Thursday, April 07, 2011
The Top 10 Tips For Improving The Worship Audio Mix Prior To Sunday
“Great mixes start way before the first service of the week"
The worship mixer’s job is executed in the mix position during worship, but its success is mostly established outside the mix position, prior to worship.
1. Rehearsal music
Get whatever rehearsal music media is available to the worship team for review (legally). Learn the arrangements by listening during the week.
Not only will your mixes come together quicker for each song, you’ll also anticipate things like guitar solos or false endings before they happen – not just after they’ve already begun. Does it really make sense when everyone on the stage knows the songs and arrangements thoroughly, but the sound tech does not?
2. Pre-production meeting
Meet with the music/worship and production teams well in advance of each planned service. Reviewing plans and expectations can ensure an appropriate audio set up, and can avoid potentially tough sound reinforcement surprises.
The worship department requests three wireless lavalier or headworn systems for a worship service. At sound check, they are placed on three actors and the tech quickly finds they’re not actors at all…they’re singers, and they’re asking for their vocals in the monitors!
If they are omni-directional it’s a tough situation at best, and practically impossible in many environments. Now, the worship department may have requested the drama style mics because the presentation or mood doesn’t suit the normal handheld vocal miking approach. But they didn’t anticipate the technical disaster that comes with their request (is it really their job to understand all the tech stuff?).
Heading this surprise off at an advance meeting allows the audio tech to suggest a better miking technique, such as normal handheld vocal mics or possibly cardioid headworn mics. But our point here is not about which mic technique is right for this application, it is that regardless of the chosen solution or compromise, it can be sorted out in advance – not at sound check.
3. RF performance check
If any wireless microphones, wireless in-ear monitor systems, wireless assistive listening systems, or any other RF devices are used in the worship space, they must be properly installed and their frequencies coordinated for compatibility.
Assuming proper installation, antenna orientation, and frequency coordination have been accomplished, it remains wise to periodically check RF performance. New sources of interference and other surprises are better found during testing – without an audience!
To properly check the systems, turn on all RF devices that will be on during worship, and turn on any equipment in close proximity to the RF devices. Portable transmitters and receivers should not be clustered together for the test - piling them together on a desk or other surface at the sound booth is convenient, but a common mistake!
They should be at least several feet apart, and located on stage or in a general area where they will be used. The outputs of all devices should be auditioned over the PA or with headphones (RF mics), on headphones or earphones (IEM receivers), or the receiver/transducer that will be used by the worshipper (assistive listening device).
4. System checks
Verify the PA system is in working order before Sunday morning. A brief walk/listen check a day (or a few) in advance can confirm that all PA zones/loudspeakers are working with no failures, and it’s wise to check other output zones too, like lobby, overflow, and monitor sends. A blown horn driver in the main PA cluster is not easy to resolve at 7:45am on Sunday!
5. Optimize microphone technique
Review the microphone selection and placements on stage. Choosing appropriate mics and optimizing placement can influence the PA mix notably by reducing leakage, increasing gain-before-feedback, and capturing better sounding sources.
6. Cue sheets
Get a copy of whatever cue/tech sheet or order of service outline is available or draw one up, if not. Clearly mark mic and roll-in cues, and any other important audio notes, in advance of sound check. Mixing notes can be added during sound check.
If mixing on a suitable digital platform, it may be possible to pre-program some or all of the cues and mix changes. But manual control should always be available, and the cue sheet should always be visible, whether in paper or electronic form. For very busy events, such as dramatic pageants, enlist an assistant to manage and announce the cues.
7. Sound check is not set up
Clearly distinguish between setup and sound check. Sound check is the time for audio personnel to dial in the mixes, with the elements (gear and musicians, etc.) working exactly as they will be during the worship service. Complete all audio setup work in advance of sound check, so that sound check really is just that – sound check!
8. I/O checks
Some worship audio techs add an Input/Output check procedure prior to sound check. This is highly recommended. I/O check takes a sound source (such as a CD), one person on stage, and one person at each mix position (2 people in many church applications).
Every input and output is briefly tested over the PA system (inputs) and over wedges or earphones (outputs). It’s a 5- or 10-minute effort at most, and this procedure verifies the entire signal paths from sources to worshippers (front of house) and sources to artists (monitors). And the occasional I/O that doesn’t work is identified and hopefully resolved before the worship team hits the stage – preserving sound check.
9. Review mixes
If you record your mixes, review them. If you are making a classic “board tape” right off the console’s PA mix, review it with the knowledge that it is mixed for the house sound and it does not include the live acoustic portion of the listening experience (which affects mix balance).
If you multi-track your services, you’ve got a great practice and training tool – play the tracks back through the front of house console. And if you’re fortunate enough to own a digital mixing platform that offers “virtual sound check” technology, you’ve got the ultimate tool for practicing, training, and fine tuning the sound reinforcement mix.
