Thursday, December 10, 2015
Extron Now Shipping 4K HDMI Matrix Switchers With Audio De‑Embedding
HDCP compliant units support data rates to 10.2 Gbps, Deep Color up to 12‑bit, 3D, and HD lossless audio formats.
Extron Electronics announces the immediate availability of the DXP 1616 HD 4K and DXP 168 HD 4K, the first two models in a new series of HDMI matrix switchers for resolutions up to 4K.
They are HDCP compliant, and support data rates to 10.2 Gbps, Deep Color up to 12‑bit, 3D, and HD lossless audio formats.
Extron technologies such as SpeedSwitch, Key Minder, and EDID Minder, along with automatic input cable equalization and output reclocking, ensure dependable system operation with switching speeds and compatibility between devices.
These 16x16 and 16x8 matrix switchers also feature built-in audio de-embedding, enabling digital audio from any input to be assigned to the digital or analog stereo outputs for streamlined integration.
The DXP HD 4K Series is ideal for use in applications that require routing of digital video and digital or analog audio signals in professional AV environments.
“Many designers are expected to deliver a sophisticated fixed I/O solution for 4K video systems that provides high reliability while also being easy for the end user to operate,” says Casey Hall, vice president of sales and marketing for Extron.
“Loaded with proven technologies from Extron, the DXP HD 4K matrix switchers provide rock solid, install-and-forget reliability for routing 4K video and independent audio within the meeting room, lecture hall, or control center.”
The DXP HD 4K Series also switches embedded digital audio from HDMI source signals, along with the corresponding video, to any or all of the selected outputs. The technologies and capabilities built into the DXP HD 4K Series ensure high performance AV signal routing, with a fully digital pathway that maintains high audio and image quality for multiple sources and displays.
To watch a product introduction video, please click here.
Posted by House Editor on 12/10 at 12:09 PM
Engineer David Kimmell Finds His Sound With Dangerous Music
Indianapolis studios, Masthead returns to the feel of live mixing with D-Box analog summing and BAX EQ.
David Kimmell has been mixing audio since 1997, and for many years mixed bands live, among other touring gigs. After several years in his own studio trying to mix ‘in-the-box’ and not enjoying it, feeling that he was “fighting the mix constantly,”
Kimmell added the Dangerous Music D-Box to his rig and everything changed.
“The light went on. I felt like, ‘This is how it should be,’ this is why I had so much fun mixing live, it just sounded good, and I didn’t have to fight anything,” says Kimmell.
Kimmell has mixed a wide variety of music styles over the past year at his Indianapolis studios, Masthead Audio, including tracks for the progressive rock group “Ladymoon” (where Kimmell recorded the band live and then mixed), the bluegrass groups “Whipstitch Sallies” and the “Flatland Harmony Experiment”, while in the EDM space he’s mixed and worked with “Digital Tape Machine” and “Turbo Suit”, and then there’s the experimental Hip Hop group “DAM!”.
Kimmell comments, “I like the variety. That was my favorite thing about mixing live, almost every show was completely different.”
“The D-Box and BAX EQ are both integral parts of my studio configuration,” Kimmell says.
“The basic nuts and bolts of my studio are a MacBook Pro running Digital Performer and Glyph hard drives, Waves, Melodyne, and other plugins, my hardware includes MOTU, Eventide, and Dynaudio and Yamaha monitors all with Mogami cabling. As far as how I use Dangerous Music gear, in short, it makes mixing fun again.”
Kimmell explains, “I come from the live world, and mixing on a killer analog desk is a joy. For years I tried to capture that feeling and reproduce it in the studio, but once I tried to mix everything down in the computer -in-the-box-it became work, and the joy was lost. When I got the D-Box it removed that wall. Now I have the best of both worlds. The power of the computer matched to the sonic chocolate that is analog, and that’s just the summing. Add in all the other features of the D-Box and it truly ties the studio together.”
Digging into his mixing experience with the D-Box he adds, “When mixing in-the-box, pretty much every time I’d add something into the mix I’d have to go back and change other things to make it work. Using the analog summing in the D-Box it alleviates that problem so I get back that give, the headroom, there’s not that brick wall at the top like there is in-the-box, in the computer. With analog summing I can just put up all the tracks and mix like I was used to doing. With the D-Box and analog summing: it sounds good, lets go.”
Kimmell uses the BAX EQ as the last stage in mixing before recording the stereo mix from the D-Box’s Sum Output back into his interface’s A-to-D.
“The BAX EQ is the next piece of the puzzle. Through mastering music of different genres I’ve found mastering plugins to be sterile sounding. The BAX reduces my need for plugin EQ correction while sounding infinitely cleaner and more musical than software EQ. Plus, the BAX’s high and low pass filters make the A-to-Ds color things less, giving me more accuracy in the conversion.”
Kimmell reveals, “I take the stereo mix from the analog summing into the BAX to clean up the signal and get rid of some of the artifacts at the bottom and the top, and put a little bit of sugar and shine on it, a little bit of punch before it goes back into the computer. I think that the BAX really helps the A-to-D converters work much more efficiently.”
“One of the other best things about having the D-Box is that I have an easy switchover to my other monitor set, plus I use the mono button to check the mix, it gives me some great options as far as monitoring,” he says.
Another favorite feature is the digital input, “I can actually monitor what’s going on when the mix is finished. I run stems of audio into the analog summing, the stereo mix comes out of the D-Box and into the BAX, and then back to my session through the A-to-D. I can listen through the D-Box’s onboard D to A converter for exactly what it sounds like-after the A-to-D, and after I’ve done some mastering to it. So I can hear more of what the final product is going to be. With one switch, I can listen between those two different things, I can tell where I am at with the analog summing and also listen to where the mix is at in the digital world. It lets me know exactly what’s going on-it’s a great tool to have. It makes mixing very simple.”
