Wednesday, November 03, 2010
Tech Tip Of The Day: Microphone Mysteries Revealed
The benefits of various polar patterns.
This Tech Tip Of The Day come to you in editorial form because we wanted explain the benefits of various polar patterns.
Variable polar patterns can actually be good for several things.
Before we go into them, we should probably briefly discuss what the main polar patterns are.
The three “main” polar patterns are cardioid (or unidirectional), figure-eight (or bidirectional), and omnidirectional.
You can find far more detailed definitions in past Microphone World features, however, in a nutshell a cardioid microphone picks up sound from the “front” only.
A figure-eight microphone picks up sound from the front and rear of the microphone. An omnidirectional microphone picks up sound from all around.
There are also a few polar patterns that fall between cardioid and figure eight that are supercardioid and hypercardioid.
The supercardoid pattern has more side rejection than cardioid…in other words, it’s even more directional…but there’s a little more pickup from the rear. The hypercardioid pattern offers even more side rejection, but there’s even more pickup from the rear.
If you look at the patterns side by side you’ll see a “progression” from cardioid to supercardioid to hypercardioid to figure-eight where the side rejection gets better but the lobe in the back grows until pickup from the front and back is equal and the rejection on the sides is almost complete.
So now that we know what they are, what are they good for? Well, first of all there are the obvious advantages that apply to certain situations.
The cardioid pattern is by far the most used, especially in the studio; as for the most part people point a microphone at a source and record it.
However, if you want to pick up, say, a group of background singers, the omnidirectional pattern would be the most appropriate as it picks up sound from all around.
It’s also useful if you want to pick up the sound of the room you’re recording in, such as when you’re using a microphone as room microphone for drums or when you’re recording an orchestra in a nice-sounding hall.
Likewise, a figure-eight microphone may be useful when you’re recording two people singing together who want to face each other as they do so.
They’re also good for picking up the sound of a room as they pick up more of the sound in the room than a cardioid microphone, although not as much as an omni.
Also, as mentioned earlier, the figure-eight pattern offers nearly complete rejection of sound coming in from the sides, so if you’re ever in a situation where you want to pick up as little of something as possible…
Say, a computer in a small home studio, or a certain instrument in an ensemble recording…you’ll do the best job or rejecting that sound aiming the side of a microphone with the figure-eight pattern at the sound you want to reject.
In addition to those obvious differences, there are some less-obvious advantages to using certain patterns in certain situations.
For instance, an omnidirectional microphone exhibits little or no proximity effect, so if you have to have the microphone extremely close to a source and you want it to avoid the buildup of low frequencies that’s inherent with a directional microphone, an omnidirectional pattern would be a good choice.
In fact, the omnidirectional pattern tends to offer the most natural sound all around as it doesn’t have the off-axis coloration that’s a byproduct of directional patterns, which employ mechanical or electrical mechanisms to cancel out off-axis sounds.
Not that that’s a bad thing…in fact, switching patterns on a microphone is often a good alternative to changing the color of the sound without resorting to equalization.
Most variable-pattern microphones will include frequency response charts for each of the patterns the microphone can be switched to as well as graphs that show the response to different frequencies with different patterns.
All microphones, for instance, become more omnidirectional at lower frequencies and more directional at higher frequencies…just to varying degrees.
Also, different patterns are required for certain stereo microphone techniques, such as Blumelein, M/S, even Decca Tree configurations.
Finally, it’s probably a good idea to mention a few differences between variable-pattern microphones and fixed-pattern microphones.
Most of what we’ve discussed here applies to both, but there are a few differences. First off, there are obviously some advantages to variable-pattern microphones.
As mentioned, not only will the pickup pattern vary as the different patterns are selected, but the frequency response and color will change as well, and it can be very handy to be able to try different colors without having to switch microphones out.
Some microphones offer just two or three patterns, some offer a few more intermediate steps, and some have continuously variable patterns, which can be great for dialing in specific sounds.
Multipattern microphones typically are condenser microphones with two capsules back-to-back, and the different patterns are achieved by applying different amounts of power to one or both diaphragms (as well as switching polarity for certain patterns).
As such, a multipattern microphone set to the omnidirectional polar pattern…which is basically two cardioids back-to-back…may still exhibit a small amount of proximity effect.
Also, as mentioned earlier, all microphones become more and more directional at frequency increases, so while a “true” omnidirectional microphone’s pickup pattern will approach that of a cardioid at higher frequencies, a variable-pattern microphone’s response will approach that of a figure-eight microphone when set to omnidirectional.
Also, depending on the level of quality control employed by the microphone manufacturer, the front and back capsules may sound quite different from one another, which could especially be a problem when using figure-eight microphones in a Blumlein or M/S configuration.
As always, we welcome input from the PSW community and would love to know your thoughts on polar patters or any other microphone-related topic. Feel free to let us know in the comments below!
For more tech tips go to Sweetwater.com
Sanken Introduces The New CS-2 Short Shotgun Microphone At AES 2010
The CS-2 is a compact mic with long reach.
Sanken Microphones in introducing at AES 2010 a new shotgun microphone, the CS-2.
This, the newest model in Sanken’s comprehensive shotgun microphone line offers extended reach in a standard length mic using the company’s gradient tube length and rectangular diaphragm design.
The CS-2 achieves supersharp directivity with a 120mm (4.75”) long acoustic tube in a standard 250 mm (10”) length / 19mm (3/4”) diameter body. As a result, a natural tone is produced throughout the frequency spectrum, emulating the sound of much longer shotgun microphones.
The light weight of the CS-2 makes it ideal for camera-mounted or boom pole operations and suitable for a wide range of applications, such as outdoor location sound, interviews, sports, drama, and variety shows.
A new Sanken feature, the “high boost switch,” compensates for the attenuation of high frequencies when the mic is used with a windscreen/windjammer for outdoor use. As a result, the full, natural sound is maintained.
An external brass chassis increases durability, while the PPS (Poly-gold Phenylene Sulfide) diaphragm membrane provides optimum resistance to changes caused by adverse temperature and humidity. The resulting sound is consistent throughout the day in a wide variety of environments and changing climatic conditions.