10. Ear training
Good mixing requires good listening skills, which require training and practice. Listen to great mixes that are relevant to your worship style, and “take them apart” mentally.
Discover the details that make good blends and mixes. Train your ears to identify frequency ranges. This skill is critical for sound reinforcement mixing and there are a number of useful training tools on the market. Or, simply practice with a tone generator and RTA (real time analyzer).
For more worship audio tips and techniques, go to Sennheiser.
ARX Introduces Multiple New Products At Musikmesse 2011
Making their debut at the show are the BSX 16 broadcast splitter and the MaxiDrive 6 channel high current line driver.
ARX Systems is pleased to announce the release of several new products at Musikmesse 2011.
New product releases include the BSX16 Active Microphone/Line Splitter, developed to deliver the performance required by the increasingly sophisticated level of today’s standards of E.N.G and broadcast audio production.
Also being shown for the first time is ARX’s new MaxiDrive - Six channel High Current Line Driver / Distribution Amplifier.
The MaxiDrive’s very low output impedance virtually eliminates the deleterious cable loading effects and high frequency loss normally associated with large scale Powered Line Arrays.
“Frankfurt’s always a great show for ARX and we’re looking forward to spending the Exhibition working with
Thomas Schrill and his team at our German Distributor I.A.D GmbH.,” said ARX’s Melbourne, Australia based Managing Director Colin Park.
“We’ll be introducing I.A.D’s local dealers and customers and our many international visitors to ARX’s new Products and explaining why investing in quality, hand made, long life products like ARX’s makes good business sense, plus we like the local Beer, and the sausages aren’t too bad either!”
Roland Systems Group Debuts R-1000 48-Track Recorder/Player For Live Events & Sound Checks
Designed to work with the V-Mixing System and can also connect with any digital console that has MADI output capabilities by pairing it with the Roland S-MADI REAC MADI Bridge
At the Prolight + Sound show in Frankfurt, Roland Systems Group has added to its V-Mixing System lineup by introducing the new R-1000 48-track recorder/player.
The R-1000 is an intuitive stand alone, dedicated recorder/player designed to work with the V-Mixing System and it can also connect with any digital console that has MADI output capabilities by pairing it with the Roland S-MADI REAC MADI Bridge.
As a recorder, the R-1000 captures up to 48 channels of discrete audio all as separate broadcast wave files ready to open in a DAW of choice.
As a playback device it can be used in live events to play back selected channels to augment a live performance or as a multi-channel playback deck in a theater or amusement park application. Sync two units together for a 96-channel recorder/player or sync to video with SMPTE (LTC) or via black burst.
All files are stored on a removable hard disk drive (HDD) or solid-state drive (SSD). Material can also be transferred via USB to a connected drive.
Virtual Sound Checks are now possible when the R-1000 is integrated with a Roland V-Mixer digital console.
Using a song previously recorded on the R-1000, simply switch to playback mode when arriving at a new venue and all the sources play back through the appropriate channels on the console.
Adjust the preamp gains on the console as you would if the band was live and the R-1000 takes care of the gain compensation. Then set compression, EQ, monitors, and effects.
When the band takes the stage, the user can be confident it will sound the way it did during the virtual sound check.
Setup and configuration can be done using the color LCD touch panel on the front panel or with the PC Remote Control software via a USB connection.
The R-1000 is based on REAC (Roland Ethernet Audio Communication) and eliminates the bulk and noise susceptibility typically associated with analog snakes and replaces it with Cat5e/6 (Ethernet/LAN) cable.
The R-1000 records superior audio by capturing the converted sound connected to the Roland digital snake systems. Analog inputs and high-quality mic preamps are located close to the source where audio is immediately converted to 24-bit digital streams and sent over Ethernet.
Using REAC, the digital audio signal is transferred throughout the complete system path en route to the R-1000 hard drive and then back to any outputs and on to limitless split positions. Using the REAC system for recording provides the highest quality possible not found in any other live multi-channel recording solutions.
The North American debut of the R-1000 will be at the upcoming NAB show in Las Vegas, at Roland Systems Group booth #C4345.
Roland Systems Group Website
Messe Frankfurt & NAMM To Launch New Russian Trade Shows In May 2012
Organizations become equal partners in new venture designed to grow new markets and benefit professional development and music education in the region.
Messe Frankfurt, producers of the Musikmesse, Music China, Prolight + Sound, and Prolight + Sound Shanghai, and NAMM, producer of the NAMM Show and Summer NAMM, have announced that they will be co-producing new shows in Russia in May of 2012.
The new shows will be called NAMM Musikmesse Russia and Prolight + Sound NAMM Russia and will take place in Moscow’s Expo Centre from 16 to 19 May 2012. NAMM Musikmesse Russia will focus on the musical instrument sector and the Prolight + Sound NAMM show will concurrently represent event technology – a combination that has been highly successful in Frankfurt.
