Wednesday, January 23, 2013

Radial Introduces Cherry Picker Preamp Selector

Radial Engineering is pleased to introduce the Cherry Picker, a unique device that enables the studio engineer to connect a microphone and select between four mic preamps in order to optimize the signal path and deliver the best sonic performance.

Radial Engineering Ltd. is pleased to introduce the Cherry Picker, a unique device that enables the studio engineer to connect a microphone and select between four mic preamps in order to optimize the signal path and deliver the best sonic performance.

Compact and easy to use, the Cherry Picker is a ‘straight wire’ passive switcher that features a balanced mic input with four outputs. Front panel ‘radio-style’ selector switches enable the engineer to select the active preamp while the Cherry Picker automatically shuts off the others.

Audio signal switching is actually performed using military-grade gold contact sealed relays. This ensures the signal will be transferred from input to output without introducing distortion, coloration or artifact of any kind.

Radial President, Peter Janis explains: “We noticed that many studios offer a variety of microphones and preamps to their clients. But switching between them to audition is often cumbersome as it usually requires muting the microphone, connecting cables at the patchbay and then readjusting the levels.

“The time delay between tests makes it practically impossible to remember one sonic signature versus another. The Cherry Picker lets you instantly audition and compare several mic preamps improving work flow efficiency.  Engaging the artist during the preamp selection process increases the comfort level of the artist and this usually results in a better musical performance, improving the sound of the track.”

To ensure quiet switching when using studio condensers, the Cherry Picker generates its own 48 volt phantom power for a stable supply to the microphone. This eliminates turning on and off the phantom power on each preamp.

Each output is equipped with a ground lift switch to help eliminate buzz and hum caused by ground loops.  A conveniently located front panel mute switch allows the Cherry Picker to be put on hold while the microphone is exchanged without having to readjust signal levels or mute preamps.

The Cherry Picker will start shipping in November 2012. Estimated retail price: $400 USD.

Radial Engineering

Posted by Keith Clark on 01/23 at 12:33 PM

Sennheiser MKH 8060 Shine on the Set of ABC’s Modern Family

MKH 8060s Provide Excellent Isolation, Versatility and Natural Response, Helping Technical Crew Deliver Outstanding Audio in Noisy On-Location Environment

Now well into its fourth season as a major hit ‘mockumentary’ series on ABC, Modern Family is regularly viewed by millions of viewers across America each Wednesday night.

The production crew regularly overcomes often daunting challenges to ensure the highest quality audio and video is captured. Recently, the team has begun to rely on several Sennheiser MKH 8060 microphones for many of its on-location shoots.

Stephen Tibbo, production sound mixer for 20th Century Fox Television, is responsible for capturing audio on the set for Modern Family — most importantly, capturing clear and pristine dialog among the characters. Since joining Modern Family in 2009, he has captured two Emmy Awards for his outstanding work as sound mixer.

The production set of Modern Family is anything but predictable. Therefore, from a production perspective, Tibbo and his crew need to be ready for just about anything.

For example, unlike a controlled studio environment with indoor sets that are more or less acoustically sealed, on-location shoots can present many challenges such as unwanted ambient noise and unexpected interruptions — such as impromptu truck deliveries. During this season, Tibbo recalls that the set of Yard Sale (episode six, in which there is a charity fundraiser for Manny and Luke’s school), was a particularly challenging one:

“This shoot took place at Jay and Gloria’s house, where the entire family is together and they are bringing things to the yard sale,” Tibbo recalls. “We had three days of shooting in front of their home but there were two construction sites literally across the street.

“There was also a local delivery truck bringing in pallets of drywall and wood - this was intermittently going on all day. ”

He says that achieving proper isolation from ambient noise was critical for the success of the shoot: “There was quite a bit of noise going on in the neighborhood, so we brought the Sennheiser MKH 8060s out — I have to say, they really shined and it was amazing.”

Following a meeting with Sennheiser’s Dave Missall last spring, the MKH 8060s have since earned a permanent spot in Tibbo’s mic pack. “Dave said, ‘Try these, I think you’ll like them,’” Tibbo recalls. “We were interior, and we saw a couple takes where we could put them through their paces to see how they sounded. And they ended up working quite nicely.” 

“I now use them on a regular basis, and they come in really handy in high ambient environments — especially outside,” he continues. In Modern Family — as any other high profile, prime time television show — there are simply no ‘second chances’ or ‘retakes’: “We have to shoot every single scene as a one-er,” Tibbo says. “You will have two or three cameras going and you get what you get. I have to pick one microphone for a given scene and make my choice on the spot.”

His primary mic choice for Yard Sale, and other episodes requiring isolation from background noise, has typically been the Sennheiser MKH 8060. “These are very versatile microphones, especially for rejection. I usually set them up on a boom, anywhere between 1 to 6 feet away from the actors.”

Tibbo also appreciates the low noisefloor and tight polar pattern of the MKH 8060s.

“Sometimes we do these very whispery scenes and I am really cranking it up,” he says. “The MKH 8060 is very quiet, and this is one of the reasons I use it outdoors. With other mics, you may hear a fan on in the distance, but with the MKH 8060 you only hear what you want to hear.”

In terms of its sonic character and response, Tibbo says the MKH 8060 is ‘punchy and warm,’ and a predictable performer: “It is a very natural sounding mic, both on or off-axis. On a show like ours where you have actors ad-libbing quite a bit, that is vital.”

In future episodes, Tibbo knows he can rely on the Sennheiser MKH 8060s —especially in noisy locations where multiple mics are required. “This is a great piece for my kit. In the case of the Yard Sale episode on Modern Family, we gave post-production a track they could work with. And in television, that is the name of the game.”