Sanken Microphones Website
Shure Helps Educators To Record The Sounds Of Native American Culture
The University of Washington music education team documented Native American music at the Yakima Nation Tribal School.
While most university music education programs concentrate on the preparation of prospective teachers for traditional work in school orchestras, marching bands, and jazz bands, the University of Washington in Seattle has added cultural relevancy to the mix.
Among the requirements of students in Dr. Patricia Campbell’s music education program is a residency for performing and teaching at the Yakima Nation Tribal School in the city of Toppenish, located on the Yakama reservation in south central Washington.
“The Tribal School has a program where students are taught to play native instruments,” said Robert Pitzer, a doctoral candidate who assisted on the project.
“They don’t have a marching band or a jazz band. Instead, the students learn powwow music, and do improvisatory music on Native American wooden flutes and handheld frame drums. It’s unique and interesting, both musically and culturally.”
“We decided it would be a great idea to record their performances, and Shure was kind enough to lend us a selection of microphones for that purpose.”
The recording process was also an education for Pitzer, who had no formal experience in that area. “I’m a music education guy, a band director, so this was new territory for me,” he said. “Since we had access to this wide selection of Shure microphones, we basically tried different models and positions and compared them in headphones.”
“When we found a combination that sounded pretty good, that was the mic we went with. I guess you could say we took an empirical approach.”
In addition to traditional tribal vocals, there were native flutes and handheld frame drums.
Fifteen performances were recorded, ranging from soloists and duos to larger groups with drums and vocals. “We ended up using three different microphones.”
“For ensembles, we used both the KSM32 and KSM44,” notes Pitzer. “We generally miked the group, using a pair of mics. Everything was done live, with no overdubs. We got much better results that way, as opposed to miking each individual.”
For solo performances, and to isolate a specific performer, the SM57 was selected. “We found that mic had a great sound for the flute and the frame drums, and gave us the best isolation,” he said.
“At the suggestion of the school’s music teacher, we recorded the flutes in the shower of the boys’ gym for the natural reverb, which sounded amazing.” Solo drum with vocals were done in the classroom, captured by a single SM57 positioned about two feet away to balance the sources.
Ensemble recording took place in the school’s highly reverberant gymnasium, and relied on a mix of KSM32, KSM44, and SM57 mics. “We used the studio mics in pairs for the vocal ensembles, just trying to get a good balance of the group while capturing the sound of the room,” said Pitzer.
“Everything was recorded live. To make sure we got everything in good balance, we had the boys doing frame drums and vocals in one area, with the four female vocalists in a different spot. It really worked out well.”
The University of Washington’s music education team was very pleased with the results of their recording experience. “This was a great opportunity to document a different approach to music education, incorporating cultural elements that make music relevant to students in a different way,” said Pitzer.
“These recordings sound like you are standing in the room during the performances, which is exactly what we’d hoped to achieve. Considering how little recording expertise we had, I have to give a lot of the credit to the Shure microphones.”
“They made it easy to get great results.” The team will give back to the Yakama youth, sharing copies of the recordings they made.
“Close relationships with our customers are key, and we are proud to have industry experts like Willem and Eric on board to foster those relationships.”
Shure Incorporated Website
Posted by admin on 11/03 at 07:29 AM
Auralex Announces Continued Green Initiatives With The Launch Of Sustainable Bamboo Diffusors At AES
This new line of products is the first line of acoustical products made from 100% eco-friendly bamboo.
Auralex Acoustics, Inc. is continuing its history of green innovation by announcing the first line of acoustical products made from 100% Eco-Friendly bamboo, the Sustain Bamboo Sound Diffusor Series at AES 2010.
The new line, which consists of the WavePrism, WaveLens, QuadraTec, Peak Pyramid Diffusor and KeyPacs, retains the longevity and acoustical qualities Auralex is known for with green and acoustical properties of natural bamboo.
“Auralex prides itself on being an environmentally friendly company and we are excited to announce an entire line of bamboo-based products as a green alternative to traditional hardwood,” said Eric Smith, founder and president of Auralex Acoustics.
“Auralex will continue to fine-tune its manufacturing in order to better sustain the earth’s resources while also producing a great acoustical treatment solution.”
The Auralex WavePrism eliminates flutter echoes and other acoustical anomalies without removing acoustical energy from the space. The closed box design configuration of the product disperses sound evenly to create a more consistent listening or recording environment.
The WavePrism is sized to drop into a suspended ceiling grid or can be wall mounted using mechanical fasteners.
Auralex WaveLens’ open-boxed design scatters and redirects acoustical energy.
The WaveLens can be beneficial in numerous applications as it can create a “large sound” in a small room, as it can optimize existing absorption panels by redirecting the sound energy.
The QuadraTec’s unique tiered design provides excellent scattering properties, resulting in a warm, musical character to the dispersed sound.
The nested pair offers two unique diffusion tools that, when used in combination with each other, can result in a more spacious feel in any room.
The lightweight design allows QuadraTec diffusors to be placed in suspended ceiling grids or attached to wall surfaces with mechanical fasteners.
Auralex’s Peak Pyramid Diffusor is optimized to provide high-quality sound diffusion while also doubling as an effective bass trap when filled with absorptive material.
These lightweight, sturdy pyramid-shaped diffusors are sized so that they can be easily dropped into a suspended ceiling grid or installed onto wall surfaces using mechanical fasteners.
KeyPacs are magnet-based panels that have the ability to mount to Auralex ProPanels or Studiofoam. Since most rooms are not dedicated to one particular application, these perforated absorption covers allow users to have a more live or dry room on the fly without having to reconfigure the entire space.
Available in three configurations ― 9 Hole, Bubble and Star¬ ― KeyPacs can fine tune any absorption treatment to be more effective and create a more acoustically balanced space.
Additionally, Auralex’s Sustain Bamboo Sound Diffusors can be helpful in achieving LEED certification, specifically in the Materials & Resources category.
Auralex Acoustics Website
Community Launches New Distributed Design Series Of Ceiling Loudspeakers
The new high performance range of loudspeakers will be on display at AES 2010.