Musikmesse, which reflects the musical instrument sector, and Prolight + Sound, which presents the world of event technology, take place concurrently at Frankfurt Fair and Exhibition Centre every year, and offer a variety of synergies for exhibitors and visitors.
Moreover, many products are also used in both the amateur and professional fields. Naturally, visitors to one fair can also attend the other. Both partners have many years of experience in organizing fairs for the musical instrument, pro audio and entertainment technology sectors. A
nd NAMM and Messe Frankfurt have been successfully cooperating on the Music China show in Shanghai since 2006. The Russian musical market is potentially the largest in Europe, characterized by the large population of around 145 million people, developed musical culture and sustainable growth trends over the last years.
The volume of the Russian musical instruments and technologies market is estimated at 600 million Euro (2008).
“With its events in Frankfurt and Shanghai, Messe Frankfurt has vast experience of the industries concerned and in the organisation of fairs and exhibitions. Moreover, Messe Frankfurt has an extremely good international network, which provides a solid base for the expansion of our activities.”
“Thanks to our expertise and our new partnership, we can offer exhibitors of the musical-instrument and event technology sectors outstanding opportunities for presenting their products and services around the world.”
“We are particularly pleased that NAMM is a partner with equally good knowledge of the markets.”
“The resulting partnership of equals will generate a multiplicity of positive effects for the musical-instrument and event-technology sectors in the CIS and the adjoining countries”, said Cordelia von Gymnich, Vice President Messe Frankfurt, Entertainment/ Media & Creation. “NAMM and Messe Frankfurt working together to open up the Russian, Eastern European and the Baltic States markets is a good thing for the global industry,” said Joe Lamond, president and CEO, NAMM.
“As partners with Messe Frankfurt in this new venture, we look forward to serving our global NAMM membership by increasing the size of the market and helping more people there experience the enjoyment and many proven benefits of playing music.” Messe Frankfurt
SurgeX International Introduces Sequenced Power Platform At Musikmesse 2011
The sequencers provide a total surge elimination, power conditioning and control solution.
SurgeX International is pleased to introduce the SEQ1200i Series sequencer, surge eliminator and power conditioner.
The SEQ1200i range provides power platform sequencing and control combined with the established, patented surge elimination and power conditioning technologies that SurgeX has become known for.
The SX1200i will be on demonstration at ProLight & Sound 2011 in Frankfurt, Germany.
Mike Mayne, SurgeX International Product Development Director, stated, “We are extremely pleased to extend the SurgeX International product line to include a total surge elimination, power conditioning and control solution.”
“The SEQ1200i offers scalable, sequential power up and power down functionality alongside the proven SurgeX surge elimination and power conditioning technologies that have reinforced SurgeX International as the number one global power conditioning solution provider.”
“The SEQ1200i is an invaluable product in its own right and also adds huge value to our remote control-compatible product offerings, such as the award winning SX1200RTi and recently released SX2200RT models.”
“The SurgeX International peace-of-mind is now backed up by the ability to sequence power up and down on systems of all sizes. We now offer a turnkey, modular power platform solution.”
The SurgeX International SEQ1200i line offers four sequenced banks of two industrial-grade IEC outlets plus two always-on outlets per unit. Bank-to-bank delay times of 2, 5, 10, 15 and 20 seconds can be specified via a front panel selector, and a series of front-panel LED status indicators display the progress of the boot up/down sequence.
A rear-panel Phoenix connector allows the unit to be connected in a master/satellite configuration with the ability to cascade an infinite amount of units. Cascading the SEQ1200i can be used to achieve limitless system sizes with sequential power up and down sequences.
SEQ1210i models are 10-amp load-capable (for China and Australia), SEQ1213i models are 13-amp load-capable (UK), SEQ1215i models are 15-amp load-capable (South Africa) and SEQ1216i models are 16-amp load-capable (Europe). All models have four sequenced banks each with two industrial-grade IEC outlets in addition to two always-on outlets.
“The SurgeX International SEQ1200i line is a logical evolution of the SurgeX family; it takes the established capabilities of our patented technologies and adds additional functionality that is advantageous in many applications.”
“The sequential power up/down functionality is prerequisite for many complicated AV, broadcast and custom design/build installations. Recent discussions during the development and launch process have drawn our attention to new sectors that our international product will be of considerable value to, including medical, IT, government and military sectors,” adds SurgeX Co-Founder and Senior Principal Michael McCook.
Advanced Series Mode surge elimination, Impedance Tolerant EMI/RFI filtering and COUVS Catastrophic Under/Over Voltage Shutdown provide the rock-solid power conditioning platform at the core of the SEQ1200i.
SurgeX International products are tested to and exceed the most stringent international standards, enabling use in the most critical environments, such as medical, government and military applications.
The reliability of all SurgeX products is backed up by a full 10-year manufacturer warranty. As with all SurgeX International products the SEQ1200i range is A-1-1 and IEC/EN 61643-1 certified.
SurgeX International products are manufactured in our privately-owned RoHS-compliant factory and are available through a worldwide network of authorized distributors and dealers.