Posted by Keith Clark on 01/23 at 11:07 AM

Tuesday, January 22, 2013

TransAudio Group Introducing Drawmer DS101 500 Series Noise Gate At 2013 NAMM

Provides the frequency conscious gating functionality of Drawmer’s DS201

TransAudio Group will introduce the Drawmer DS101, the first noise gate module designed specifically for the 500 Series rack system, at this week’s 2013 NAMM Show in Anaheim (booth 6945).

The new DS101 is based on a single channel of the company’s renowned DS201 and incorporates the fully variable high- and low-pass filters and “key listen” monitoring functionality pioneered by Drawmer that enable frequency conscious gating.

The DS101, which can be used for both gating and “ducking” (for voice-over applications or the elimination of unwanted transient artifacts), incorporates comprehensive envelope controls — threshold, attack, hold, decay and range — plus a host of other features, including Drawmer’s “traffic light” metering.

The module’s low-noise circuitry also incorporates Drawmer’s proprietary hysteresis or “trigger stabilization,” which prevents “chatter” around the threshold level.

When installed next to each other in a 500 Series rack, two or more DS101 gates may be linked using Drawmer’s new infrared triggering mechanism.

Linked in this fashion, the envelope shaping capabilities of any number of DS101 gates (up to a maximum 10 units in a single rack enclosure) may be independently adjusted while the trigger pulse passes through unchanged. This enables the user to achieve an almost limitless range of “envelope follower” effects.

The DS101 additionally allows external triggering via a front panel 1/4-inch key input connector.

U.S. MSRP for the Drawmer DS101 is $449.

TransAudio Group is the U.S. distributor of Drawmer signal processing products.

TransAudio Group

Posted by Keith Clark on 01/22 at 09:23 AM
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Monday, January 21, 2013

ADL Releases Penteo 3 Pro And Appoints GC Pro As Exclusive U.S. Distributor

GC Pro continues to serve the post-production/pro audio markets by offering clients the new groundbreaking Penteo up-mix plug-in

Guitar Center Professional (GC Pro), has been appointed the exclusive U.S. distributor of a new plug-in from ADL, a developer of proprietary solutions in stereo-to-5.1 surround technology.

The plug-in, called Penteo 3 Pro, discretely converts stereo to 5.1, offering very fine control over sound image placement, thereby creating the warmest and best-sounding surround experience.

Designed for post-production, broadcast and DJ professionals, Penteo 3 Pro’s proprietary algorithm does not artificially manipulate the sound. The listener hears a perfect surround experience with zero sonic artifacts, and uniquely, Penteo is 100% ITU down-mix compatible to the original stereo.

Featuring an intuitive visual interface modeled after vintage gear, the plug-in offers six automated preset modes as well as manual fine-tuning for discerning ears. Penteo 3 Pro will be available beginning February 2013 online at and, with demos available at multiple GC Pro locations.

Jeffrey Read, President, Penteo Surround Inc., stated, “We wanted to partner with an organization that knows all the main players in the post-production and pro audio market sectors – one partner that could give us national coverage in major markets like L.A., New York, Chicago, Atlanta and Nashville as well as the smaller regional markets.

“We are thrilled to be represented by a company that understands the full spectrum of post-production and pro audio worlds. GC Pro is a fantastic organization that meets all of these criteria, and we look forward to a fruitful relationship with them and the wider Guitar Center organization.”

GC Pro Vice President Rick Plushner added, “We have seen a steady increase in demand for 5.1 content, especially among our post-production customers, so it feels very appropriate for GC Pro to represent Penteo. The sound quality offered by Penteo 3 Pro is astounding.

“GC Pro did extensive listening and testing with our engineering, sales and management teams, and to everyone’s ears, Penteo is a superb piece of technology. There are thousands of potential uses for this plug-in, and we are excited to see it put to great use. We also look forward to offering demonstrations at our locations around the country, especially our new high-tech demo facility in Hollywood.”

GC Pro

Posted by Keith Clark on 01/21 at 03:15 PM
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Rupert Neve Designs To Launch Two 500 Series Modules At 2013 NAMM Show

For the 2013 NAMM show, Rupert Neve Designs is announcing two new 500 Series modules: the 511 Mic Pre with Silk, and the 542 True Tape Emulator with Texture.

For the 2013 NAMM show, Rupert Neve Designs is announcing two new 500 Series modules: the 511 Mic Pre with Silk, and the 542 True Tape Emulator with Texture.

As two of the only 500 Series modules actually designed by Mr. Rupert Neve, the 511 and 542 combine the classic tone and versatility expected from a Rupert Neve design with the form factor and value of the 500 Series format.

511 Mic Pre with Silk
With a truly legendary preamp and the flexibility of variable Silk/Texture, the 511 provides exceptional sonic performance and flexibility for only $650 (list).

Incorporating the pristine preamp circuitry from the 517, the sweepable high-pass filter from the 5012, and the power of a variable Silk circuit derived from the flagship Portico II Channel, the 511 is a Rupert Neve-designed workhorse for any user’s most important tracks.

For company founder and lead designer Rupert Neve, the design process was far more intensive than copying existing designs.

“Although creating a functional 500 Series mic pre is relatively simple, designing those modules to equal their non-500 Series counterparts with the current, voltage and space restraints is quite challenging. In creating the 511 we experimented with a number of different transformer and circuit topologies to achieve the same presence and sweetness found in the Portico Series of modules.

“The result of these efforts is that outside of the slightly lowered headroom, the 511’s performance is nearly indistinguishable from standard Portico Series modules.”