Community Professional Loudspeakers has announced the launch of its new Distributed Design Series of ceiling loudspeakers.
The line will be officially unveiled at the Audio Engineering Society Convention in San Francisco, November 5-7.
Designed for exceptionally high quality sound with very wide, uniform coverage, the series debuts Community’s Drop-Stop and Twist-Assist features, making the installation process easier, faster and safer.
Drop-Stop and Twist-Assist are features to aid installation, providing that elusive ‘third hand’ installers frequently wish for.
The Drop-Stop feature provides spring-loaded legs to support the back-can on the included rails and C-ring, so the installer no longer needs to hold the can against the tile while tightening the clamps.
Twist-Assist performs a similar function for the baffle assembly: when the loudspeaker back-cans are pre-installed into a conduit system for later termination, a simple twist of the baffle into the back-can brings the Twist-Assist clips together, supporting the baffle assembly while the installer fastens the screws.
The series consists of seven products, including five full-range, standard-depth back-can models. The D4 is a 60W, 4.5-inch model; the D5 and D6 are 5- and 6-inch 100W models; the D8 is an 8-inch, 150W design, and the D10 is a 10-inch, 200W model.
Also available is the D4LP, a low-profile, shallow back-can model that is only 3.6 inches (92mm) deep, for installations where available depth is limited. Completing the range is the D10SUB, a 200W 10-inch subwoofer that provides powerful LF reinforcement for entertainment applications.
Each of the full-range models is a true coaxial loudspeaker, with a compression driver concentrically located so that high frequencies emerge through the center of the low-frequency driver’s magnetic structure and cone via a precisely contoured Tru-Phase HF waveguide.
This special construction provides consistent, wide dispersion up to 16 kHz, all but eliminating high-frequency beaming. Further pattern improvement is achieved by precisely mounting the low frequency driver (and its concentric HF driver) dead-center in the baffle, as contrasted to the offset designs used by others.
“Most ceiling loudspeakers typically have a half-inch or greater ‘step’ in their baffle where the grille is attached, with the low frequency driver rear-mounted behind the grille plate,” said Community Founder and President Bruce Howze.
“These designs produce a host of unwanted reflections and diffraction effects. We’ve designed the Distributed Design Series with baffles less than 1/8-inch from the grille, creating an uninterrupted planar surface between the baffle and ceiling surface for diffraction-free HF driver acoustic loading and a smooth, predictable pattern. The result is optimal coverage without dead zones or overlaps.”
The Distributed Design Series incorporates Community’s Carbon Ring Cone Technology, which increases the effective surface area of the cone. Powerful magnetic structures and efficient coils ensure high output, resulting in greater headroom per power amplifier. And all models in the series comply with ETL standards, including UL1480/UL2043 and CSA60065.
Each model is available as a complete unit with everything needed for standard installations. The drive unit and face plate assembly and the back-can may also be purchased separately.
Community Professional Website
Tuesday, November 02, 2010
Mic Technique Basics For Musical Instruments
Tips on how best to apply your microphone technique within recording sessions.
Suppose you’re going to mike a singer, a sax, or a guitar.
Which mic should you choose? Where should you place it?
Your mic technique has a powerful effect on the sound of your recordings.
In this article we’ll look at some general principles of miking that apply to all instruments.
Which Mic Should I Use?
Is there a “right” mic to use on a piano, a kick drum, or a guitar amp? No. Every microphone sounds different, and you choose the one that gives you the sound you want.
Still, it helps to know about two main characteristics of mics that affect the sound: frequency response and polar pattern.
The frequency response of a mic is the range of frequencies it can pick up at an equal level (within a tolerance, such as +/- 3 dB).
Most condenser mics have an extended high-frequency response—they reproduce sounds up to 15 or 20kHz. This makes them great for cymbals or other instruments that need a detailed sound, such as acoustic guitar, strings, piano, and voice.
Dynamic moving-coil microphones have a response good enough for drums, guitar amps, horns, and woodwinds. Loud drums and guitar amps sound dull if recorded with a flat-response mic; a mic with a presence peak (a boost around 5 kHz) gives more edge or punch.
Suppose you are choosing a microphone for a particular instrument. In general, the frequency response of the mic should cover at least the frequencies produced by that instrument.
For example, an acoustic guitar produces fundamental frequencies from 82 Hz to about 1 kHz, and produces harmonics from about 1 to 15 kHz.
So a mic used on an acoustic guitar should have a frequency response of at least 82 Hz to 15 kHz if you want to record the guitar accurately.
Listed below are the frequency ranges of some instruments (including fundamentals and harmonics):
Male voice: 100 to 12 kHz.
Female voice: 200 to 12 kHz.
Kick drum and bass: 40 Hz to 9 kHz.
Guitar through an amp: 82 Hz to 4 kHz.
Acoustic guitar: 82 Hz to 15 kHz.
Cymbals: 500 Hz to 20 kHz.
Toms and snare drum: 100 Hz to 12 kHz.
Fiddle: 200 Hz to 15 kHz.
The polar pattern of a mic is a graph of how well it picks up sounds coming from different directions.
Omnidirectional picks up equally well in all directions.
Bidirectional (figure-eight) picks up best in two directions - the front and back of the mic.
Unidirectional picks up best in one direction - in front of the mic. Examples are cardioid, supercardioid and hypercardioid.
All else being equal, bidirectional and unidirectional patterns pick up less leakage, ambience and feedback than the omnidirectional pattern. Leakage is unwanted sound from instruments other than the one at which the mic is aimed.
Ambience is the acoustics of the recording room—its early reflections and reverb. The more leakage and ambience you pick up, the more distant the instrument sounds.
An omnidirectional mic picks up more ambience and leakage than a directional mic when both are the same distance from an instrument. So an omni tends to sound more distant.
To compensate, you have to mike closer with an omni. Some clip-on mics have an omni pattern. It can provide good isolation and good gain-before-feedback because the instrument is miked extremely close.
How Close Should I Place the Mic?
Once you’ve chosen a mic for an instrument, how close should the mic be?
Mike a few inches away to get a tight, present sound; mike farther away for a distant, spacious sound. (Try it to hear the effect.)