Radial Engineering Releases Updated H-Amp At Musikmesse 2011
The speaker to headphone converter is now available, having previously been "backburnered" pending completion of other products.
Radial Engineering is pleased to announce the Radial H-Amp, a speaker to headphone converter that has been updated from the original spec and is now available for delivery.
“The redesigned H-Amp is one of those products that had unfortunately, been ‘backburnered’ pending completion of other products,” said According to company President Peter Janis.
“Several of our professional touring customers have been waiting on this product release. They can see the merit of being able to take any speaker feed and quickly convert it so that it can feed a headphone.”
“Well the wait is over, we have worked through the various designs and have come out with a new spec that we believe will address the concerns that were voiced when we showed some of the early prototypes.”
The H-Amp is a novel device equipped with two Neutrik Speakon connectors. One merely takes the ouput from a loudspeaker like a wedge monitor, connects it to the H-Amp and it automatically converts the speaker signal to a safe level producing a mono output for headphones.
The production version of the H-Amp has been upgraded from one output to two, enabling two musicians to share the same feed. Because the H-Amp is also able to be driven directly from a power amplifier, multiple H-Amps can be used in series by connecting through using more Speakon NC4 cables.
Each output is equipped with a separate level control plus a three-band filter set can be inserted into one of the headphone outs to alter the signal should two different types of headphones be connected at the same time. This allows sufficient control to help accentuate certain frequencies for improved audio.
Like all Radial products, the H-Amp is designed to endure the abuse of professional touring. 14 gauge steel construction throughout provides excellent shielding while the I beam internal frame ensures the PC board is stress free.
The innovative book-end construction provides a protective barrier around the switches, controls and connectors further lengthening life span. Plus a full bottom no-slip pad provides both mechanical isolation and electrical insulation.
Wednesday, April 06, 2011
DiGiCo Launches New SD Ten Digital Console At Prolight + Sound 2011
SD Ten sits comfortably between the SD7 and SD8 both in terms of performance and price, and is outfitted with some exciting new options.
The SD Ten is the newest console in the DiGiCo SD Series of digital consoles, unveiled at the Prolight + Sound show in Frankfurt at stands 8.0 F60 and G56.
The SD Ten sits comfortably between the SD7 and SD8 both in terms of performance and price, and is outfitted with some exciting new options.
The new console’s worksurface is constructed from anodized aluminum, overlaid with polycarbonate panels to provide clear and concise user feedback. A large 15-inch, touch sensitive screen provides both information and fast control of all the main parameters.
The control surface features 37 100mm touch sensitive faders, providing fast access to the console’s large number of channels, which include 96 with full processing, 12 of which can be configured as full Flexi Channels - and outputs that can be assigned across the surface. All inputs have dual mono inputs for fast ‘Main’ and ‘Alt’ channel switching.
Smart Key Macros are provided, accessed via four layers of ten backlit keys. The user can program these to control any functions, simple or complex, that they want to recall at the push of a button.
Local I/O, positioned on the rear of the console, comprises eight mic inputs, eight line outputs, eight mono AES I/O, two MADI connections with redundant cabling connections, 16 GPI and GPO connections (with the option to expand to 32 GPI and GPO), MIDI, plus Wordclock, MADI and Optocore for synchronization.
Standard input channel processing includes channel delay; single and multi channel presets; HPF and LPF with an industry leading 24db per octave; four bands of parametric EQ with band curve selection; compressor and date; dual insert points and access to all busing.
Standard output channel processing includes output delay; eight bands of parametric EQ (previously only seen on the SD7); compressor and gate; dual insert points; groups with bus to bus routing, plus Auxes that have direct talk to output with dim control.
Dynamic EQ provides both expansion and compression on all four bands of parametric EQ. These processors can be assigned to any of the input or output channels, whether stereo or LCR, with ten units being allocated as required.
In addition, any input or output channel can be mastered via the multi-band compressors - good for managing complex in-ear monitoring or difficult input channels - again, 10 units can be allocated.
Forty-eight assignable buses can be configured as mono or stereo groups, or auxiliary buses. There is also an additional stereo or LCR Master buss and 16 x1 2 output matrix, highlighting the SD range’s routing flexibility.
Meanwhile, dual solo buses give monitor engineers the comfort of accurate monitoring security.
Insertable FX and graphic EQs can be routed, controlled and snapshot recalled for the most complex show design. The 24 graphics can easily be inserted and controlled from the worksurface and 10
Stealth stereo FX units can be configured at any time from the palette of 33 Stealth FX. Integration with Waves plugins offers yet more processing options.
Unlike all other SoundGrid platforms, DiGiCo provide complete control of plug in parameters, as well as recall of snapshots and single loading/saving, directly from the worksurface. Sixteen stereo SoundGrid racks can be inserted, with up to eight plug ins in each rack.