The 511 can be used for either mic or line sources, and pairs perfectly with any ribbon, dynamic, condenser or tube microphone. A polarity reverse switch is available to conquer phase issues when using multiple microphones. The 12 dB/octave swept high-pass filter can be dialed-in to remove rumble with minimal artifacts, and also to control proximity effect in close-mic’d vocals or other sources.

To coax more rich harmonic content from the output transformer, the Silk Red mode can be engaged to add more thickness and sparkle in the high end as the Texture level is increased. While a little Silk Red can sound great on nearly any source, it is especially useful on dynamic and ribbon microphones that are inherently lacking in high-frequency energy and excitement.

542 True Tape Emulator with Texture
The second new module, the 542, is a follow-up to the acclaimed Portico 5042. As such, it delivers a remarkable simulation of classic tape sound through the inclusion of genuine tape drive circuitry while also incorporating a number of new methods for adding analogue color to individual tracks and mixes. List price is $895.

The 542’s “True Tape” emulation circuit provides the nostalgic rounding and compression typically achieved only through the use of tape, and can offset the harshness often found in digital recordings. Unlike digital emulations, the “True Tape” drive circuit works by feeding a tiny magnetic “record head,” which in turn is coupled to a correctly-equalized replay amplifier.

As the voltage rises on the “record head,” saturation increases, and a soft-clip circuit engages at higher levels to round off harsh peak transients. The sound of the tape circuit can be further modified with selectable 15 and 30 ips modes, providing a “saturation equalization” of sorts, and a pre/post-tape blend control.

In addition to the tape circuit, the 542 also includes the variable Silk/Texture circuitry found in the Portico II series of modules (with both much-loved Red and Blue modes), which allows the engineer to fine-tune the harmonic ratio and tonality of the output transformer.

For engineers, the 542 is an intuitive and dynamic tonal control. The non-linear qualities of the “True Tape” head, Soft-Clip and Silk circuits can be combined and tuned by simply adjusting the saturation, blend and texture controls. These effects can help breathe new life into sterile tracks and enhance performances with their dynamic response.

For example, a snare drum captured with a dynamic microphone that sounds anemic and “dead” in the high-end could be run through the 542’s tape and Red Silk circuits to simultaneously thicken the low end and sweeten the high end. If the snare is overly dynamic, the saturation knob can push the signal into soft-clip mode, thus reducing transient spikes. Additionally, the blend control can be used to preserve the drum’s natural dynamics and transient content even with more extreme applications of soft-clip and saturation.

Similarly, using two 542s across a mix, the gain staging can be optimized such that the tape circuit provides extra intensity and excitement in the loudest sections before final compression and limiting. Using this technique can help retain a more dynamic feel, even after the dynamics have been reduced, as the instances with the most compression correspond to the instances that have more pleasant harmonic distortion provided by the “True Tape” circuitry.

With the introduction of the 511 and 542 to the Portico 500 Series, which also includes the existing 517 Mic Pre/Compressor/DI and the 543 Mono Compressor, the 500 Series now has four of the most versatile tone controllers and mic preamps available on the market. In a world with numerous imitations, only the Portico 500 Series modules carry on the legacy of tone, quality and craftsmanship synonymous with the Rupert Neve name.

Rupert Neve Designs

Posted by Keith Clark on 01/21 at 02:51 PM
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Lectrosonics Excels On Location With Production Sound Mixer David Kelson

Wireless technology helps in overcoming surround noise, digital RF interference and little time for experimentation

Los Angeles, CA-based production sound mixer David Kelson is constantly challenged with the uncontrollable variables that come with his job, including surround noise, digital RF interference and little time for experimentation, which he overcomes with assistance from a range of Lectrosonics wireless products.

Having worked in location sound for the TV and film industry since 1982, some of Kelson’s recent projects include The Walking Dead on the AMC network (American Movie Classics) and the TV movie Steel Magnolias, and he is currently working on a feature film entitled Last Vegas with Robert DeNiro, Michael Douglas, Morgan Freeman, and Kevin Kline.

Specifically, he utilizes an assortmant Lectrosonics SM Series super-miniature transmitters, HM plug-on transmitters, Venue receiver systems (fully stocked with VRT receiver modules), UCR411 receivers, and for his IFB/Comm system, a T4 IFB transmitter and R1a IFB beltpack receivers. All of these units employ the company’s Digital Hybrid Wireless technology.

“I’ve been using Lectrosonics wireless microphone systems since 1992,” Kelson says. “I’ve always been very impressed with the sound quality, rock solid build quality, RF agility, and compact form factor of the gear.

“I also augment all of my Lectrosonics equipment with the company’s ALP500 shark fin and SNA600 dipole antenna systems, which go a step further toward ensuring the range and dropout-free performance so essential in my line of work.”

Currently, on the Last Vegas project, Kelson has been using a combination of SM, SMV, SMQV, and HM transmitters with his two Lectrosonics Venue receiver systems, which are set up in true diversity mode, where two receivers are set to work with a single transmitter.

He gave the example of a Steadicam shot that presented some real challenges on the motion picture.

“There’s an evening scene in Last Vegas where Douglas and Mary Steenburgen are walking on Las Vegas Boulevard in front of the Bellagio Hotel, and there was a phenomenal amount of RF interference from all the nearby shows in the area,” Kelson explains. “I used Lectrosonics’ one-touch SmartTune auto frequency selection feature on the Venue receivers to identify available frequencies in blocks 21 and 22.

“Once we identified those frequencies and locked them in, we achieved superb range with terrific audio quality. We set the transmitter power at 250 mW per unit, and using the Lectrosonics amplified antennas, we achieved very strong signal modulation, a surprising amount of range, and no dropouts whatsoever. The gear really delivered.”

To communicate with his two assistants, Kelson described his private communication system setup.