“Another benefit of the D-Box is that I have all those monitoring controls in one place,” adds Kimmell. “Like speaker switching, input monitoring for analog or digital inputs, and the headphone amp-which is definitely top of the line sounding!-it’s all in one unit. It used to be that the interface was the hub of the computer studio, at this point it’s the D-Box. It’s where I go first, I barely touch the interface anymore.”
On doing some basic mastering for his clients as well, Kimmell says, “I feel that an engineer should be able to give a client the entire process. From that first musical note to something they can put out to the masses. I think the ‘beginning to end’ cohesiveness is a part of what we’ve lost from classic recordings. To further that ideal I’ve begun to master more and more of my clients’ material-providing even more value to them. The BAX EQ really helps my mastering setup.”
Concluding his thoughts as a ‘wish list’ Kimmell says, “I can’t wait to get the Dangerous Compressor next. I wish I had everything Dangerous Music makes. [laughs].
Posted by House Editor on 12/10 at 11:50 AM
BAE Audio Releases New 10DCF Compressor/Limiter
Featuring Carnhill and Jensen transformers with a new low frequency bypass filter that engages at 50, 80, 160 and 300 Hz.
BAE Audio announces that its new 10DCF Compressor is shipping and is available at select authorized dealers following its unveiling at the 139th AES Convention, held in New York City last month.
The 10DCF, which builds on the capabilities of BAE Audio’s10DC compressor/limiter, features Carnhill and Jensen transformers, all discrete circuitry while adding a brand new, inductor-based bypass filter.
The new bypass filter delivers increased flexibility while recording low frequency ranged instruments, and engages at 50, 80, 160 and 300 Hz — perfect for users who want to compress a broad frequency range while leaving lower frequencies uncompressed.
The 10DCF units are also stereo linkable, making them the perfect complement for the output stage of a mixer.
“Our new 10DCF is an incredibly versatile tool for both tracking and mixing,” commented Mark Loughman, president of BAE Audio.
“The unit retains the same top grade, Class A circuitry that is present in our other gear, while providing bypass filter options that make it more customizable, especially useful for drum and bass applications.”
Aside from its sound quality, the 10DCF also features a useful range of features on its front panel, including Elma stepped switches for each control, BAE Audio’s trademark Marconi knobs, and a gastank style analog meter. The combination of these elements provides users with accurate visual and tactile reference points as they adjust and shape their sounds. The metering on the 10DCF incorporates a simple design, with easy to read white lettering set against a black background in a gastank style encasement.
The 10DCF, which is available now and priced at $2,100 (including power supply), is hand assembled in California using only premium grade analog components.
“As a company, we are proud to embrace vintage gear philosophies while constantly evolving and refining our product line,” said Loughman.
“For us, it is very important to take the very best of our legacy products and build on that, which in turn enables our customers to be more creative and realize their sonic aspirations.”
Pricing is set at $1,900 for a single unit without power supply, $2,100 with power supply, and $4,000 for a pair with power supply.
Posted by House Editor on 12/10 at 09:48 AM
Red Bull Studios In Amsterdam Chooses Audient
Studio upgrades from high-maintenance vintage console to the large format ASP8024 for variety of creative clients.
36 channels of Audient ASP8024 console are at the heart of Red Bull Studios in Amsterdam, a place where studio manager Jasper Cremers reckons, “Anything is possible.”
This mantra extends across the worldwide Red Bull Studios Network, as Jasper explains: “The idea behind Red Bull Studios is creating a place for musicians and artists where they can create whatever they want.”
One of the biggest draws, is the free use of the studios - a rare phenomenon. “The artists that come here are free from any obligations - and that’s when the creativity starts flowing. Our motto is ‘giving wings to artists’ and with our studios we aim to do just that.”
The recording studios are booked up to the end of the year and beyond, so these unique opportunities are certainly being utilized, according to Cremers - by an assortment of artists.
“The best thing about the Red Bull Studios in Amsterdam is that we really encounter a diverse array of musical genres. Amsterdam is known for the Amsterdam Dance Event so there’s a very strong electronic music scene in this country, but there’s also a huge band culture here. And let’s not forget the big hip hop scene in all its forms. The list of artists goes from Mayer Hawthorne and Dennis Ferrer to Diplo with Ghostface Killah in between.”
It was this variety of musicians coming through the doors of the Amsterdam studios that led Cremers to choose Audient.
“Due to the diversity in genres we have in our studio, we wanted a neutral sounding desk that could go in any direction.” The large format ASP8024 console, complete with with Dual Layer Control (DLC) and integrated patchbay is actually the second console to grace the Amsterdam complex. Cremers admits, “The first console was a vintage one. Looked great but the maintenance was high on that baby.
“The best thing about the Audient is reliability. You don’t want to get in a situations where you need to stop recording because the console isn’t working properly. That’s a killer for every studio and engineer. Our Audient took away that concern.”
Red Bull Studios Amsterdam also boasts a live room big enough to host showcases and intimate one-off gigs, and there are regular workshops and lectures on offer to help inspire young artists, welcoming anybody who makes music, including budding producers. “They have the opportunity to enhance themselves,” agrees Cremers. “All we can do is offer them the tools to grow, the rest is up to them.
“Come prepared and use your imagination.” he says. Pretty good advice all round, we reckon.