The farther a mic is from the instrument, the more ambience, leakage, and background noise it picks up.
So mike close to reject these unwanted sounds. Mike farther away to add a live, loose, airy feel to overdubs of drums, lead-guitar solos, horns, etc.
Close miking sounds close; distant miking sounds distant. Here’s why. If you put a mic close to an instrument, the sound at the mic is loud.
So you need to turn up the mic gain on your mixer only a little to get a full recording level. And because the gain is low, you pick up very little reverb, leakage, and background noise (Figure 1A).
If you put a mic far from an instrument, the sound at the mic is quiet. You’ll need to turn up the mic gain a lot to get a full recording level. And because the gain is high, you pick up a lot of reverb, leakage, and background noise (Figure 1B).
Figure 1. (A) A close microphone picks up mainly direct sound, which results in a close sound quality. (B) A distant microphone picks up mainly reflected sound, which results in a distant sound quality.
If the mic is very far away—maybe 10 feet—it’s called an ambience mic or room mic. It picks up mostly room reverb. A popular mic for ambience is a boundary microphone taped to the wall. You mix it with the usual close mics to add a sense of space.
Use two for stereo. When you record a live concert, you might want to place ambience mics over the audience, aiming at them from the front of the hall, to pick up the crowd reaction and the hall acoustics.
Classical music is always recorded at a distance (about 4 to 20 feet away) so that the mics will pick up reverb from the concert hall. It’s a desirable part of the sound.
Leakage (Bleed or Spill)
Suppose you’re close-miking a drum set and a piano at the same time (Figure 2). When you listen to the drum mics alone, you hear a close, clear sound.
But when you mix in the piano mic, that nice, tight drum sound degrades into a distant, muddy sound. That’s because the drum sound leaked into the piano mic, which picked up a distant drum sound from across the room.
Figure 2. Example of leakage. The piano mic picks up leakage from the drums, which changes the close drum sound to distant.
There are many ways to reduce leakage:
Mike each instrument closely. That way the sound level at each mic is high. Then you can turn down the mixer gain of each mic, which reduces leakage at the same time.
Overdub each instrument one at a time.
Record direct. Record an acoustic guitar off its pickup during tracking, then overdub the guitar with a mic. Record an electric guitar off its pickup during tracking, then play the guitar signal through a guitar-amp modeling plug-in during mixdown. Or record the electric guitar through a Line 6 Pod, which is a guitar-amp emulator.
Filter out frequencies above and below the range of each instrument.
Use directional mics (cardioid, etc.) instead of omni mics.
Record in a large, fairly dead studio. In such a room, leakage reflected from the walls is weak.
Put portable walls (goboes) between instruments.
Use noise gates on drum tracks.
Don’t Mike Too Close
Miking too close can color the recorded tone quality of an instrument. If you mike very close, you might hear a bassy or honky tone instead of a natural sound.
Why? Most musical instruments are designed to sound best at a distance, at least 1-1/2 feet away.
The sound of an instrument needs some space to develop. A mic placed a foot or two away tends to pick up a well-balanced, natural sound.
That is, it picks up a blend of all the parts of the instrument that contribute to its character or timbre.
Think of a musical instrument as a loudspeaker with a woofer, midrange, and tweeter. If you place a mic a few feet away, it will pick up the sound of the loudspeaker accurately. But if you place the mic close to the woofer, the sound will be bassy.
Similarly, if you mike close to an instrument, you emphasize the part of the instrument that the microphone is near. The tone quality picked up very close may not reflect the tone quality of the entire instrument.
Suppose you place a mic next to the sound hole of an acoustic guitar, which resonates around 80 to 100Hz.
A microphone placed there hears this bassy resonance, giving a boomy recorded timbre that does not exist at a greater miking distance.
To make the guitar sound more natural when miked close to the sound hole, you need to roll off the excess bass on your mixer, or use a mic with a bass rolloff in its frequency response.
If you mount a clip-on mic onto the guitar’s body, usually you can find a sweet spot that sounds natural and well balanced—but it’s not in the sound hole!
You want the mic to pick up a good mix of the body, sound hole and strings. The same principle applies to other instruments miked with a clip-on mic.
The sax projects highs from the bell, but projects mids and lows from the tone holes. So if you mike close to the bell, you miss the warmth and body from the tone holes.
All that’s left at the bell is a harsh tone quality.
You might like that sound, but if not, move the mic out and up to pick up the entire instrument. If leakage forces you to mike close, change the mic or use equalization (EQ).
Usually, you get a natural sound if you put the mic as far from the source as the source is big. That way, the mic picks up all the sound-radiating parts of the instrument about equally.
For example, if the body of an acoustic guitar is 18 inches long, place the mic 18 inches away to get a natural tonal balance. If this sounds too distant or hollow, move in a little closer.
Where Should I Place the Mic?
Suppose you have a mic placed a certain distance from an instrument. If you move the mic left, right, up, or down, you change the recorded tone quality.
In one spot, the instrument might sound bassy; in another spot, it might sound natural, and so on.
So, to find a good mic position, simply place the mic in different locations—and monitor the results—until you find one that sounds good to you.
Here’s another way to do the same thing. Close one ear with your finger, listen to the instrument with the other ear, and move around until you find a spot that sounds good.
Put the mic there. Then make a recording and see if it sounds the same as what you heard live.
Figure 3. Microphone placement affects the recorded tonal balance.
Don’t try this with kick drums or screaming guitar amps! You could also move a mic around while monitoring its signal with good headphones.
Why does moving the mic change the tone quality? A musical instrument radiates a different tone quality in each direction.
Also, each part of the instrument produces a different tone quality. For example, Figure 3 shows the tonal balances picked up at various spots near a guitar.
Other instruments work the same way. A trumpet radiates strong highs directly out of the bell, but does not project them to the sides.
So a trumpet sounds bright when miked on-axis to the bell and sounds more natural or mellow when miked off to one side.
A grand piano miked one foot over the middle strings sounds fairly natural, under the soundboard sounds bassy and dull, and in a sound hole it sounds mid-rangey.
It pays to experiment with all sorts of mic positions until you find a sound you like. There is no one right way to place the mics because you place them to get the tonal balance you want.