“My boom operator and utility tech each wear an SM transmitter that is outfitted with the company’s belt mounted mute switches,” he details. “These transmitters feed a pair of UCR 400A receivers that are summed by a small mixer and sent to the comm-in feed of my main mixer.

“I use my Lectrosonics T4 IFB transmitter to communicate with their R1a IFB beltpack receivers. This is a closed loop system where all three of us can communicate with one another privately, on the fly, and it’s been very effective.”


Posted by Keith Clark on 01/21 at 09:16 AM

Friday, January 18, 2013

In The Studio: Bruce Swedien On Developing Your Own “Sonic Personality”

Insight on finding a benchmark for our mind’s ear that has as its basic component true “reality” in sound

Excerpted from the excellent “Make Mine Music” by Bruce Swedien, available from

It’s my opinion that after all is said and done, psychoacoustics is really why we are interested in recording music in the first place.

Psychoacoustics can be defined simply as the psychological study of hearing. The true aim of psychoacoustic research is to find out how our hearing works.

In other words, to discover how sounds entering the ear are processed by the ear and the brain in order to give the listener useful information about the world outside. I’ve never felt that psychoacoustics is concerned with how sounds produce a particular emotional or cognitive response. That is another matter entirely.

To me, the three most fascinating areas of psychoacoustic analysis are:
—How does the human ear separate sounds occurring simultaneously (e.g., two musical instruments playing at once)?
—How do we localize sounds in space?
—How does the human ear determine the pitch of, say, a sound source or, more important, a musical instrument?

Psychoacoustics is not about sonic mind control. (I must confess that I was a little disappointed when I first learned that fact!)

Determining the abilities and limitations of human hearing is invaluable to us involved in the production of music recordings. Any resource that produces sound for the purpose of human listening should \ take into account what the listener’s ears are going to do with that sound, if we are going to take that resource to its utmost potential.

I have always been a very curious person. I have always had to know why things are the way they are, especially when it comes to the recording of music.

Sound is so important to us in so many different areas that it has always been fascinating to me to think about why we perceive sounds the way that we do. I have heard it said that the purpose of the ears is to point the eyes.

Knowing that, I think it is safe to say that the primary use of our sense of hearing is to localize sound sources. Keep these thoughts in mind the next time you are doing a mix.

Sound as a stimulus is the arena of the physicist. Sound as a sensation is in the arena of the psychologist. We, as professional music recording people fall somewhere in between these two areas of

In actuality, to be truly successful in music recording, we may have to be a little bit of both. So, what I hope to accomplish is to help you discover, with the help of the little bit of the psychologist that I think is present in all of us, your own “sonic personality.”

A Card-Carrying Record-Buying Junkie
I think the first step on the road to developing our own “sonic personality” is to find a benchmark for our mind’s ear that has as its basic component true “reality” in sound. From that stark, uncolored point we can then add a new viewpoint for the listener that we can call truly our own.

Many recording engineers and producers spend a lot of their time listening to and trying to learn their craft from records. In my opinion, this is a serious mistake and is precisely the reason why there are so few engineers and producers in the industry today, who have a truly unique sonic character to their work.

A certain amount of information can be gained by listening to other people’s records, but my problem with this approach is that one’s own “audio personality” is short-circuited.

Bruce Swedien at work (click to enlarge)

In other words, if you try to learn about music mixing by listening to records, in actuality what is happening is that you are hearing the music, or sonic image of the music, with someone else’s “audio personality” already imposed on the sonic image.

I do believe that it is true that we must listen to records to keep up with sonic styles and trends.

Personally speaking, I am a bona-fide, card-carrying record-buying junkie. When I hear a record on the radio or in a club that has interesting music or an interesting sonic hook, I am off to the record store in a minute and buying a copy for myself.

However, to have an “audio personality” that is truly your own, you must start your personal sonic development with a knowledge of natural, acoustical sounds.

Let’s Talk About Acoustical Support
To take that line of thought a step further, I think I should say that I feel that the best way to develop your ears’ “benchmark” is to hear good acoustical music in a fine acoustical setting. How many of you get out to hear live music on a regular basis? It’s very important!

Let’s talk about acoustical support as it relates to music. All music is conceived to be heard with some sort of acoustical support. This does not necessarily mean long concert-hall-type reverberation. It can mean very short, closely-spaced early reflections and minimal reverb content. Both of those components constitute acoustical support.

Once we know what music sounds like in a natural setting with good-quality acoustical support, we can then take that “audio benchmark,” and through our work, give our sonic images our own distinctly personal touch.

An engineer’s or producer’s listening ability does not descend on him in a single flash of inspiration. It is built up by countless, individual listening experiences. So let’s make a real effort to hear the music and sound with as open a mind as possible.

One of our most important abilities as a professional listener is judging balance. So let’s consider balance as the first thing to listen for today. The balance of an orchestra’s instruments in classical music is the sole responsibility of the conductor.

In our work of recording music, that responsibility is transferred to us. It doesn’t matter whether the orchestra is acoustical instruments or the orchestra is represented by a synthesizer. We must be able to judge balance.

Over a long period of time, if we have the native ability, we will develop a seemingly uncanny sense of hearing nuances of balance and sound that would pass unnoticed by the inexperienced.

This ability seems to be acquired almost by osmosis through thousands of seemingly insignificant listening experiences. This random approach is effective and vital.

The antithesis of balance is imbalance. When you are at a concert listening to good music in a good acoustical situation, listen for any imbalances that might be there. Think about your spontaneous reactions later.

When you are at a concert, ask for very good seats. That way, you should be able to judge balance and many other elements with a certain amount of accuracy. Listen for spectral balance. In other words, how well balanced is the frequency spectrum of the orchestra in that specific acoustic setting?