As with all DiGiCo consoles, the SD Ten software runs on a standard PC or Intel-based Macintosh for offline preparation and remote control of the console. One SD Ten can also be linked to another, using a standard Cat-5 crossover cable, providing 74 faders for control. In this configuration, the audio engine of the first SD Ten provides complete redundancy for the other.
Further expansion options:
• Second-generation Optocore optic connections. Just like the SD7, an SD Ten can connect to up to 14 SD or D-Rack IDs with 448 audio channels on a single redundant optical loop.
• Five redundant consoles can share all inputs from the stage racks, and outputs on the system can be assigned in blocks of eight. For example, a single rack can provide 56 inputs to both consoles with 40 of the outputs assigned to the monitor console and 16 used by front of house for feeds back to the stage.
• One console can even directly route outputs to another console on the loop, for convenient tie lines.
• D-Rack, SD-Rack, SD7 and SD Ten can operate together at 96 kHz. The audio advantage here is clear, but also means just over 1ms of latency when routing a stage input through a channel and bus, with processing back to a stage output.
• SoundGrid Waves can be added, allowing full connection and control of a Waves SoundGrid, providing low latency plugins on a floating point digital console.
• The SD Ten comes with different I/O options because tailored DiGiCo systems and complex set ups are completely user configurable.
Finally, the SD Ten also features dual hot swap, switch mode, PSUs as standard.
Waves Audio Now Offering SoundGrid Compact System For Yamaha Consoles
Allows Yamaha users to process their live performance using Waves plugins
Waves Audio is now offering the new SoundGrid compact system for Yamaha consoles. The system includes everything Yamaha users need to process their live performances using Waves plugins, as well as the tools to capture their performances into a DAW.
The SoundGrid compact system represents the first time Waves has assembled all the components needed to run Waves plugins on a Yamaha console into a single, affordable package.
Users can process up to 16 live audio channels using up to 40 plugins in real time with super low latency; record MultiRack-processed tracks in real time, direct to a DAW in the same computer, pre- or post-processing (or both, for maximum flexibility); and adjust their plugins, racks and console settings.
Hardware includes the SoundGrid compact server (measuring 19 cm x 29 cm x 5.5 cm), the WSG-Y16 mini-YGDAI I/O card, a network switch, iLok USB key and Cat 6 network cables.
Software components include the following popular Waves plugins: MultiRack Plugin Host Application, Renaissance Reverb, Renaissance EQ, Renaissance Axx, Renaissance Bass and H-Delay.
Waves’ SoundGrid compact system is compatible with the following Yamaha products: PM5D, DSP5D, M7CL, LS9-32, LS9-16, DM2000, DM1000, 02R96, 01V96, DME64N, DME24N and TX6n/TX5n/TX4n. It is now available with a U.S. MSRP of $1,100.00.
Waves Audio Website
Tips To Overcome The Top Five Pitfalls Of Small Worship Room Miking
If you're not careful you can easily find yourself making these mistakes which can prove deadly in a live setting.
In a grand hall, a huge coliseum, and even a huge sanctuary, the more likely our mentality is “let’s do this the right way.”
Running sound in large venues drives us towards excellence. Now what is your mentality in a very small room?
If you’re not careful, you’ll find yourself slipping into five pitfalls. Learn those pitfalls and how you can avoid them.
1. We don’t need no stinkin’ microphones
The biggest pitfall in a small room is the lack of microphones.
I’ve fallen into this thinking a few times myself.
Each time, it revolved around one instrument; the piano. A small room, a grand piano…maybe a baby grand…lift the lid and let the music waft out…um, no.
The problem with thinking that the piano doesn’t need a microphone is while it might sound pretty good during a practice, when the room fills with people, it can get lost.
Use microphones on instruments no matter how small the room. You’ll be able to bring up the volume when it’s needed…and of course you can’t put an instrument into a monitor if it doesn’t have a mic!
2. Amp’s rule, boys drool
Ok, I’ve been hanging around my 11-year old daughter a lot and her lingo is starting to wear off on me.
In a small sanctuary, it’s easy to think that a guitar amp can fill the room with enough sound. While that can be true, there are two critical issues with this thinking; volume and direction.
First off, if you allow an amp total control over the volume of the room, then you lose the ability to control the volume.
As soon as the guitarist switches from pedal A to pedal B with 2x the volume, you’ll be suffering.
The other issue is direction. Whoever is sitting in the sanctuary and is inline with the amp, is getting an earful of that instrument.
There are a few things you can do; mic the amp while making sure the amp’s volume is low, use a line-out option on the amp, and point the amp up at the guitarist so that’s not directed at people in the congregation.
The only exception to this is a bass amp. If the room is small, you might not need to mic it as long as you’ve got a good steady volume. But that’s another story.
3. Priority Navel Communication!
Notice I said “navel” and not “naval.” I’m not talking about seafaring issues; I’m talking about singers using bad mic techniques.
The smaller the sanctuary, the easier it is for singers (and speakers) to think their vocal microphone is less important.