AES and Syn Aud Con member Bruce Bartlett is a recording engineer, audio journalist, and microphone engineer. His latest books are “Practical Recording Techniques 5th Ed.” and “Recording Music On Location.”
SSL To Showcase MADI-X8 Low-Cost MADI Router At AES 2010
The device delivers advanced eight port MADI router/splitter/aggregator functionality at an affordable price.
Solid State Logic will showcase the MADI-X8, a cost-effective 8-port MADI audio router/splitter/aggregator system at AES 2010.
“MADI-X8 is a product developed in response to requests from many of our clients for a system that enables audio to be distributed and shared throughout a multi room facility,” said Niall Feldman, director of new products for Solid State Logic.
“Delivering a low-cost, 512 cross point router with eight MADI I/O and clock distribution, which can be controlled from cross platform browser software from anywhere on a network, is a significant achievement.”
“This kind of technology has never been available at this price before, so it lowers the barrier to what can be achieved with an intelligent audio network in challenging multi room facilities.”
MADI-X8 is an affordable, powerful and versatile MADI routing system.
In addition to offering simple point to point bulk routing, MADI-X8 offers Source Distribution, Device Splitting and Source Aggregation.
MADI-X8 is therefore a unique low-cost solution to audio distribution and aggregation challenges in a wide range of applications, including large scale film/game scoring, multi-room studio complexes requiring real time asset sharing, broadcast production studios & OB vehicles and live sound.
The MADI-X8 is a 1U unit with six MADI fibre and two coax I/O. It provides a 512 x 512 point routing matrix controlled by cross-platform SSL Logictivity Browser software. The browser interface makes creation of most complex routing configurations straightforward.
The MADI-X8 hardware can store up to 128 presets and continues to function when the Logictivity Browser is disconnected. MADI-X8 can act as a Master Clock source or as a clock distribution system via its MADI and/or Word Clock connections.
It has Intelligent Clock Sensing with Auto Failover, which can monitor both a Primary Master Clock and a Secondary Clock source, and can switch automatically should the Master Clock source fail.
Up to four MADI-X8 units can be controlled by a single browser and multiple browsers can control a single MADI-X8 over a wired or wireless network.
Solid State Logic Website
Communicating With Non-Technical Parishioners
Honing your ability to communicate with non-technical individuals is just as important as your ability to dial in that killer mix.
I was reading an article a few weeks back about normal people who use computers and the power users (AKA IT guys) that support them.
The article talked about the routine problems normal people had using computers and how challenging it was for IT guys to help them.
Not because the IT guys couldn’t solve the problem, but because the IT guys couldn’t understand how the normal people ran into that problem in the first place.
For example, let’s say your browser is acting up. Most tech-oriented people know that you should try flushing your cache. Normal people, however, think cache is what you get from an ATM machine.
Or if a peripheral is acting up the IT guys might suggest updating a driver. To a normal person, a driver is someone behind the wheel of an automobile. Can you see the disconnect?
I’m about to tell you something you already know, but perhaps haven’t thought about this way before. What we do is highly specialized, takes years of training to master and requires a certain personality to fully comprehend.
You know that; but perhaps what you haven’t thought about is that the vast majority (and I mean 99%) doesn’t even begin to understand what we do.
To them, it’s all magic. This poses all number of challenges for the TD and volunteer techie alike.
Let’s say you get called into the women’s Bible study to help them figure out why they are having trouble with audio. You might look at the console and in 3 seconds determine that their gain structure is wrong, they’re not using a comp on the speaker’s mic and the EQ is adjusted improperly.
Now, if I told you that, you would immediately understand what I was saying and make the necessary changes. The poor gal sitting behind the console, however, is still stuck at gain what?
That’s what we’re faced with. And let’s not forget that the people we interact with are not stupid. They simply don’t work with the technology we do every day and because of that, we may as well be speaking a different language.
In fact, this past weekend, I tweeted, “Giving a few Audix heads a try this weekend. Really like the OM6 on our worship leader. Very smooth.”
A little while later, one of our pastors replied, “huh dude? Huh?” He caught me in the hall later and said, “I don’t understand almost anything you tweet.”
Now keep in mind, he’s a gifted pastor, teacher and great student of the Bible.
But trying to explain the difference between an OM6 and an RC35 is like me reading the Old Testament in the original Hebrew. It makes no sense.
I get this all the time from people on the worship team who’ve stumbled upon my writing. When they tell me they’ve been reading, it usually goes like this; “Most of the time I have no idea what you’re talking about, but it’s cool that you’re writing it.”
The longer I do this TD thing, the more I realize that while my understanding of technology is important, it’s equally (if not more) important that I understand how to relate to non-techies. And that’s hard.
I vastly prefer spending a few hours taking with my friends Jason & Dave about the latest compression techniques–conversations that can go on for hours, literally!–than trying to explain to a women’s group leader how to properly EQ an e6 for maximum gain before feedback.
However, as TD of my church, that’s exactly what I need to be able to do in order to really be successful at my job.
As much as I enjoy practicing mixing with virtual soundcheck, calibrating projectors, and focusing lights, I also need to give time to thinking about ways to make these incredibly complex systems more accessible to more people, and coming up with ways to explain to non-technical people how to use them effectively.
And few things will develop patience in us normally impatient TDs than explaining, for the fifth time, that turning all the EQ knobs all the way to the right is not considered a best practice.
So the next time you are tempted to get frustrated when having to explain a “basic” technical concept to a non-techie, remember to cut them some slack.
More importantly, remember that honing your ability to communicate with non-technical professionals is just as important as your ability to dial in that killer mix. After all, what we think is basic might as well be a lunar landing to others.
Are you doing a good job of communicating with the non-technical parishioners within your church? Let me know in the comments below!
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Community Loudspeakers Chosen By Pomona Fairplex
Community loudspeakers used across multiple projects within the consistently evolving venue.
From its rural beginnings in 1922 as the L.A. County Fairgrounds, Los Angeles County’s Fairplex has grown to become the fourth largest fairgrounds in North America.
At 543 acres the facility is twice the size of Disneyland and hosts a year-round variety of concerts, conventions, shows and sporting events.