See how your ears and psyche react to the overall volume level of the orchestra, particularly at fff (extremely loud) dynamic levels. How does the orchestra sound at ppp (extremely quiet) dynamic levels?

Make sure that you have a good working knowledge of the different levels of musical dynamics and learn how they are expressed in musical terms. This will help you later on when you discuss these very important values with the musicians and composers that you will be working with.

Here are some important aspects of sonic values to listen for when you are listening to good music in a good acoustical situation:
—Listen for early reflections in the acoustical support of the hall. Listen for the reverb quality of that specific room.
—Listen for reverb spectrum.
—Listen for the amount of reverb that you perceive in relation to the direct sound of the orchestra – in other words, reverb balance.

Let’s Talk A Bit About Reverberation And Echo
Most of the time, we are unaware of how much of the sound that we hear comes from reflections from environmental surfaces.

Even when we are outdoors, a significant amount of sonic energy is reflected back to the ears by the ground and nearby structures – even by surrounding vegetation.

We only begin to notice these reflections when the time delay is more than about 30 milliseconds to 50 milliseconds, in which case we become consciously aware of them as individual sounds and call them echoes.

Special rooms called anechoic chambers are built as research rooms to absorb reflected sound energy. In a test situation staged in an anechoic chamber, only the directly radiated sound energy reaches the ears.

Upon entering an anechoic chamber for the first time, most people are astonished by how much softer and duller any sound source sounds. If reflected sound is so common in an ordinary acoustic environment, I’ve always wondered why these reflections don’t interfere with our ability to localize sound sources.

I guess it’s because our binaural hearing sense can quickly adapt to a new acoustic environment. I do know that our hearing system uses only partially understood mechanisms to suppress the effects of reflections and reverberation.

The fact that we localize sound sources on the basis of which signals reach our ears first is known as the precedence effect. This is not to say that we are unaware of the reflections that follow. Actually, we subconsciously use the subsequent reflections to estimate range, or the distance we are from the sound source. In my opinion, a music producer/engineer is no better than his tools.

Our main tools are, of course, a good pair of ears and the wonderful brain to which the ears are connected. If the hearing is faulty, only faulty judgments can result. Please try to remember that good hearing is a rare and wonderful gift.

How Do We Achieve Depth, Or That Third Dimension, In A Stereo Image?
The feeling of depth perception in a recording is the result of a combination of values, including the ratio of direct to reverberant sound. The intensity of a sound source relative to others in the same field, and even EQ, especially in the presence area of about 1.5 kHz to, say, 5 kHz.

Probably the most important factor in creating a feeling of depth is the change in the ratio of direct to reverberant sound. As reverberant energy becomes more prominent, the source appears to move back.

The absence of early reflections in a sound source makes it seem much closer. As you change the quality of early reflections in a soundfield, they greatly affect the depth of field. These reflections are generally less than 40 milliseconds.

When they are longer than that, the ear can pick them out as individual reflections, but below 40 milliseconds they tend to smear into one sound. Early reflections in a sound source must be part of the sound-field of the original recording to be effective. There are virtually no effects devices that seriously address this important issue!

Thinking out and carefully designing a sound-field yields big benefits. Careful thought and intent will make your work memorable and separate you from the also-rans.

That is also why intelligent use of pre-delay with reverb devices can give a tremendous feeling of depth of field. By increasing the length of the pre-delay of a reverb device (to make sure the reverb itself does not cover the early reflections), your recording will have a unique sonic character that is truly your own.

The Attack Wall
Here’s something about “tube traps” that you may find interesting. I am very excited about a new and intriguing recording-room acoustical treatment.

In essence, this new theory creates a reflection-free listening zone for music recording and mixing. The concept was perfected by my friend Arthur Noxon of Acoustic Sciences Corporation.

The “attack wall” at Westviking Studio. (click to enlarge)

It’s called the “attack wall.” It is a free-standing wall that surrounds the monitor speakers. (I think we could call this speaker position “midfield” monitoring.) It acoustically loads the monitor speakers, and causes them to play as if there actually mounted into a wall. This gives the monitor speakers increased acoustic efficiency.


With an array of studio traps behind the listening space, the “attack wall” makes a 100 percent acoustically “dead” space. This creates a reflection-free zone for music mixing and recording.
I have found that with the “attack wall,” no monitor EQ is necessary.

With good monitor speakers, you hear smooth, linear sound. The low end is exceptionally clean and articulate. One of the additional advantages of the “attack wall” is its portability. It can be moved from place to place with a great deal of predictability and reliability.

In my next article, I’ll be discussing speakers, amplifiers, control room volume levels and much more.

Click to enlarge book cover

This is an excerpt from Bruce Swedien’s Make Mine Music. To acquire a copy of this book, click over to  NOTE: ProSoundWeb readers can enter promotional code NY9 when checking out to receive an additional 20% off the retail price plus free shipping (offer valid to U.S. residents, applies only to media mail shipping, additional charges may apply for expedited mailing services).

Posted by Keith Clark on 01/18 at 05:29 PM
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Argentina’s Fort Music Records Latin Grammy Winner On Expanded API 1608

Under the advise of SL Audio, the API distributor in Argentina, Fort Music upgraded its existing 16-channel API 1608 to thirty-two channels with full automation.

Seven years ago Jorge Fort, a successful Argentinean businessman and a splendid trumpet player, recognized the need for a high-end recording studio in Argentina.

With the help of well-known Buenos Aires recording engineer Oscar Gimenez, he built and developed Fort Music to become one of the premier recording destinations in Argentina.

Under the advise of SL Audio, the API distributor in Argentina, Fort Music upgraded its existing 16-channel API 1608 to thirty-two channels with full automation.

