiCo MINI and NANO racks offer a wide range of input and output options for any DiGiCo SD audio system.
Multiple DiGiCo mixing consoles can be positioned in an Optocore 2G optical loop, ideally suited to complex live or broadcast productions where multiple consoles need to share and sub-mix I/O. An example of how this can work in the real world is a scenario of front of house, monitors and a live broadcast feed.
Where the MINI and NANO racks come into their own is that, instead of all the I/O connections having to be in one place, they can be distributed throughout a venue at the most convenient points.
“With a digital system it makes no sense to have long lengths of analog cabling between your audio sources or amplifiers/loudspeakers and a central I/O rack,” says DiGiCo marketing director David Webster. “In a theatre you might want 56 mic inputs and 24 outputs as a main I/O rack, then a few more each side of the stage, perhaps a few for an event in the foyer and some more in an adjacent rehearsal room.
“Now you can use an SD-Rack for the main onstage I/O rack, but have a NANO rack each side of the stage, another in the foyer and a MINI rack in the rehearsal room, all communicating and working with up to five redundant consoles.”
Alternatively, at a sports broadcasting event, a combination of I/O racks can be distributed about the field of play, all backed up on a redundant single or multimode optical loop. Up to 14 rack IDs can be defined on each loop providing a full optical distribution system.
The MINI rack has 4 x standard SD hot swappable I/O card slots. These can be populated with any combination of the SD-Rack I/O cards; currently these include Mic/Line, Line output, AES I/O, AES IN, AES OUT, ADAT, AVIOM, DANTE and an in development HD-SDi card. Standard on the rack are MADI I/O connections along with the choice of either HMA, OpticalCon or ST optics.
Half the physical size of the MINI rack, the NANO offers two SD hot swappable I/O card slots, with the same card options. Optical connections are again user defined with HMA, OpticalCon or ST options.
With DiGiCo’s Gain Tracking, all consoles can share the inputs of all racks, while any slot of eight outputs on any rack can be allocated to any console on the optical network, provided it has not been previously allocated by another console.
“Another advantage of the system is cost savings,” continues Webster. “For example, if the FoH engineer only needs eight outputs, he can use a slot of outputs on the rack that the monitor guy is using - so it means you don’t need to buy two racks.”
Together with DiGiCo’s SD and D racks, the MINI and NANO racks provide a completely flexible I/O rack solution for any situation.
Wait - Why Are We Doing This? Progress On The System Interoperability Front
We want to have products all speak the same language, but allow them to retain their unique personalities...
The professional audio industry in 2011 has been abuzz yet again with a recurring theme that I like to refer to as the “can’t we all just get along?” conversation.
As in years past, the topic of unifying standards between different manufacturers’ equipment to make exchange of information easier and more seamless was everywhere.
This has been going on for some time, and has lived under different names and terms: “interconnectivity,” “convergence,” and the most recent model, “interoperability.”
However, this year marked a rather interesting turn of events. An increasing number of folks finally seem to all agree that this is not just a good idea, but further, serious conversations are being had about just how to go about getting it done. Customers are asking for it, manufacturers are honestly and openly talking about it, and huge levels of cooperation are emerging from the sturm and drang that frequently hinders real progress in this area.
Industry standards and initiatives such as AVB (IEEE 802.1 Ethernet Audio/Video Bridging) and X192 (AES standards task group for audio interoperability over high-performance IP networks) are not only starting the conversation about transport and content interoperability, but are backing it up with real work to promote the topic, bringing people to the table and making it happen.
Previously, the topic of interoperability was focused mainly on media content and transport. This is a critical and highly valuable part of the conversation, to be sure, but something was missing: system control. It’s an element that every system designer is acutely aware of and struggles with daily, yet until recently, it was absent from most discussions.
While the ability to exchange media freely between devices provides an obvious benefit, there is still a huge issue of how to tell these devices what to actually do with it once they have it. In other words, how do you control, configure, monitor, reconfigure, operate, adjust, modify, edit and generally manage these devices?
To tackle this issue, a new organization has been formed called the OCA Alliance, made up of individuals from nine companies who share the vision of an open, flexible, powerful system control solution for professional media networks. To this end, the group is collaborating closely to develop a system control network protocol suite known as the Open Control Architecture, or OCA. It’s the goal of the alliance to transfer OCA into the public standards domain as soon as possible, so that anybody may use it.
After the announcement of the OCA Alliance went public, I had numerous conversations with people from highly divergent areas of the AV industry, and a lot of their reactions went pretty much like this:
“You guys are working on an open control standard, huh? Cool!” (Pause) “So that means that anybody would potentially be able to implement and use this in their products? Neat!” (Slightly longer pause) “Wait - why would anybody actually want to do that?”