Posted by House Editor on 12/10 at 09:36 AM
Properly Matching Microphones & Preamps
Selecting the right preamplifier for a given microphone, or conversely, selecting the right mic for a given preamp, involves two major factors along with several minor ones. First, the two big ones:
1) Input headroom. Do you have enough?
2) Noise. What will the preamp add to your mic?
You need to determine whether the mic, under worst-case conditions, is going to overload the preamp input stage, and also whether the preamp is going to materially degrade the noise performance of the mic.
Actually, mics have relatively few specifications. Most are sold on sound, reputation and price. Specifications rarely enter into it. Even so, enough exist to make the right decision.
Another issue is proper input impedance. The trend has been moving toward higher input impedances, with many now rated at 2 kilohms and higher. Because the connected impedance (i.e., mic plugged into the preamp) determines the noise performance, and the mics are low impedance (150-200 ohms), then there is no noise penalty for providing higher input impedances.
Yet another aspect to examine is phantom power. Is it provided? Do you need it? Is it the correct voltage, and does it source enough current for your mic? This is an area requiring informed decisions. There’s a huge myth that mics sound better running from 48 volts, as opposed to, say, 12 volts, or that higher phantom power increases the dynamic range of a mic. For the overwhelming majority of microphones both of these beliefs are false.
Most condenser microphones require phantom power in the range of 12-48 VDC, with many extending the range to 9-52 VDC, leaving only a very few that actually require just 48 VDC. The reason? Internally, most designs use some form of current source to drive a low voltage zener (usually five volts; sometimes higher) that determines the polarization voltage and powers the electronics.
The significance is that neither runs off the raw phantom power; both are powered from a fixed and regulated low voltage source inside the mic. Increasing the phantom power voltage is never seen by the microphone element or electronics; it only increases the voltage across the current source. But there are exceptions, so check with the manufacturer, and don’t make assumptions based on hearsay.
Final selection details involve checking that the preamp’s gain range is enough for your use, that there are overload indicators or metering to help in set up, that the plumbing is compatible with your wiring needs, and that the color doesn’t clash with your tour jacket.
Finding The Data
Determining input headroom compatibility requires knowing the mic sensitivity rating and the maximum sound pressure level (SPL) allowed. The sensitivity rating is usually the easiest and least ambiguous number to find on the data sheet, rated at 1 kHz and expressed in millivolts per pascal (mV/Pa). One pascal is the amount of pressure resulting from a loudness level of 94 dB (written as 94 dB SPL). For example, a sensitivity rating of 20 mV/Pa tells you that when a sound equal to 94 dB SPL strikes the mic element, resulting output voltage is 20 millivolts.
The sensitivity rating provides a voltage level at one reference point; now all that’s needed is the mic’s maximum SPL to calculate the maximum output voltage. Then this is compared against the maximum input voltage rating of the mic preamp. The maximum allowed SPL is stated in several ways: maximum SPL often with a stated total harmonic distortion (THD) level; maximum acoustic input; and sound pressure level for “X” percent THD. All are variations for the same rating.
Table 1: Mic maximum output level (dBu).
With these two specifications, it’s a simple matter to calculate the maximum output level in volts and convert that into the familiar dBu units found on microphone preamp data sheets. To make it even easier, see Table 1. To obtain the microphone maximum output level in dBu, find the mic’s sensitivity rating on the left side and then move right until directly below the mic’s maximum SPL rating.
As an example, for a microphone with a sensitivity rating of 20 mV/Pa and a max SPL equal to 130 dB, Table 1 tells us that the maximum output voltage is +4 dBu. You now have what you need to compare preamps regarding maximum input level.
Another example using Table 1: block out all possibilities that could overload a specific preamp. For example, the red triangle area represents all combinations that could overload Rane’s MS 1b Mic Stage. Its maximum input level is rated at +10 dBu, therefore all microphone sensitivity and maximum SPL combinations resulting in greater than +10 dBu are excluded from consideration. Used this way, any new mics can be quickly checked for overload threat.
Caveats. Remember that this output level only occurs under the worst-case condition of SPL equaling the maximum allowed by the mic, meaning that if an application has sources that cannot achieve the maximum SPL, then the input overload requirement can be relaxed accordingly. For instance, if you know your source is never going to exceed 110 dB SPL, and your mic is rated for maximum levels of 130 dB, then you can take 20 dB off the levels shown in Table 1, widening your preamp choices considerably.
Note also that input overloading is a strong function of the preamp’s gain control setting. Most preamp manufacturers measure the maximum input level with the gain control set at minimum. This means there is a real danger that even after carefully matching the output and input levels of a mic and preamp, the mic can still overload the preamp.
This happens when the system needs the preamp gain turned up (correspondingly reducing input headroom) and the microphone is used for a wide dynamic range source. Not only do you have to worry about matching your mic and preamp, but also about real-world sources and gain settings.
Individual Noise Floors
Microphones and preamps each have their own noise floors. When selecting a preamp, it’s important to know to what degree the preamp’s noise degrades the noise of the mic.
Different mic technologies use different terminology to describe noise. Dynamic microphone data sheets rarely list noise as a specification because there is no active circuitry to generate noise; there is only a magnet and a coil. This category of mic is properly called electromagnetic or electrodynamic.
Output noise is very low—so low it’s not listed. However, some noise is generated, and this can be calculated by knowing the mic’s impedance.
Obtain the dynamic microphone impedance rating from the data sheet and use Table 2 to convert that into units of dBu, A-weighted. This noise is the white noise generated by the resistance of the wire used to create the coil, plus a correction factor of 5 dB for A weighting. (This is somewhat arbitrary, as true A weighting may decrease the level anywhere from 3-6 dB depending upon the nature of the noise, but agrees with Holman’s article—noted later—and measured results.)