Therefore, instead of holding it right up to their mouth, they hold it much farther away - sometimes by their navel. At this point, it’s hard to pick up much of anything in the microphone.
This is a training issue in which you need to explain to singers/speakers that while the room may be small, their voice can easily disappear in the mix and the congregation finds it hard to follow along.
This is especially true of new songs. And just like the piano without a microphone, if they aren’t singing into the microphone, you can’t put it in the monitors for other musicians to hear.
4. Drum roll, please
The drums can be one of the biggest pains in the, in the, um…the drums can be a pain to mic in a small room. Do you mic every drum kit piece? Do you mic nothing?
This is where you need to spend time testing out different drum mic’ing techniques and determine which gives you the best overall sound and control.
You should at least use a single overhead microphone placed about a foot over the drummer and pointed at the drum kit. You won’t get the thump of the kick drum but you’ll have the ability to work a bit of EQ and volume on the snare and the cymbals.
If you check out Microphone World here on ProSoundWeb you can read all about the different drum mic’ing techniques, which you can also find in my free ebook.
5. Feedback frenzy!
I’d be remiss not to mention feedback issues in a small room.
A small room usually means a small stage. A small stage means vocal microphones are in close proximity to the monitors. This is where it’s crucial you have proper monitor levels and proper vocal mic technique by the singers.
The feedback issue I usually see (hear) is when a singer lowers their microphone by their side and places it inline with the floor monitor.
6. Let’s call this bonus #6
Not long ago, I had the stage set up and wasn’t getting the right sound from the kick drum.
As it turned out, the floor monitor for a guitarist was being shared with the drummer and the kick drum mic was inline with the monitor. So, the drum mic was picking up the monitor sounds.
Now this was a simple change in monitor location before the gig. Therefore, be careful where your monitors are pointing because those instrument microphones might pick up more than you want.
What problem have you experienced in a small room/small sanctuary? How did you overcome it? Be sure to let us know in the comments below!
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.
Roland Systems Group Unveils M-480 V-Mixer Digital Console At Prolight + Sound 2011
Built on REAC (Roland Ethernet Audio Communication), eliminating the bulk and noise susceptibility typically associated with analog snakes, and replaces it with Cat5e/6 cable
Roland Systems Group has unveiled the M-480 digital mixing console as the new flagship of the V-Mixing System lineup, where it also integrates with digital snake, personal mixing and multi-channel recording components.
The new M-480 V-Mixer offers 48 mixing channels and 6 stereo returns for a total of 60 channels. Busing is strong with a total of 27 consisting of 16 auxiliaries, 8 matrices and full support for mono, stereo or LCR sound design.
With a configurable choice of available digital I/O boxes, the M-480 can support up to 90 inputs and 90 outputs - all fully assignable via the digital patchbay. The patchbay also has the unique ability to route any input to any output without going through the mixer.
Each mixing channel includes 2 stages of dynamics processing, 4-band PEQ, and delay. Dedicated 4-band PEQ, limiting and delay are available on every output.
The intuitive interface of the V-Mixer lineup has been retained in the M-480, making it easy to learn and easy to use. It has rapid recall of setups/scenes, 25 - 100mm motorized faders, a high-resolution
color screen, and dedicated channel strip knobs for all bands of EQ, as well as pan and gain.
Effects processing is strong with six dual-mono effects processors as well as twelve 31-band GEQs. Built-in stereo recording/playback uses uncompressed WAV files via USB flash drive.
The M-480 also has a “Cascade” function, allowing two units to be connected together enabling a 96-channel mixing solution. The two connected consoles share AUX/Matrix/Main/Solo buses with bi-directional communication, providing a compact, affordable and powerful high channel mixing solution.
This fully digital system is built on REAC (Roland Ethernet Audio Communication), eliminating the bulk and noise susceptibility typically associated with analog snakes, and replaces it with Cat5e/6 cable.
The system generates excellent sound quality by converting analog inputs to 24-bit digital streams at the stage end via high-quality mic preamps located near the source. The V-Mixing System then secures the quality of audio signal throughout the complete system path en route to the M-480 V-Mixer, back to any outputs and on to limitless split positions.
The two 40-channel assignable output patchbays are ideally used with the M-48 Personal Mixing system and with the new R-1000 48-track recorder/player. The R-1000 in particular can be used as a multi-channel recorder or player and is perfectly suited for live multi-channel capture or virtual soundchecks, backing tracks, and training.
The M-480 V-Mixer integrates with PCs via its Remote Control Software for additional control or sending/receiving setup data allowing users to prepare channel setups and configurations before arriving at the venue. When connected via USB it allows independent control of the M-480.
The M-480 supports V-LINK/MIDI Visual Control for synchronization with video equipment. Used in combination with the Roland V-1600HD multi-format video switcher enables an “audio follows video” setup ensuring that audio levels are raised when and associated video source is live.