It’s a valid question. Upon first hearing about a unifying control technology, people are generally filled with joyful visions of how the AV industry might finally have the benefit of control interoperability that MIDI, DMX512 and others have provided to other technologies and markets. However, as one begins to think about the practical ramifications and implementations of a technology like OCA, doubts begin to creep in and one may wonder why exactly this is something that any sane manufacturer would actually want to adopt.
I can assure you that the members of the OCA Alliance are quite sane and have a very clear vision of what a technology like this means for our industry. But to see that vision, we need to look beyond how we have been working within our industry and take a longer view of how we would like to work.
So for the moment, let’s set aside the hows, bits, bytes and technical details of OCA and focus on the whys.
What We’re Talking About
What exactly do we mean when we say “control?”
Before getting into a discussion as to why all of this is important, we need to first get a handle on what we’re talking about, and - equally important - what we’re not talking about, because “control” can mean a lot of different things to a lot of different people.
On the one end of the spectrum, there are low-level details that need to be addressed within a media network. Functions such as configuring the network switches and routers, and discovering all the networked devices, are critical elements. These details are being addressed by the transport and infrastructure standards groups that are creating technologies such as AVB, but it’s not what we’re talking about here.
On the other end of the spectrum are the definitions of how network devices actually function and operate. This includes details such as what kind of DSP features and functions are available in a given device, the parameters that are available within those functions. and how those algorithms are actually coded.
This kind of control approaches dangerous territory, since it may touch on aspects of products that make them unique (for better or for worse).
For example, a frequent topic of discussion in regards to DSPs and filters is how the function of Q is defined within an equalizer algorithm. This is certainly a valid discussion, but is not part of the OCA conversation or concept. Another example might be a compressor function available in a certain product that, alongside the more standard parameters of Threshold, Ratio, Attack and Release, might contain some additional parameters that are unique to that device or implementation.
Interact, Not Standardize
OCA avoids these issues by confining itself to interacting with parameters and functions, but not defining the functions themselves. Details such as unifying or standardizing algorithms, parameters, and device functionality are soundly outside of OCA’s scope. OCA can set the Q parameter of an equalizer - but exactly what the equalizer does with that Q value is up to the equalizer, and is not standardized by OCA.
To clarify, let’s look at the more familiar world of MIDI. When we connect a keyboard controller to a MIDI tone module and press a key, a message is transmitted to the tone module telling it to execute a certain function (make a sound) within certain parameters (at the velocity of the key press, at this specific note, etc). This standard control message behaves exactly the same way regardless of the manufacturer of the tone module, and the desired function occurs.
However, that control message has absolutely nothing to do with the inner workings of the tone module itself. Details such as polyphony, synthesis method, or subjective quality of the sound that is generated, have absolutely nothing to do with the control message itself and are unique to the device that is carrying out the function.
In a nutshell, then, the goal of OCA is to create a standard method to interact with devices and their functions, not to standardize the devices and functions themselves. We want to have products all speak the same language, but allow them to retain their unique personalities.
Now that we’ve identified exactly what OCA is attempting to do and how it intends to fit in with the rest of the industry and technology at large, the next obvious question is “sounds great - but what’s in it for me?” I’ll answer that question here next month.
Ethan Wetzell has worked in audio for over 20 years, in positions ranging from front of house and studio engineer to global product manager for Electro-Voice DSP. He currently works as platform strategist for Bosch Communications Systems and works with the OCA Alliance.
Posted by Keith Clark on 11/04 at 09:49 AM
Wednesday, November 02, 2011
PreSonus FireStudio (26 x 26) Now Lion-Compatible
With this release, all FireStudio Series interfaces, including discontinued models, are compatible with OS X Lion
PreSonus has released FireControl 2626, a stand-alone application that provides Mac OS X 10.7 Lion compatibility for the original FireStudio (26 x 26) audio/MIDI interface.
With this release, all FireStudio Series interfaces, including discontinued models, are compatible with OS X Lion.
FireControl 2626 is exclusively for the FireStudio (26 x 26) and OS X Lion; FireStudio users who are running earlier versions of Mac OS X or who are using Microsoft Windows do not need and should not install this release.
Users of all other FireStudio-series interfaces and users of StudioLive-series mixers should update to the recently released PreSonus Universal Control 1.5.2, which provides Lion compatibility and other enhancements for those products.
FireControl 2626 allows FireWire daisy-chaining of two FireStudio (26 x 26) interfaces, and it is possible to chain FireStudio interfaces with other FireStudio-series interfaces by running both FireControl 2626 and Universal Control 1.5.2 simultaneously.
FireControl 2626 and Universal Control 1.5.2 are free downloads and are available immediately here.
Clear Path: The “Right” DI For Computers In Audio
When it comes to best audio quality practices, they’re sometimes not ideal.