Table 2: Output noise for dynamic mics (20 Hz - 20 kHz, 20 degrees C/68 degrees F).
The noise of the measuring standard 150 ohms (200 ohms for Europe) source resistor makes a good noise reference point. In Table 2, it equates to -136 dBu (A-weighted; -131 dBu when not). This means that you cannot have an operating mic stage, with a 150 ohm source, quieter than -136 dBu (A-weighted, 20 degrees C/68 degrees F, 20 kHz BW). Looking at Table 2 confirms that dynamic microphones, indeed, are quiet.
Use Table 3 to compare microphone output noise with preamplifier equivalent input noise (EIN). As an example, if your dynamic microphone’s output noise equals -136 dBu, and you are considering a preamplifier with a rated EIN of -136 dBu, then the difference between them is 0 dB.
Table 3 illustrates that this preamp with this microphone will degrade the total noise by 3 dB. That is, the combination of mic and preamp adds 3 dB noise to the total. More on how this table works shortly.
Condenser, capacitor, or more properly, electrostatic microphone technology involves a polarizing voltage network and at least a buffer transistor built into the microphone housing, if not an entire preamp/biasing/transformer network—all of which contribute noise to the output. Electrostatic microphones are quite noisy compared to dynamic designs, but are very popular for other reasons.
Table 3: Output noise for condenser mics (dBu).
Different manufacturers use different terminology on their electrostatic microphone specification sheets for noise: Self-Noise, Equivalent Noise SPL, Equivalent Noise Level, Noise Floor, and just plain Noise all describe the same specification. Microphone noise is referenced to the equivalent sound pressure level that would cause the same amount of output noise voltage and is normally A-weighted.
This means the noise is given in units of dB SPL. A noise spec might read 14 dB SPL equivalent, A-weighted, or shortened to just 14 dB-A (bad terminology, but common). This is interpreted to mean that the inherent noise floor is equivalent to a sound source with a sound pressure level of 14 dB.
Problems arise trying to compare the mic’s noise rating of 14 dB SPL with a preamp’s equivalent input noise (EIN) rating of, say, -128 dBu. Talk about apples and oranges!
Luckily (again), tables come to the rescue. Table 4 provides an easy look-up conversion between a microphone’s output noise, expressed in equivalent dB SPL, and its sensitivity rating, in mV/Pa, into output noise expressed in dBu, A-weighted.
Using Table 4, a direct noise comparison between any microphone and any preamp is possible. The example shown by the blue column and row is for a mic with a noise floor of 14 dB SPL and a sensitivity rating of 20 mV/Pa, which translates into an output noise of -112 dBu, A-weighted.
Table 4: RMS noise summation for connected mic and preamp.
Now, time to return to Table 3. Unfiltered electronic noise, whether from a resistor, a coil, an IC, or a transistor is white noise consisting of all audible frequencies occurring randomly. Due to this randomness you don’t just add noise sources together, you must add them in an RMS (root mean square) fashion. Mathematically this means you must take the square root of the sum of the squares—which is why Table 3 is so handy—it does the RMS conversion for you.
Use Table 3 to convert a mic’s rated noise output into units of dBu. Find the difference in dB between the mic’s output noise and the preamp’s input noise. Find that difference in the left column of Table 3 and read what the preamp added noise will do to the mic’s noise in the right column.
For example, if the mic’s output noise translates into -120 dBu, and the preamp has an EIN of -127 dBu, then the difference between the mic and the preamp is -7 dB. That is, the preamp is 7 dB quieter than the microphone. Table 3, at the row marked -7 dB, tells you that this preamp will degrade the mic’s noise by only 0.8 dB. Looking at Table 4 tells us that after about a 10 dB difference, the noise added by the preamp becomes insignificant.
Similar to Table 1, you can use Table 4 to map out a preamp’s A-weighted noise to show the combinations that add insignificant noise. If you use a -10 dB difference figure as a guide, then the preamp’s noise amounts to less than 0.4 dB increase.
The red-shaded triangle area in Table 4 shows an example of this. The areas not shaded represent all possible combinations of mic sensitivity and noise specifications that can be used with Rane’s MS 1b Mic Stage, for instance, and add less than 0.4 dB of noise.
If you allow 1 dB net added noise, then even more combinations are possible. (The shaded area is figured by taking the EIN of the MS 1b at -128 dBu, reducing it to -133 dBu with the 5 dB factor for A weighting, and using the -10 dB difference found in Table 3 for 0.4 dB added noise, resulting in all combinations less than -123 dB being blocked out.)
The author would like to point out that this note was inspired by an article authored by Tomlinson Holman, published in September 2000 Surround Sound Professional magazine, titled “Capturing the Sound, Part 1: Dynamic Range.”
Dennis Bohn is a principal partner and vice president of research & development at Rane Corporation. He holds BSEE and MSEE degrees from the University of California at Berkeley. Prior to Rane, he worked as engineering manager for Phase Linear Corporation and as audio application engineer at National Semiconductor Corporation. Bohn is a Fellow of the AES, holds two U.S. patents, is listed in Who’s Who In America and authored the entry on “Equalizers” for the McGraw-Hill Encyclopedia of Science & Technology, 7th edition.
Tony Visconti Upgrades To Barefoot Sound
Grammy-winning producer/engineer, musician and singer featured in newest episode in the "Masters of the Craft" video series
Grammy-winning producer/engineer, musician and singer Tony Visconti has outfitted his studio in New York City with Barefoot Sound MM27 monitors.