The new arrangement helped propel trumpet legend Arturo Sandoval’s recent release, Tango, Como Yo Te Siento, to popular success and a Latin Grammy win for “Best Tango Album.”

“Lots of now-famous Argentinean records in the Jazz, Pop, Rock, and Tango traditions have been recorded on Fort Music’s API 1608,” said Daniel Paracha, who, together with Sergio Levinsonas, are on staff at SL Audio and assisted with the original purchase, as well as the more recent purchase of the 16-channel expander. “But it was in the hands of Arturo Sandoval that Fort Music received well-deserved recognition in the form of a Latin Grammy. Sandoval and Fort produced and engineered Tango, Como Yo Te Siento.”

The compilation of the world’s greatest tangos are elevated by Sandoval’s masterful vision and the skills of a vast ensemble of world-renowned musicians. Many of the recording sessions involved huge, multi-mic setups that were made possible by the 32-channel API 1608.

Its 500-series expansion slots are loaded with API processors and other VPR-approved units. Outboard gear includes an API 2500 Stereo Buss Compressor, and Pro Tools HD serves as the recording and editing platform.

“The recording sessions for Sandoval’s Grammy-winner consisted of renowned artists from many different genres,” expanded Paracha. “Indeed, there were as many as sixty-four musicians on any given song. Some moments feature fifty string players!

“The location is huge, and this is a recording without precedent in Argentina. A-list vocalists, such as Valeria Lynch, Patricia Sosa, and Raul Lavie, among others, also contributed to the project.”

Fort Music’s upgrade to thirty-two channels makes it the owner of Argentina’s largest API console and is a decisive factor in attracting new artists and producers to work at the studio.


Posted by Keith Clark on 01/18 at 01:15 PM

Wednesday, January 16, 2013

In The Studio: Mixing With Mastering In Mind—Dynamics

Strategic ‘whole project’ goals to serve the music
This article is provided by the Pro Audio Files.


There are a lot of different ways to conceptualize ‘dynamics’ when we begin to think about a contemporary recording project. Some of them are powerful, and some of them are pointless.

This article points out a few useful techniques for addressing dynamics during mixing, with mastering in mind.

Let’s start by considering dynamics at the largest most ‘zoomed out’ level. The fundamental question is whether it is musically appropriate for an album to be loud compared with other albums.

Maybe So
If overall loudness is the goal (and a smart one), know that your mixing decisions will have to rely on tonal dynamics to preserve and convey musical expression. Tonal dynamics are the natural changes in tone over the course of a phrase, or between different song sections.

In the case of really loud music, level dynamics within the mix (and inconsistent approaches to level dynamics from mix to mix) will work against an interesting execution of loudness in mastering.

Counter intuitively, this means that mixing a ‘loud’ record should rely as much as possible on simple level controls, as opposed to trying to control audio dynamics with compression. Ride the faders and automate. Heavy dynamics processing will usually moderate or distort the natural changes in tone that tell us exciting useful things like, “the singer is screaming now,” or “that’s a really nasty guitar part.”

Mutes are another useful tool for manufacturing a sense of change-over-time in mixes that are destined to be squeezed a bit. Reserve non-essential arrangement elements like keyboard and guitar layers, ad-libs, and pads for later re-iterations of the same musical form (i.e. the next verse or hook). These types of elements won’t really make the track jump up to a new loudness, but their addition will create tonal complexity and interest.

Finally, if you’re trying to effectively mix for loud or loudish mastering, be sure to consider the intersection of these two questions:

1) What is the loudest element of a typical mix in this style of music?

2) What is the brightest, most high frequency-focused element of my mixes going to be?

Most importantly, don’t let your answer for these two questions be the same. If your genre calls for loud vocals, let the drums, for example, be the brighter element. If the drums are the loudest element, be careful about how much non-essential material is presented above 10 kHz or so. Let the guitars, keys, or other harmonic elements shimmer a little more in comparison.

These simple approaches can be a great set up for intensity if they are consistently executed across each mix in a project.

Maybe Not So Loud
If weaponized loudness doesn’t make sense for the music, then the level dynamics within musical phrases, and terraced levels between song sections become really useful tools for creating a powerful musical experience. The key is to make some smart choices about how much level change is expressive, and execute those choices consistently across all of the mixes.

One approach is to setup dynamic mixes from the highest level down. In other words, start mixing the tune at the top of its dynamic peak, like the big bridge or the most intense hook. Rough in each element at a level that works for its loudest passage. From that point, a lot of the work of mixing becomes pulling things up into focus when appropriate.

Instead of making sure the intensity doesn’t get too intense, this approach establishes the most expressive, full mix moment as the benchmark. Bounce reference mixes so you can compare these high points as you’re setting up subsequent mixes for the same project.

Not every recording project is destined to end up in a specialized professional mastering studio. In either case, mixing with strategic ‘whole project’ goals is a great way to make sure your mixing is serving the music.

Rob Schlette is chief mastering engineer and owner of Anthem Mastering ( in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.

Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.

Posted by Keith Clark on 01/16 at 05:49 PM
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IK Multimedia Announces iRig HD

The sequel of the most popular guitar interface of all time for iPhone, iPad, iPod touch & Mac from IK Multimedia.

IK Multimedia is proud to announce iRig HD the new digital guitar interface for iPhone, iPad, iPod touch, and Mac. iRig HD is the next generation of the popular iRig guitar interface adapter.

iRig HD is a high-quality digital guitar/bass/instrument interface that allows users to plug their guitar or bass into their iPhone, iPad, iPod touch or Mac. It can be used with the AmpliTube range of guitar amps/effects apps and software or any other real-time processing app/software, like GarageBand and more.