Ready, brace yourself, this one is going to hurt: Computers are not made for audio. There, I finally said it!
Computers are made for crunching numbers, and they also happen to be able to manage audio and video tasks really well because of their tremendous processing power.
However, when it comes to best audio quality practices, they’re sometimes not ideal. Often we use computers to feed tracks to PA systems and as playback machines for things such as backing tracks. Getting the sound from the computer into a sound system is relatively easy: Connect the 3.5-mm (1/8-inch) unbalanced output jack and away you go. If only it were that simple…
Anyone who has done this knows that more often than not, it can introduce a ground loop or induce noise via the unbalanced line. Even PA system noise can find its way into the computer, adding noise to the program material output. Amplify any of this with 20,000 watts and you have a problem.
Several companies produce direct boxes that are specifically designed for computers. These are usually stereo, and more often than not, are passive or transformer based.
In other words, the transformer not only converts the unbalanced signal into a balanced one, but also introduces galvanic isolation to eliminate stray DC currents from traveling in between the computer and the audio system. And when the ground is lifted, all of the audio passes through the transformer disconnecting the ground thus eliminating the ground loop.
Because the computer’s output is buffered (usually by a -10 dB consumer level or headphone jack), a passive DI is perfectly suitable for computers. Transformers can usually handle a lot more signal before distortion when compared to phantom powered active DI boxes. This makes them a better choice when using the headphone jack.
Passive boxes for interface computers include (left to right) the Whirlwind pcDI, Proco AV1B and Radial ProAV2.
The active direct box was originally developed as a means to eliminate loading that would occur on low output electric bass pickups. By introducing a buffer, the bass signal going to the artist’s stage amp would not be affected thus conserving his sound while the PA system would be fed a hotter signal.
Buffers are essentially amplifiers. This means that they need power (voltage and current) to make them work. The preferred power source is 48-volt phantom because it does not require running separate AC for the DI box.
The other hidden advantage of a buffer is that the signal will only go one way. Unlike a transformer that is bi-directional, buffers do not allow signals to go backwards. Where this matters in our world is preventing noise from polluting the computer.
And because most program material is limited during the mastering process, one can get sufficient headroom using phantom power to generate a relatively clean signal. The problem, unless dealt with, is the lack of galvanic isolation; active DIs don’t solve ground loop problems.
There are some DI boxes that combine the benefit of an active direct box with transformer isolation. These are usually a little more expensive than a simple passive or active DI because they offer the best of both worlds. The transformers isolate the computer from the PA, while the buffers inhibit PA noise from polluting the computer.
Peter Janis is the president of Radial Engineering and has worked in professional audio for more than 30 years.
Posted by Keith Clark on 11/02 at 11:50 AM
Monday, October 31, 2011
Gepco Unveils Two-Channel Heavy-Duty Tactical Cat-5e Cable
Solution for applications that require multiple or redundant channels of Cat-5e cables in remote production or staging applications
Gepco International has introduced CTS2504HDX, a 2-channel snake consists of two elements of Gepco’s CT504HDX heavy-duty tactical Cat 5e cable under a rugged TPE jacket.
It is a solution for applications that require multiple or redundant channels of Cat-5e cables in remote production or staging applications.
Typically, the electrical performance and bandwidth of conventional Cat 5 cable is degraded through physical damage when used in portable applications, with the unique double-jacket construction of the CT504HD series designed to eliminate this issue.
While the inner jacket maintains the proper physical spacing between pairs to achieve ISO/IEC or TIA/EIA Cat 5e specifications, the durable TPE outer jacket protects the cable from physical damage or abuse.
In addition to the new CTS2504HDX, the CT504HD Series of heavy-duty Cat 5e cables includes three other types. The original CT504HD has 24 AWG stranded conductors for exceptional flexibility, while the CT504HDX features 24 AWG solid conductors for lower attenuation that allows for the full, recommended TIA distances for Cat 5e network cable. With the same basic construction as the new CTS2504HDX, the CTS4504HDX is a 4-channel snake consisting of four elements of CT504HDX under an overall rugged TPE jacket.
Heavy-duty tactical category 5e Assemblies provide a pre-terminated cabling solution for hostile environments. The CT504HD, CT504HDX and each element of the CTS2504HDX and CTS4504HDX can be terminated with either standard Cat 5 RJ45 connectors or ruggedized Neutrik etherCON connectors.
“The concern among Cat 5e cable users in the professional audio/video industry has been that it isn’t durable enough to handle the traditional wear and tear associated with the workload,” states Joe Zajac, market development manager for Gepco brand products. “Our CT504HD series was designed specifically to meet the needs of portable applications and provides the answer to professionals who are looking for a Cat 5e solution in remote environments.”