“I’ve been in this studio for five years,” he explained. “I’ve added a little more equipment, including these wonderful speakers behind me but nothing much has changed. I’d like to keep things status quo for awhile.”
Since the late 1960s, Visconti has worked with a wide array of recording artists; his lengthiest involvement with any artist is with David Bowie: intermittently from Bowie’s 1969 album “Space Oddity” to 2013’s “The Next Day,” Visconti has produced and occasionally performed on many of Bowie’s albums.
“The first time I heard a pair of Barefoots was during the Sonic Solutions sessions at the Magic Shop in New York City when Butch Vig brought his own pair,” Visconti recalls.
“They were on the board and at first I thought it was the in-wall monitor speakers but it was the Barefoots sitting on top of the console that I was listening to. Then I finally got it. These are great speakers and I’ve got to have them.”
Other notable artists Visconti has worked with include T-Rex, Badfinger, Iggy Pop, Argent, Thin Lizzy, Boomtown Rats, Moody Blues, Sparks, Semi Precious Weapons, Kaiser Chiefs, and Morrissey.
“I know some monitors are very colorful; they’ll give you extended low end, and a crunchy mid-range. I’ve been deceived too many times by getting a souped-up pair of monitors in my studio. I’ve tried everything and spent years and years of research to find something that I’m really happy with.”
See the entire “Masters of the Craft” Tony Visconti interview here.
Wednesday, December 09, 2015
Convention Committee Announced for AES 140th International Convention In Paris
Will offer four full days of in-depth programs and presentations, facility tours, and a three-day manufacturer exhibition
The Audio Engineering Society (AES) has announced the official convention committee for the 140th International AES Convention, set to take place June 4-7, 2016, at the Palais des Congrès de Paris in Paris, France.
Co-chaired by Michael Williams and Umberto Zanghieri, the 140th convention will offer four full days of in-depth programs and presentations, facility tours, and a three-day manufacturer exhibition.
Charged with leading the paper sessions at the convention are paper co-chairs Thomas Gorne (Germany), Wolfgang Klippel (Germany), Bergane Periaux (France), Robin Reumers (Belgium), and Dejan Todovoric (Serbia). Co-chairs for the convention’s workshops presentations will be Natanya Ford (UK) and Rob Toulson (UK).
The technical tours will be chaired by Phillippe Labroue (France), while additional support will be provided by facilities co-chairs Layan Thornton (France) and Nadjia Wallaszkovits (Austria).
“Our convention chairs Umberto Zanghieri and Michael Williams have put together an impressive team to build the program for the 140th AES Convention,” says Bob Moses, AES executive director. “The 140th is going to be a great event in a great city. If you are serious about audio, you seriously need to join us in Paris.”
Audio Engineering Society (AES)
31st Annual NAMM TEC Awards Honoring Don Was With Les Paul Award
TECnology Hall of Fame ceremony at Winter NAMM in Anaheim to induct 10 iconic musical products
The NAMM Foundation is presenting the 31st annual NAMM Technical Excellence & Creativity (TEC) Awards and the NAMM TECnology Hall of Fame on January 23 at the 2016 NAMM Show in Anaheim. The events honor the best in audio and sound production as well as the most impactful audio technology products from the last 75 years.
Record producer and music industry executive Don Was will receive the 2016 Les Paul Award, an annual recognition of those who embody the creative spirit and legacy of a revered musical genius. Saturday night’s NAMM TEC Awards will be hosted by comedian Sinbad.
Was co-founded former Detroit-based band Was (Not Was) with childhood friend David Was (Weiss) before going on to produce decades of commercially successful and critically-acclaimed recordings for top artists.
He has earned multiple Grammy Awards for his production work over the past three decadesm including Bonnie Raitt’s Nick Of Time (’89), Producer Of The Year for work with artists ranging from The Rolling Stones to Willie Nelson and Roy Orbison (‘94), and Ziggy Marley’s Best Musical Album For Children, Family Time (‘09).
As president of Blue Note Records, Was oversees the label’s extensive reissue campaigns that serve audiences in both the analog and digital realms. He joins a group of Les Paul honorees including Stevie Wonder, Sting, Pete Townshend and more, who have “set the highest standards of excellence in the creative application of audio and music technology,” according to the Les Paul Foundation.
The January 23 event will be held in the Hilton Anaheim Pacific Ballroom. A reception begins at 6 pm, with the ceremony following at 7 pum. Tickets are available here.
Earlier on that day, the NAMM TECnology Hall of Fame will induct 10 audio products and innovations released between 1928 and 2002 that have made a significant contribution to the advancement of audio technology. A panel of more than 50 recognized audio experts, including authors, educators, engineers, facility owners and other professionals selected the nominees.
The Neumann KM84 microphone was invented by Georg Neumann five decades ago and is still in use today via the improved KM184 model. It stands out as the first microphone to use the now-standard “phantom powering system.”
Also entering a 50-year milestone is the Shure SM58 microphone, a standard in the eyes of many. After three years and hundreds of tests involving dropping, throwing, cooking, salt spray immersion and submersion, the SM58 was born under the watchful eye of Ernie Seeler, a classical music fan (who hated rock ‘n’ roll).
Turn up Supertramp’s “Logical Song” and Steely Dan’s “Do It Again” and you’ll hear the distinct sounds produced by Wurlitzer’s Electronic Piano, originally designed as a portable and substitute for the acoustical piano. It has become a mainstay of rock, pop and jazz artists worldwide.
The Roland RE-201 Space Echo was first released when Richard Nixon was still U.S. president, but its vintage sound continues to find its way into the recordings of musicians that include Fatboy Slim, Mr. Oizo, Sneaker Pimps, Radiohead, Lauryn Hill and more.