Using AmpliTube, users can play with the sound of their favorite amplifiers and effects, record their performance and compose entire songs, everywhere. iRig HD features crystal clear digital signal thanks to its superior 24 bit converter, an onboard gain control for perfect level setting, a low power consumption circuit for longer device battery life, plus an ultra-slim design and interchangeable adapter cables for maximum portability and universal compatibility. As many other IK accessories, iRig HD is also made in Italy for the highest possible manufacturing quality.

iRig HD is a simple “plug in and play” interface featuring a 1/4-inch instrument input jack, that plugs directly into the digital Lightning or 30-pin connectors of iPhone, iPad or iPod Touch, using one of the included cables. Players can easily connect iRig HD to any Mac with the included USB cable. With its ultra-compact design, iRig HD fits easily into any guitar case, gig back or pocket for true mobility.

For a complete out of the box playing experience, iRig HD comes with the AmpliTube Free App (downloaded from the iTunes App Store: that provides a great sounding “ready to go” expandable guitar rig complete with effects pedals, amplifiers, speaker cabinets and a single track recorder.

Players can play, practice and record anywhere and everywhere on their iPhone, iPad or iPod touch. Plus, AmpliTube effects, amps and gear selection can be greatly expanded via in-app purchase with the entire range of AmpliTube apps including AmpliTube, AmpliTube Fender, AmpliTube Jimi Hendrix and AmpliTube Slash.  Guitar and bass players now have the widest range of amplifiers and effects at their disposal with over 50 superior sounding mobile gear for jamming and recording on the go.  Plus, iRig HD can be used with any app that supports digital audio processing, like GarageBand and many more.

Not only is iRig HD the perfect interface for the guitar and bass players on the go but can also be used on any Mac to take advantage of the superior processing power of the Mac platform. Users can download AmpliTube Custom Shop for free or use it as a standalone amp and effects processing powerhouse, or with many popular digital audio workstation program (DAW), such as GarageBand or Logic, as a plug-in for more sophisticated recording and composition. AmpliTube Custom Shop gives access to hundreds of top quality amplifiers and effects from world’s top manufacturers like Fender, Ampeg, Orange and more that can be purchased a-la-carte for a truly customizable software rig.

iRig HD features at a glance:

• High-quality instrument-level 1/4” Hi-Z input jack
• Detachable cables for Lighting, 30 pin and USB connector compatibility
• Preamp gain control
• High-quality low-noise, high-definition preamp
• High-quality 24 bit A/D conversion
• Powered by the iOS device or USB
• Ultra-compact and lightweight
• Comes with AmpliTube apps

Price and availability

iRig HD will be shipping in the Spring of 2013 and will be available in musical instrument and electronic retailers worldwide. Price is to be announced.

IK Multimedia

Posted by Keith Clark on 01/16 at 02:35 PM
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Fostex Introduces TH-900 Professional Headphones & HP-A8C DAC/Amplifier

Sold separately but optimized to work together

Fostex has introduced the TH-900 dynamic headphones and the HP-A8C 32-bit DAC/headphone amplifier, sold separately but optimized to work together.

The TH-900 headphones include specially engineered driver unit with a 1.5 tesla magnetic circuit that allows the unit to achieve significantly wider range, as well as the company’s proprietary “Biodyna” 50mm biodynamic diaphragm for purity of sound reproduction.

The headphones are crafted with Japanese Cherry Birch housings, which offer a rigid and extremely dense texture that creates a robust acoustic performance. Style embellishments include housings finished in a traditional Japanese (“Urushi”) lacquer in brilliant Bordeaux red by a 100-year-old artisan group.

The unit’s 3m cord with a gold plated 6.3mm stereo plug is also engineered for maximum durability using 7N grade oxygen free copper (OFC). The TH-900 ships with a headphone stand.

The HP-A8C is a 32-bit Digital Audio Converter and headphone amplifier offers DSD audio playback function, as well as a high-precision electronic volume knob and TCXO clock for Asynchronous mode.

Additional features of the HP-A8C:

• Complies with USB Audio Class 2.0 up to 32bit/192kHz (24 bit on Windows)
• Large capacity toroidal power transformer
• Built-in SD (SDHC) card drive for epoch-making DSD file (DSF format) reproduction in addition
to the firmware update
• Selectable between the internal clock and the external clock (except for USB and SD card)
• Built-in up-sampling function of x2 and x4
• Digital Filter selection between the conventional “sharp roll-off” and the “minimum delay” developed by Asahi Kasei to eliminate pre-echo
• Direct Out mode by passing the volume control circuit
• Variable headphone amplifier’s gain from 0 dB to -12 dB by 0.5 dB step for perfect match with any type of headphones
• Various inputs including USB, AES/EBU, Coaxial, Optical (x2) and analog RCA
• A dedicated Remote Controller is supplied

Style features such as the enclosure with a glass front panel and sleek black side panels and striking accents like the pure white Organic LED display enhance the aesthetic value.

The TH-900 and HP-A8C are separately priced at $1999.99 each and are available now at select authorized US Fostex retailers. Go to American Musican And Sound (here) for a list of retailers.

American Music And Sound

Posted by Keith Clark on 01/16 at 12:47 PM
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Tuesday, January 15, 2013

Harman Professional Names Jaime Albors Sr. Director Of Global Sales Operations

Tasked with oversight of all Harman global sales office operational activities

Harman Professional has named Jaime Albors to the position of senior director of Global Sales Operations. 

Reporting directly to Scott Robbins, Harman Pro executive vice president of Worldwide Sales, Albors is tasked with immediate oversight of all Harman global sales office operational activities.