The North American debut of the M-480 will be at the upcoming NAB show in Las Vegas, at booth #C4345.
Roland Systems Group Website
Tuesday, April 05, 2011
Yamaha Commercial Audio Systems To Launch Dugan-MY16 At 2011 NAB Show
Will enable control of live microphones via the real-time voice-activated process known to many as the DUGAN SPEECH SYSTEM as a plug-in card for the Yamaha mixers
Yamaha Commercial Audio Systems and Dan Dugan Sound Design announce the launch of the Dugan-MY16 card for current model Yamaha digital mixers and processors including Yamaha 01V96, DM1000, 02R96, DM2000, M7CL, LS9, DSP5D, PM5D, and DME24/64N.
Yamaha Commercial Audio Systems will be the sole distributor for the card, which will be available to customers within North America.
The Dugan-MY16 card will make its official debut during the upcoming NAB Exhibition in Las Vegas at Yamaha Booth C1325 and Dan Dugan Sound Design, Booth C1623.
Dan Dugan Sound Design automatic mic mixing products eliminate cueing errors, reduce feedback and ambient noise pickup, allow for smooth transitions between talkers, provide consistent system gain no matter how many mics are open, have the ability of handling up to 64 live microphones, and can be remotely controlled.
A solution for live for broadcast, corporate A/V, houses or worship and other sound reinforcement applications requiring automatic mic mix capabilities, the new Dugan-MY16 card for Yamaha digital mixers is unique in that it will enable control of live microphones via the real-time voice-activated process known to many as the DUGAN SPEECH SYSTEM as a plug-in card for the Yamaha mixers.
The Dugan-MY16 card provides up to 16 channels of automatic mic mixing per card at 48kHz and will run at 96kHz with 8 channels of operation. It is patched into input channels using the consoles’ set up screens.
Channels can be partitioned into as many as three independent automixers. An internal web server will provide a full virtual remote control panel over a local network.
“Dan Dugan Sound Design products are highly respected within our industry, and the new Dugan-MY16 card was designed in response to an overwhelming request by our customers to incorporate Dugan technology into our products,” states Marc Lopez, marketing manager, Yamaha Commercial Audio Systems. “The new card will certainly complement our digital mixer product line.”
Multiple Dugan-MY16 cards can be linked for use in larger system situations; i.e., two cards will provide 32 channels of processing. The new Dugan-MY16 card can also link with all other DSP-based Dugan automixer products such as Models D-2, D-3, E, and E-1.
“My customers in the staging business are heavy users of Yamaha mixers,” says Dan Dugan. “They have been asking me for an MY-card version of my automatic mixing controller for years. I’m very pleased to fill their need.”
The Dugan-MY16 card will be available during second quarter of 2011 at a targeted MSRP of $2,700.
Dan Dugan Sound Design Website
Yamaha Commercial Audio Systems Website
Wohler Appoints Mike Descoteau as VP of Sales for North America
The industry veteran brings 25 years of experience to new role.
Wohler Technologies has announced the appointment of Mike Descoteau as VP of sales for North America.
He immediately assumes responsibility for Wohler’s sales operations throughout the eastern U.S. and Canada, and will report directly to Kim Templeman-Holmes, Wohler’s EVP of worldwide sales.
“Mike brings more than 25 years of experience and a truly vast amount of industry knowledge to his new role, and we are very excited that he is joining our team,” said Templeman-Holmes.
“He has held positions of responsibility at all levels of sales and regional management within the professional audio/video and broadcast media industries.”
“His solid relationships with major customers and strategic partners will be an important advantage as we continue to drive sales throughout North America.”
Prior to joining Wohler, Descoteau served as director of sales and marketing for CBT Systems, a leading TV systems integration, design, and consulting firm.
He also served as general manager and sales manager for Riedel Communications, a provider of digital intercom and communication solutions.
Other previous positions include director of broadcast services for Masque Sound, broadcast sales manager for Dale Pro Audio, and VP of sales broadcast for Solid State Logic. Descoteau has a Bachelor of Arts degree in political science from Plymouth State College.
Wohler Technologies Website
Clear-Com Concert Feature Enabled In Associated Press ENPS Application At 2011 NAB Show
The new release of Concert simplifies newsroom communications for journalists around the world.
Clear-Com has announced the introduction of Version 2.6 of the company’s Clear-Com Concert Intercom-over-IP Communications Solution at the 2011 NAB Show.
This latest version features the debut of “Concert for Newsroom,” an integrated intercom solution to Associated Press’ ENPS (Essential News Production System), allowing journalists using Concert to quickly, easily, and cost-effectively communicate with other newsroom members who also have access to the system.
Ideal for ENG applications, Concert for Newsroom will assist busy reporters chasing stories in the field to get in touch with their editors, producers and other production team members with the click of a mouse.
To start a chat with Concert, ENPS users need only to scroll through their contacts and click the appropriate name or icon to initiate a call and/or chat, simultaneously.