Other inductees include the Manley VoxBox and Avid Digidesign Pro Tools HD, which are both still in production today.
The TECnology Hall of Fame ceremony will be presented Saturday, January 23, from 4 pm to 6 pm in room 202A of the Anaheim Convention Center in the TEC Tracks educational area. Seating is limited and available to credentialed NAMM attendees, inductees and their guests. Find out more about the TECnology Hall of Fame honorees here.
NAMM Technical Excellence & Creativity (TEC) Awards
NAMM TECnology Hall of Fame
CAD Audio Celebrating 85th Anniversary
Originally founded in 1931, as the Astatic Corporation, CAD will be celebrating with new product and events to mark this special year.
In 2016, CAD Audio will be celebrating its 85th Anniversary with a number of new product introductions including the CADLIve Wireless Systems and D88 Kick Drum Mic, the “Classic” A77 Dynamic Mic and the U37SE Special Edition USB Mic, along with a series of events to mark this special year.
CAD Audio originally took shape as the Astatic Corporation, founded in 1931 by C.M. Chorpening and F.H. Woodworth, two ham radio operators with a passion for clean, clear audio transmissions.
This passion led them to the development of the D-104 mic, revolutionary in its ability to perform without static.
This “Astatic” characteristic became the company’s namesake and led to many innovative products including phonograph pickups and cartridges, while also laying foundations for the production of recording heads.
Astatic Commercial Audio Products has continued its growth since then with industry leading solutions for commercial AV applications such as airports, schools, courthouses, Houses of Worship and corporate boardrooms.
The company remains key to CAD Audio’s portfolio for today’s integrators, consultants and architects with patented innovations such as Variable Pattern microphones.
CAD (Conneaut Audio Devices) Professional Microphones was initially formed as the Pro Division of the company to serve the emerging recording and live sound markets in the early ‘80s, segments not served by Astatic. Initially, CAD introduced a servo circuit console and processing products.
The company went on to develop studio microphones such as the Equitek Series E100, E200, E300, VSM1 and VX2. Capitalizing on innovative engineering and home grown capsules/diaphragms, CAD emerged as a leader in recording microphones and the project studio revolution of the late 1980s and early 1990s.
More recently, CAD Audio has emerged as a multifaceted company with a variety of the sub brands that extend to market segments such as Recording, Broadcast, Live Performance & Production, Commercial and Consumer Electronics. These sub brands include the CADLive, Acousti-Shield, StageSelect, USB and Sessions series of products. CAD Audio is also actively engaged in the Wireless Microphone, Wireless In-Ear-Monitor System, Studio accessory, Headphone and USB Microphone markets.
CAD Audio continues to build on over 85 years of innovation and success in the audio industry by delivering innovative and highly useful solutions for performers, engineers and a wide range of audio professionals.
Tuesday, December 08, 2015
dBs Music Berlin Utilizes Antelope Audio As Teaching Tool
International music school outfits production suites for recording with Zen Studio interfaces.
For 16 years dBs Music has provided students interested in music production with coursework at locations throughout the United Kingdom. Expanding its program to Germany two years ago, the teaching organization set up its studios at Funkhaus radio studio in former East Berlin, building out the acoustically treated spaces into two tracking rooms and four production suites.
With a generously sized space of 30 square meters, the production suites themselves were large enough for tracking, and dBs Music sought to find the right interface for each of them. dBs Music operations manager, Zak Davies, found the solution in the Antelope Audio Zen Studio.
Davies first encountered the Zen Studio at the Antelope Audio booth at Musikmesse.
“It just sounded great, and I was really impressed,” he recalls. So when it came time to outfit dBs Music’s production suites for recording, the Zen Studio immediately came to mind. “The production suites are big enough to record in and they’re really nice acoustically, so we built them out as both mixing suites and mini recording studios,” Davies explains.
“We wanted to install really nice interfaces and the Zen Studio fit the bill.”
Among the compelling features of the Zen Studio were the 12 high-quality mic preamps, easy-to-use software, and USB connectivity. “Having 12 built-in preamps really sets it apart from other devices we’ve had before and makes it much more comprehensive for us in terms of equipping a studio,” Davies notes.
With both its diploma and degree level courses, dBs Music Berlin endeavors to impart strong technical foundations while fostering the creative aspects of the recording process.
“We try to go deeper than just tech knowledge,” says Davies. “It’s more about the student as an artist and how they develop over time and find their niche.” The Zen Studio has served as an ideal companion for artistic discovery. Students of all levels have found it “easy to get along with,” Davies adds. “It’s very intuitive; it gets out of the way and the students are able to just get on with the creative part of their work, which is exactly what we want them to do.”
The Zen Studio’s operation also helps maintain a clear path for instruction on other aspects of the recording process.
“All of our instructors are working in the industry in some capacity, so they know how high-quality gear performs,” Davies says, and the Zen Studio delivers the same reliability the instructors would expect in their work outside the classroom. “It keeps the lesson flowing because the software is very well designed. We are able to set it up and carry on with whatever we’re trying to do, rather than fussing around in software for hours.”
Both staff and students are also satisfied with the sound of the Zen Studio, thanks in large part to Antelope Audio’s trademark AD/DA conversion. “We had read tons and tons of great reviews about the clocking and conversion and it has definitely lived up to expectations”.
Pleased with what the Zen Studio has done for dBs Music, Davies and his team are starting to take advantage of its portability to find uses for it beyond the classroom as well.