He previously held the position of senior director, Sales, Intercontinental Region, where he headed operations of Harman’s San Juan, Puerto Rico sales office and was responsible for working closely with channel partners and dealers in Canada, Mexico, Central and South America, and The Caribbean Islands. 

He also previously worked with Crown Audio in Elkhart, IN, and holds a BA in Management from Ashford University in Clinton, Iowa.

“Jaime has consistently been among our top sales performers for many years,” says Robbins. “He has a deep technical understand of our products and understands the importance our customers place on digital technology and system integration, and the increasingly mass deployment of affordable, reliable networking.

“Jaime Albors has long recognized this fact and, for me, he is the model sales leader with a unique capacity to listen to the needs of his partners and customers while providing them with the systems and tools they need to be successful.  I am very pleased that Jaime has accepted this expanded role and I look forward to him applying his experience on a global level.”

Albors adds, “Harman has market-leading technologies and systems that provide unique value to customers in a host of professional markets around the globe.  Our message has resonance among channel partners and customers. Whereas I have enjoyed my role leading Harman’s Intercontinental sales initiative, I look forward to the new challenge of managing Harman Professional’s International global sales operations.”

Harman Professional

Posted by Keith Clark on 01/15 at 10:47 AM
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Call For Registration: Central Indiana Audio Student Workshop 2013

Will be held free of charge to give all students equal access to top-flight audio instruction

The Central Indiana Section of the Audio Engineering Society is hosting the Second Annual Central Indiana Audio Student Workshop on February 16, 2013 on the campus of Indiana University – Bloomington in the Recording Arts Department studios.

The workshop aims to provide audio students of all ages and working professionals within the state the opportunity to meet, hang out, and improve their skills. Experienced presenters will be focusing on topics that include recording, mixing, live sound, and acoustics.

The event is open to anyone interested in audio, including local professionals and high school students.

With the support of Title Sponsor Solid State Logic, as well as other sponsors, the Central Indiana Audio Student Workshop will be held FREE of charge to give all students equal access to this top-flight audio instruction, as well as free lunch.

Space is limited, so advance registration is required. All registrants are also entered in a drawing for prizes that include SSL plugins, JBL monitors, Auralex studio foam, custom Sensaphonics earplugs, and more.

For additional information and to register, go here.

AES Central Indiana Section

Posted by Keith Clark on 01/15 at 09:21 AM
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Monday, January 14, 2013

Argosy Debuts New Line Of Spire Speaker Stands For Studio Monitors

Safe, sturdy solution for monitor isolation

Argosy Console has debuted the new line of Bioacoustics-enhanced Spire Speaker Stands, which provide an attractive and stable solution for studio monitor isolation.

Two models are offered in the line—the Spire 420i with a height of 42 inches, and the Spire 360i, which is 36 inches tall.

The Spire Stands, which are comprised of 5/4-inch powder-coated substrate for stability and resilience, offer a modern design with die-cut side panels and beveled edges to present high-tech look and feel.

The patented Bioacoustics technology is designed to keep all movement on-axis to move in the direction of the loudspeaker cones’ travel while resisting movement in other directions. The result is that when any monitor is placed on a Spire 420i or 360i, there is a tightening of the low end as well as improved stereo imaging.

Dimensions of both Spire Stands are 18 x 18 inches at the base, 7.75 x 9.9 inches at the top, and again, with heights of 36 inches (360i) and 42 inches (420i). The Spire 360i is available for $379 per pair and the 420i is available for $399 per pair.

The new Spire Speaker Stands will be on display at the upcoming Winter NAMM show at both the Bioacoustics Booth (Hall E, 1631) as well as the Waves Booth (Hall A, 6824.)


Posted by Keith Clark on 01/14 at 06:05 PM
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JoeCo Adds Features To BlackBox Family

Latest JoeCo software update supports polyphonic wav files and provides 64-channel Player option for MADI and Dante systems

JoeCo’s latest software update for the award-winning BlackBox brings several new features and enhancements to the multi-channel live audio recorder/player series.

With version 2.2.7x, both the BlackBox Recorder and the BlackBox Player now support the replay of polyphonic wav files, from multiple stereo files up to 24 or even 64 channel PolyWAV files. Mono, stereo and multichannel files can even be used together in the same song massively enhancing the system’s flexibility.

JoeCo’s iXML plug-in also enables PolyWAV files to be recorded – a feature aimed at TV and film location recordists to better serve the needs of post production facilities, or systems where audio material is ingested in single large files.

The ability to import polyphonic wav files from a DAW has also brought BlackBox Player capability to JoeCo’s 64-channel MADI and Dante systems. Previously available as a dedicated 24-channel playback option for the BlackBox BBR1 series, BlackBox Player software can now be installed on BBR64-MADI and BBR64-Dante systems enabling 64 channels of 24bit/48kHz audio to be simultaneously replayed and triggered via footswitch, QWERTY keyboard, time code, MIDI and SONY 9-pin.

The soon to be released JoeCo BlackBox app for iPad will provide a further control option for triggering backing tracks and multiple surround stems for live shows and themed entertainment.

Further BlackBox Player enhancements resulting from the recent software update include the implementation of a “Chase” mode enabling the unit to commence playback in sync when it receives incoming time code. This option can either be controlled from the front panel or via newly extended SONY 9-pin support. The extended 9-pin commands also allow the user to rehearse shows that are normally locked to time code via remote control and are compatible with most edit suites and show controllers. On the BlackBox Player and Recorder, time code can now be shown on the main time display in both record and playback modes.

Other new BlackBox features include more accurate metering and smoother headphone monitor output, especially when changing levels. An additional “mix” feature has also been added to the headphone output on the 64-channel MADI and Dante systems.


Posted by Keith Clark on 01/14 at 05:47 PM
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