The Fairplex grounds are home to more than 325,000 square feet of indoor exhibit space, including a 12,000 square foot Fine Arts educational center affiliated with the Smithsonian Institution.
The complex also houses carnival grounds, parks, plazas and picnic areas, as well as a 250-suite Sheraton Hotel and a 185-space KOA RV park. Also calling the Fairplex home is NHRA-rated Pomona Raceway and Fairplex Park, a 5/8-mile horse racing facility. Both racetracks also host numerous concerts and performance events.
Not surprisingly for a facility of its size, the Fairplex is constantly evolving, growing and modernizing its infrastructure, currently including a high-end revamping of the complex’s audio systems. Woodland Hills-based Campbell-Shaw Incorporated, dba CSI Multimedia (CSI) has been involved in the design and installation of these projects-along with Fairplex IT Manager Ralph Schorbach-including a powerful assortment of Community R-Series loudspeakers in various locations.
“We were originally approached to offer a proof-of-concept demonstration for one of the nine exhibition buildings,” said CSI President Rick Shaw. “Our Community representative, Steve McNeil of Mac West Group, set up three Community R.5 loudspeakers at one end of the hall, powered by an MC2 amplifier, and it outperformed the thirty or so horns they had been using for the same space.”
“They immediately asked us to do the racetrack and grandstands as well. Then we got to talking about the trams, and it’s been onward and upward from there.”
Convention Hall 9, the first building to be completed, is covered by 80 Community R.5-94TZ two-way full-range loudspeakers, with low end enhanced by six R.5SUB 12-inch subwoofers. Outside there are 16 R.25-94TZs and five R.5s.
An Aphex Anaconda digital snake system provides fiber optic distribution to Building 4, which is the communication hub for the exhibit halls.
Other ongoing projects include Building 4, one of the largest structures at Fairplex at 105,500 square feet, covered by 135 Community R.25s and 14 R.5SUBs. Convention Halls 5, 6 and 7 will receive a total of 320 R.25-94TZ units and 36 R.5SUBs, with another 86 R.25-94TZs and 20 R.5s covering the outdoor areas between buildings.
The Racetrack and 10,000-seat Grandstands have received 16 R1 loudspeakers and are slated to receive another ten R1s, with eight more in the auxiliary areas.
“A lot of the equipment in this facility had been designed 20 to 30 years ago, and much has changed since then,” said Fairplex’s Schorbach. “When you go from a system designed for 1,000 people to now handling 10,000, the old system just won’t accommodate those needs any more.”
“We needed a lot of volume and power to fill that capacity, which is one of the reasons we looked to Community loudspeakers. For the money, they gave us the raw power with a lot of overhead and great sound.”
“We knew the Community R-Series would be a good choice in terms of their power and range, but I think the client was mainly impressed with their voicing,” added CSI engineer Terry Galati.
“The sound is very consistent across the entire range, both up close and far afield, and they have a really pleasant voicing that people pick up on. Very few systems can deliver both intelligibility and musicality equally well, but the Community R-Series does it.”
“One thing that really impressed us was how accurate Community’s documented specifications were,” Galati added. “We hooked up an oscilloscope and did full power testing on the R1 and the R.25, running sine waves through it and bringing up the gain slowly.”
“We were able to ramp the speakers up to full power with no sloppiness and no high-frequency anomalies, and that’s very uncommon. Across the entire range, Community’s published specs agreed with our own test results.”
Another renovation included Barrett’s Horse Auction site and Hinds Pavilion, which have been transformed into the Finish Line Sports Grill, and Satellite wagering facility. The Finish Line Sports Grill is outfitted with 107 screens (plasmas, LCDs and projectors), along with 30 Community CLOUD6 ceiling loudspeakers for its main sound and two R.5s for focus on the lead-in area for when the auctions are held.
Three Aphex 230 Master Voice Channel Processors are also installed at the lead-in area. “The auctioneers just love what the 230 does for their voices,” says Shaw. Three Community iBOX iHP1264s and an iBOX i215SW above the plasma screens supply sound for the Hinds Pavilion, the auction and wagering area. Outside, ten R.25s cover the newly tented warm-up ring for the horses.
Tech Tip Of The Day: Runaway Volume At Live Shows
Controlling the volume over the course of the night.
Q: When I do live sound gigs I find that as the night wears on I seem to have to turn the volume up more and more to keep the energy level up and to just keep the volume the same.
Sometimes people say it gets too loud, but I don’t hear it that way. Are my ears lying?
A: Yes, but there is more to it than that. There are actually many, many factors at work here.
One of the more significant problems is a phenomenon known as Temporary Threshold Shift, which are your ears effectively turning themselves down for protection.
All engineers face this problem and it is very dangerous to your long-term hearing.
One way to help prevent volume runaway is to mix with earplugs and carry a SPL meter. I know, I know, ear plugs screw up the sound, you say. . .
Well, there are actually some good, though not perfect, custom ones you can get that keep things pretty well in order. However, at the very least, use a SPL meter to keep track of the volume.
There are other factors that may cause you to turn up the volume over the course of the night. Alcohol has the same effect on your hearing senses as it does the rest of you. Just say no to drinking on the job. This is totally unprofessional and, more importantly, unsafe.
Sometimes turning one or two instruments up in a mix (for a perfectly valid reason) will result in some other instruments being covered up a bit. The solution is not to come behind yourself and turn these other instruments up too.
You need to keep track of where you are in the mix and if you turn something up you should remember to turn it back down when it is no longer the focus.
I can’t tell you how many times I’ve seen engineers end up at the end of a night with all of the individual faders on their board almost all the way up while the master fader is relatively low. That’s a clear sign of an engineer that let it get away from him.
Even if you can’t hear what you are doing this should at least be a visual clue. It’s easier said then done, but try not to let the instruments fight each other. EQ them so each has its place. Sometimes you can turn up a guitar for a solo by simply adding a little upper midrange to the EQ and not touching the volume at all.
You just have to remember to take it back out once the solo is complete. Keep track of where you are and try not to mix by always turning something up. Sometimes it is appropriate to turn three things down instead.