“One of our tutors actually took a Zen Studio to a castle for a weekend to record parts for his album and was just raving about it when he got back,” says Davies. As for dBs Music’s students, they graduate with what their curriculum refers to as “skills for success.” For Davies, that means leaving their program with “their own distinct creative approach to what they’re doing and how to apply it to the real world.” But they also garner a healthy appreciation for gear like the Zen Studio that allows that nascent creativity to flourish.
“It offers so much more than any other interface we’ve used before and makes it feel so easy.”
What Is Workflow And Why Is It Important In The Studio?
I wanted to write a brief post addressing workflow as it relates to anything in music production, but specifically in studio recording.
Too many people get caught up in gear, the science of behind sound, discovering quick recording tricks, and more, that they forget about the fundamentals.
What is workflow?
Broadly defined, workflow is: The sequence of industrial, administrative, or other processes through which a piece of work passes from initiation to completion.
To put it another way, workflow is the pattern you follow when working
How does workflow apply to the studio?
If you’re striving for excellence in the studio, then you should be cognizant of what your workflow is, and how you can improve it. The better you understand HOW you work, the easier it will be to make positive changes to your methods and get MORE EFFICIENT when you’re doing anything in the studio.
Tips for workflow improvement
Think about the last time you tried to record a track, and ask yourself questions such as:
—What was the first thing I recorded?
—Did I run into any issues that took a long time to figure out?
—How many retakes did I need to get it right?
—Was I prepared for my recording session beforehand?
—Did I spend too much time on mixing or mastering?
Answering these (and there are many other great questions you should consider) can help you get a sense of how you’re working currently. From there, you can start focusing on your weak spots and come up with ways to improve upon them. You can also decide to change your recording order up.
There’s no “optimal” order of doing things when it comes to recording any one song, but you may find that one way seems to give you better results (in both quality and time).
Once you figure this out, the key is repetition. Discover a workflow that works best for you then stick to it, and be a stickler about it. A great quarterback spends hours practicing the same throwing motion to perfect his passing abilities, that’s how he gets better. Treat your workflow in the studio like that, and you’ll see improvements you never thought possible!
Read the original article and contact Scott here.
Scott Hawksworth is the founder of RecordingExcellence.com and is passionate about helping musicians learn more about home recording and mixing. He has been playing piano for over 15 years, and has been a member of various church choirs, glee clubs, and local bands. He lives in Chicago with his wife and two cats
Posted by House Editor on 12/08 at 12:22 PM
Grammy-Winner Rafa Sardina Relies On Manley
Los Angeles-based producer/engineer leans heavily the Massive Passive stereo, four-band EQ and Variable Mu stereo limiter-compressor.
With 12 Grammy Awards and more than 40 American and Latin Grammy nominations to his credit, Rafa Sardina leans heavily on processors from Manley Labs.
The Los Angeles-based producer/engineer’s client list includes such luminaries as Stevie Wonder, Elvis Costello & The Roots, Lady Gaga, Michael Jackson, Placido Domingo, Celine Dion, Alejandro Sanz, Harry Connick, Jr., Luis Mighuel, D’Angelo, and on and on- the list seems endless.
When you add his earlier engineering credits at Ocean Way and Record One Studios, the list is downright stunning. Though best known for working on pop, rock, and R&B, Sardina has recorded and mixed in almost every musical genre and every medium, including TV and major films.
You don’t work at Sardina’s level unless you have great ears, high standards, and uncompromising taste in audio equipment. Sonic quality is the biggest reason he loves Manley signal processors.
“Some gear has a very specific sound but not always in a good way. It might have a use, of course, but you can’t put it on everything,” he explains.
“But all my Manley processors sound great, regardless of how you use them, and they never compromise the frequencies. I use them on everything.” Reliability is another plus for Sardina. “I’ve never had issues with Manley products; they’re extremely well built,” he declares.
A fan of classic EQs, Sardina’s modern favorite is the Manley Massive Passive stereo, four-band EQ, which he describes as “like a Pultec on steroids.”
“I use the Massive Passive for both tracking and mixing, and for every musical genre,” relates Sardina. “The Massive Passive is extremely versatile and allows me to be very surgical. I have a few of them, and I love to use them on stuff where I have to dig into specific frequencies to make something out of almost nothing because the original sound was not even close. And when recording orchestra, I always have a Massive Passive across my stereo bus.”
The Massive Passive is a favorite of many of Sardina’s colleagues as well. “I know a lot of engineers who use it for final mixing and scoring, and they have three Massive Passives so they can apply the same type of EQ to the full 5.1 mix,” he notes.
Sardina also makes extensive use of the Manley Variable Mu stereo limiter-compressor. “I love to use the Variable Mu for tracking bass,” he relates. “I also use it for mixing. And I always rely on it for background vocals. When we have very aggressive background vocals, I use the Variable Mu to bring the background vocals forward, and with the processing, they’re not as aggressive anymore so they sit well in the mix.”
Having used a lot of Manley gear over the years and loved it all, Sardina is eager to see what’s next.
“I loved the older ELOP stereo electro-optical limiter,” he says. “I used it more as a limiter than as a compressor, especially for vocals. If Manley were to make a new product like that, I’d be all over it. I use their Gold microphones, too, and would love to try a new one.”
Posted by House Editor on 12/08 at 12:11 PM
Al Jazeera Balkans Expands And Upgrades With Calrec Audio
Company adds router core, two Artemis audio consoles, and a Summa console to existing Hydra2 network.
As part of an expansion and upgrade project, Bosnia’s Al Jazeera Balkans has installed a Calrec router core, two Artemis audio consoles, and a Summa console. The router core provides the hub for a centralized audio network that allows resources to be shared among all of its studios.