In cases where multiple members need to have a conference about a story, the user can drag-and-drop multiple contacts into a conference, or members can double-click on a session link and join the call—all while working within ENPS stories.
Users can see immediately the availability of any other ENPS participants for a call or text message, saving journalists time when deadlines are fast approaching.
“Concert has truly defined Intercom-over-IP with its ability to seamlessly integrate with traditional intercom systems and external audio interfaces over standard IP networks,” says Patrick Menard, Product Manager of Concert at Clear-Com.
“Now with the incorporation of the application into ENPS, users have even more options with which to communicate, helping them focus on the story and not technology. Time is often an enemy to those working in the field.”
“A low cell battery or being out of cell range can often cause issues when gathering info for a time-sensitive story. With Concert, this concern is taken out of the equation as you are communicating through ENPS.”
Concert can easily interface with external audio systems, including partyline systems, paging systems, program feeds and other matrix systems using a four-wire interface over a standard IP network, providing a seamless communications network.
Those using Concert within ENPS can also benefit from additional features, such as the ability to initiate simultaneous call or chat sessions without disconnecting the current session and receive audio program feeds from the newsroom and/or other users.
Program feeds can be interrupted by important communications from the news director or producer speaking over intercom, ensuring that all critical communications come through to the intended recipients.
Gefen Announces Extra Long Range HDMI Over CAT-5 Extender
The new extra long range extender from uses HDBaseT and power over line technologies to send HDMI, power, ethernet, IR and RS-232 over a single CAT-5 cable.
Gefen has announced its newest device for an extra long-range extension of HDMI devices.
The GefenPRO ELR Extender for HDMI with PoL offers a long-range, multi-signal method of extension up to 330-feet (100m) over just one CAT-5 cable, a significant improvement over previous solutions, which required two cables to reach shorter distances.
Gefen PoL technology based on the Power over Ethernet standard enables the delivery of power to the receiver unit, giving an added advantage to the integrator.
Power is provided to the receiver using the AC power connected at the sender unit.
Each receiver unit is also equipped with an additional 5v output to power a supplemental product, such as a scaler, switcher or splitter at the display location.
The GefenPRO ELR Extender for HDMI with PoL is ideal for installations where power requirements are limited at the receiving side, such as when mounting projectors on the ceiling.
By reducing the cabling requirements from two to one, it allows installers to easily tap into existing in-wall CAT-5 type cabling.
HDMI video at 1080p full HD resolutions, 3DTV support, audio, IR remote, RS-232 control, Ethernet and power are all delivered over the same CAT-5 cable.
HDMI features supported include 3DTV, deep color, lip sync and audio delay. Both the IR backchannel and RS-232 connection can be used to control the display from the remote location.
As a GefenPRO product, the ELR Extender for HDMI with PoL comes with advanced technical support, a two-year warranty, and plug and play performance.
Vintage King Audio Introduces The AMS Neve 1073LBEQ 500 Series Mono Equalizer Module
Vintage King has partnered with AMS Neve on the second 500 series incarnation of the 1073 classic module.
Vintage King Audio has announced the release of the Neve 1073LBEQ 500 Series Mono Equalizer Module, from AMS Neve.
The EQ is the second 500-series format module AMS Neve has built in the image of the Neve 1073 Classic module.
Their 1073LB mic pre debuted at AES 2010 last fall, receiving remarkable critical acclaim and incredible comparisons to the legendary Classic 1073.
The Neve 1073LB 500 Series EQ is available in the United States exclusively through Vintage King Audio and select distributors worldwide ($1295.00 USD).
The module will begin shipping April 28, 2011.
The original Neve 1073 module is arguably the most desirable EQ and microphone preamp and first choice for countless engineers since its inception 40 years ago.
Imitated (but never matched) by a number of manufacturers, AMS Neve has officially recreated the 1073’s big punchy sound for the second time, and made it available for your lunchbox with the Neve 1073LBEQ.
The EQ joins the Neve 1073LB mic pre in the company’s roster of 500-series format lunchbox modules.
“The industry’s response to the Neve 1073LB mic preamp was overwhelmingly positive and we anticipate that the 1073LBEQ will do equally as well in the marketplace,” says Mike Nehra, Co-Owner of Vintage King Audio.
“Vintage King’s customers specifically asked for the 1073 in a 500 series format and we worked closely with Neve for some time to bring it to the market. You can tell Neve really took pride and went to great lengths with this EQ as well as the 1073LB mic pre.”
“The build quality is solid and the tone of this unit stands neck in neck with classic and new Neve 1073 modules.”
Crafted in England by Neve engineers, the 1073LBEQ retains the unique sonic characteristics of its original predecessor by using the same architecture, matching components and legendary 3-band EQ design.
With new features like a signal presence LED and Neve’s Audio Processing Insert design, the 1073LB EQ takes you lunchbox to the next level and allows the audio to/from the EQ module to be inserted into the audio path of an existing 1073LB.
Vintage King Audio