Sometimes you really do have to turn up the volume over the course of a night just to keep it the same. As a room fills up with people, or dancers pack onto the dance floor (in front of the PA), they will absorb sound. If you intend to keep the volume close to the same in the back of the room you will have to turn it up some to compensate for this.
It also sometimes becomes necessary to add a little top end (high frequencies) to the house EQ. As the speakers in the PA heat up their impedance will rise a little bit. This makes them less sensitive and will result in more power needing to be applied to reach the same volume, at which point they heat up more, and so on…
There are other factors that contribute to this phenomenon, but your own ears are usually the big variable. Use objective measuring techniques (SPL meter) and trust the measurement.
As always, we welcome input from the PSW community and would love to know your thoughts on live show volume control. Feel free to let us know in the comments below!
For more tech tips go to Sweetwater.com
DPA 4017 Selected By Danish Film Recordist
Holmberg chose to use the 4017 shotgun on a new Susanne Bier movie.
Danish film location recorder Allan Holmberg has been using a DPA 4017 shotgun mic for Susanne Bier’s new film, The Revenge (Haevnen).
“I’d used the 4017 on a documentary I’d worked on, called Blekingegadebanden,” said Holmberg.
“The clarity and natural sound of the 4017 was a huge success on the documentary, so I decided to try the mic on this feature film.
“My only concern was that the mic pattern may be too wide, resulting in extra background noise and ambience, but after a short talk with Eddie Møller, the film’s supervising sound designer, we both agreed to give it a try.”
“The film contains some improvisation, and we wanted to avoid dubbing as much as was possible, so that was another good reason for using a wider mic.”
Once on set, any misgivings Holmberg had were quickly dispelled. “The 4017 performed excellently, and the ‘extra ambience’ I’d been concerned about wasn’t a problem at all,” he said.
“In the film’s more silent scenes, with sensitive dialogue, it was a thrill to hear how lifelike and crisply the mic performed. Every breath and change in the voice was recorded, and has made it into the film, making the dialogue and overall feel of the film a nice experience.”
After making English-language film Things We Lost In The Fire in the US, director Susanne Bier returned to her native Denmark to work on The Revenge. Bier also directed Oscar-nominated Danish film, After The Wedding.
Posted by admin on 11/02 at 08:15 AM
NOA Audio Solutions Helps Memnon Archiving Services Achieve One-Half Million Digitized Recordings
NOA's MediaLector ingest tools perform quality controlled mass migration of DATs and CDs at Memnon.
NOA Audio Solutions has announced that Brussels-based Memnon Archiving Services S.A. has digitized one-half million audio recordings relying on NOA systems.
Memnon customers include Danish Radio, Bibliothèque nationale de France (BnF), the British Library, Vlaamse Radio- en Televisieomroep (VRT), and the European Parliament.
NOA systems support multiple Memnon projects, including migration of 250,000 digital audio tapes for Danish Radio and 350,000 CDs for BnF.
Memnon has chosen to rely on NOA over competing systems because NOA’s Job Database is capable of managing 60 stereo channels in a single central workflow solution.
In addition, the failsafe quality assurance of the NOA JobDatabase instantly identifies a problem on a device and then enables an operator to find and fix quickly and easily any content that might have been affected.
“NOA is for me the ‘Google’ of digitizing within an archiving workflow,” said Nicolas de Beco, operation and quality assurance director at Memnon.
“Like finding a needle in a haystack, NOA enables us to search and find errors within thousands of performance recordings. And because results are managed in a central database workflow solution, it only takes a few clicks to reorganize the workflow and mitigate the problem.”
With offices and studios in Belgium and Sweden, Memnon is a leader in European archiving. Its multidisciplinary team of sound engineers, computer specialists, and documentation experts has more than 15 years of experience working with sound technology throughout Europe.
The company offers a full range of services for sound archives as well as audio digitization, restoration, indexing, and segmentation of sound archives.
“Since we acquired our first NOA system in 2003, we have continued to partner with the company and have integrated their systems within our own production asset management platform for most sound carriers,” said Michael Merten, Memnon managing director.
“The integration of NOA solutions within Memnon’s platform has allowed us to achieve high efficiency and impressive output at the highest quality, with low administrative work.”
At Memnon, NOA MediaLector ingest tools perform quality controlled mass migration of DATs and CDs under control of the NOA Job Database workflow system. NOA MediaLector software running on the N6073 host is capable of transferring eight stereo sources in parallel with up to 24-bit, 44.1-kHz/48-kHz quality.
Additionally, MediaLector monitors the status of the DAT drives’ error correction stages and annotates the quality-relevant status data synchronously to the audio stream with Sony PCM 7040 and NOA-adapted R500 DAT.
The result is a thoroughly documented audio transfer with a digitization proof stamp written to the database along with the audio material.
NOA Audio Solutions Website
Renkus-Heinz Loudspeakers Installed In North Dakota State Fair Grandstand
The upgraded sound system features Renkus-Heinz STX-Series loudspeakers.
The North Dakota State Fair in Minot is one of the largest events in the state, held in late July every year since 1922.
Attendance has consistently increased since 2000, attracting more than 300,000 people in 2010.
Dubbed “the entertainment capital of North Dakota,” the State Fair Center complex is home to the nine-day annual event that includes agricultural and livestock competitions, carnival rides and concessions, and concerts.
Acts featured in 2010 included Brooks & Dunn, Kiss, and Darius Rucker. Other events are held at the facility over the course of the year, including demolition derbies, drag races, and a collectors car show. And during the summer months, weekly stock car races are run on a 3/8-mile oval dirt track on the fairgrounds.
In August 2009, after funding was provided by the North Dakota Legislature, the Center’s old 3500-seat grandstand was demolished and construction began on a new $15 million grandstand.
The new 7000-seat facility features larger, more comfortable seating, a pub and other amenities, including an upgraded sound system featuring Renkus-Heinz STX-Series loudspeakers.
“They wanted a system that offered high output and good intelligibility, but was more than just a voice system,” said Dallas Anderson of Bismarck, ND-based Tricorne Audio, Inc., the company behind the design and installation of the new system.