“Al Jazeera Balkans started in November 2011 with one studio and one stand-alone digital audio console. When we decided to expand our facility, we chose Calrec because they offer very powerful, rock-solid consoles built with broadcasters in mind. That’s why Calrec is at the heart of our audio infrastructure,” said Mirad Isakovic, manager of the broadcast technology department at Al Jazeera Balkans.
“Our operators like Calrec consoles because they are user-friendly, the surface is clearly organised and arranged in a way that makes sense, and everything is readily available. Calrec simply makes the operators’ lives easier. Now we have two television studios and we can share cameras, microphones, and other sources from one studio to another.”
Al Jazeera Balkans has installed identical Artemis Light consoles in its TV studios and a Summa console — the first Summa in the Al Jazeera organization — in the new radio studio. The consoles, along with eight I/O boxes placed throughout the facility, connect to the router core via Calrec’s Hydra2 networking protocol. By using the Hydra2 network, the consoles can share signals and resources as needed.
The Artemis consoles also integrate with the broadcaster’s Mosart automation system. Operators can control functions such as patching, routing, equalization, and faders if needed, while Mosart controls only the faders. This ability to automate fader control speeds up operations and makes workflows more efficient.
“Calrec provided excellent support. We had to replace our existing console with the Artemis overnight so that it would be ready for live news production the next day. That meant our installation timeline was extremely tight,” Isakovic said.
“We had an excellent work experience with the Calrec engineers and support team. They helped us ready the Hydra2 network and router core and configure the Artemis ahead of time, so the changeover went smoothly.”
Currently Al Jazeera Balkans is only using one of the Artemis consoles (in the main TV studio) for live news. The broadcaster is conducting dry runs with the second Artemis and the Summa and expects them to go live in February 2016.
“Al Jazeera Balkans needed a fully integrated audio solution that ensures all resources and signals will be available on any console at any time, which is precisely what our Hydra2 network is designed to do,” said Mike Reddick, international sales manager for Calrec.
“Now, instead of having one stand-alone console with limited I/O, Al Jazeera Balkans has multiple, larger consoles with more faders and layers; more PGM, AUX, group, and track buses; more functionality; more flexibility; and more efficiency — all linked on a Hydra2 network. That complete audio-networking capability gives Al Jazeera Balkans the redundancy that’s critical for any live broadcast environment. We believe this benefit is one reason the Al Jazeera organization has chosen to standardise on Calrec whenever possible.”
Sonnox Releases New ENVOLUTION Plugin
Allows complete manipulation of the audio’s envelope to produce creative and even extreme results.
Sonnox announces the release a brand-new product, Oxford Envolution, a frequency-dependent envelope shaping plug-in to radically modify the sound of individual tracks, buses and even master outputs.
Envolution boasts comprehensive and independent control of transients and sustain, with ‘Tilt’ or ‘Focus’ controls to choose where in the frequency spectrum the effect is applied.
This allows complete manipulation of the audio’s envelope to produce creative and even extreme results.
Transients can be boosted to add presence and punch, or reduced to create perspective; while the sustain section can easily make the ambience around a recording bloom, or can be cut back for quick and precise gating.
Senior sales & marketing manager, Nathan Eames, comments, “With such a wide range of potential users in so many audio sectors, from post, music to live sound, we’re excited about the creative possibilities that Envolution will bring. In keeping with all Sonnox designs, we’ve kept three core values in mind throughout the development process of Envolution – ease of use, an intuitive interface and most importantly – a great sound.”
Envolution is available in Native format – AAX Native, Audio Units and VST - at a retail price of $270 (USD) and AAX DSP format at a retail of $435 (USD)
Synergy SoundwerX Upgrades Studio With Allen & Heath
QU -32 digital console with built-in DSP replaces a 4-foot rack of analog signal processing and simplifies studio setup.
Synergy SoundwerX is a small recording studio that provides a variety of services from multi-track recording and mix-down to commercial voice-overs and professional photography.
Recently, the studio upgraded to an Allen & Heath digital console from American Music & Sound to prepare for growth.
When its 1990’s-era analog mixer became unreliable and studio owner Shawn Helton began to search for a replacement. He wanted a digital mixer with “scenes” to store setups for repeat customers and a digital patch bay to simplify studio wiring.
He wanted versatile DSP for the many different instruments and voices he records. He wanted multi-track recording and mix-down capabilities with simple connections to his iMac computer. And, he wanted the kind of audible quality normally associated with an analog mixer.
After online research and discussions with other engineers, Helton chose Allen & Heath’s new QU- 32 digital mixer.
“Even though I seldom use more than 18 microphones, I wanted the 32-channel model for future expansion,” he said, “and the QU -32 has features like its ‘Qu-Drive’ direct multi-track recording and 32x32 USB audio interface that gives me a one-cable connection to the recording software on my iMac.”
The QU -32’s built-in DSP replaced a 4-foot rack of analog signal processing. “It really cleaned up my studio,” Helton said. In the future, he wants to add an Allen & Heath dSNAKE to further simplify the studio’s wiring.
Helton, who is also a musician and front of house engineer, has a critical ear for music and sound and his studio enjoys a reputation for excellent acoustics and recorded sound quality. He says the QU- 32’s audible sound quality was a key factor in his purchase decision.
Listening to recent recordings over his Rokit 8 control-room monitors, he commented, “I’ve never heard it sound so good. The QU-32 makes me want to go back and remix everything I’ve done in the last ten years.” He added, “The QU- 32 has everything I need for my recordings and voice-overs and I feel like I’ve entered the true digital age.”
Allen & Heath
American Music & Sound