A total of 12 of Renkus-Heinz’s STX4/94 three-way systems cover the seating area, affixed to the canopy some 80 feet above the audience.
While the speakers would not actually be used during the winter months, they would still be subject to whatever Mother Nature chooses to dish out, including snow, wind, rain, and temperatures that can reach 20 degrees below freezing.
To withstand the North Dakota winters, as well as the region’s summer heat and humidity, Anderson specified the full weatherization option for the speakers which includes a fiberglass coating and additional weatherization on the grill. Electro-Voice CPS Series amplifiers power the loudspeakers, with Biamp Audia Flex providing system drive and processing.
“The end result is nothing short of phenomenal,” said Anderson. “The sound is clear, intelligible and musical, with more than enough power to be heard over the din of the demolition derbies and stock car races.”
Monday, November 01, 2010
Neumann M 150 Microphones Selected By The Aspen Music Festival
Neumann tube microphones chosen in shootout among 30 microphones as Aspen enters a new phase of recording and broadcasting.
The Aspen Music Festival and School hosts one of the most famous music festivals in the world.
Since 1949, it has been fulfilling its guiding principle of fostering the growth of the human spirit through music, hosting more than 350 events every summer.
Recently, the Festival began to increase both the frequency and quality of its recordings and broadcast sessions, which are now featured regularly on American Public Media (APM) and other stations.
Working with Mike Pappas, owner of Pappas Consulting LLC, the engineers, technicians, guest conductors and music directors of the Aspen Music Festival began searching for a signature microphone to capture its performances and proceeded to blind test over 30 pairs of them, side by side.
“Our goal was to accurately reproduce the characteristics of the hall—we wanted to be able to capture the bloom that comes from the music tent.”
“We were also looking for precision in our imaging and wanted to make sure we had detailed and defined images,” recalls head engineer of Aspen Music Festival, Chris Cecere.
The Aspen Music Festival’s performances are mostly orchestral in content and can range anywhere from small chamber ensembles (3, 4 or 5 musicians) up to over 100 musicians on stage.
The tent, which has a capacity of just over 2,000 people, can be a challenging recording and/or broadcast environment given its immediate proximity to the outdoors and occasional swings in temperature. The chosen microphone would have to deliver both a quality sonic performance wile remaining reliability.
Over 30 pairs of microphones representing the world’s top manufacturers were positioned side by side onstage to help Cecere and his team to make an informed, objective evaluation.
Cecere recalls the moment of truth: “I had the assistant at the desk bring up the faders one by one—we listened to one pair, then another, then another; I had no idea what make or brand of microphones they were. When they brought up the Neumann M 150 Tube mic, the clarity and sense of space were all there. It was the clear winner, by far.”
Thanks to a gift from the Sidney E. Frank foundation, the Aspen Music Festival now relies on four Neumann M 150s across the front of the stage as their primary broadcast and recording microphones.
One pair, placed at center stage and spaced about 3 feet apart and 15 feet high, helps define the center image of the stereo field. The other pair, which is hung from a similar height, flanks both far sides of the stage.
Since the Neumann M 150s have been put in place, the response has been outstanding. “Everyone has been extremely happy with the sound quality of the microphones, which deliver a rich and full bodied sound character—this is why we bought four of them,” says Cecere.
“The M 150 makes the listening experience as authentic as possible and it has been fantastic to have a set up that we can rely on time and time again. We are very pleased to have chosen Neumann as our ‘signature sound’.”
Avid Announces Technology Demonstrations & Customer Presentations At AES
Butch Vig and Dave Hill lead line-up of main stage presentations
Avid has announced its line-up of technology demonstrations and customer presentations scheduled for the company’s booth at the 129th AES Convention from Nov. 5-7 at the Moscone Convention Center in San Francisco.
The sessions will provide attendees with a look at an array of new audio solutions that support open, collaborative audio workflows for use at home, in the studio or on stage.
Avid guest speakers will include Butch Vig, Garbage drummer and Grammy winning producer of Green Day’s “21st Century Breakdown,” and Dave Hill, founder of Cranesong and co-developer of Avid’s HEAT software, among others.
Sharing stories and best practices honed from years of professional experience, these speakers will share audio workflow tips, behind-the-scenes production details, and impressions about how Avid has taken what we’ve heard from our customer base and delivered on our promise to provide open, flexible solutions for interoperable production environments.
In addition, a series of main stage demonstrations will include:
Tours through the creation, tracking and mixing of a song while moving from laptop, to desktop, to studio platforms
Post production sound demonstrations based on HBO’s Emmy award winning series “The Pacific”
Technology previews of Avid’s MC Mix, part of Avid’s Artist Series of consoles (formerly Euphonix Artist Series) running on a Windows 7 system.
Rotating technology demonstrations on Avid’s main stage and show floor will showcase audio innovations for independent and studio professionals, featuring:
Open, flexible workflows in interoperable environments
Best-in-class audio solutions, including:
Pro Tools HD Native, the newest addition to the Pro Tools family that harnesses faster CPUs to deliver customers the industry-leading capabilities of Pro Tools HD with native, host-based performance
Pro Tools HD IO, OMNI, and MADI converters for best in class A-D/D-A conversion and the highest sound quality
HEAT software, a revolutionary software add-on that brings the warmth of analog components to the Pro Tools│HD mixer.
Mbox family of portable recording solutions, providing new A-D/D-A converters, cleaner preamps and improved drivers for Pro Tools and third-party DAW support.
Artist Series and Pro Series Consoles, streamlining production with award winning ergonomic control.
Also appearing at AES as part of the workshop and seminar series, Avid’s Robert Scovill will present as part of an Economics-Driven Change of Touring live sound seminar on Thursday, November 4 and a Live Monitoring and Latency with Digital Audio Networks workshop on Friday, November 5.
In addition, Avid’s Sheldon Radford will participate in an Audio/Video Bridging (AVB) product design panel discussion on 11/5. And in addition, Engineer Billy Bush and founder and CEO of Sonic Magic Studios and post mixer Jonathan Wales will be making an appearance on the Avid booth’s main stage, speaking to the strength and flexibility of Avid’s new HD solutions.