Thursday, September 04, 2014

Hal Leonard And Groove3 Announce Strategic Partnership

Collaboration will transform Hal Leonard's content using Groove3's online video delivery system and subscription model

Hal Leonard Books, a division of Hal Leonard Corporation, a leading publisher of books and digital content on the music business, audio technology, and instrument history, and Groove3, the audio community’s source for informative and effective online tutorials, announced today a long-term strategic partnership to develop and deliver authoritative content.

This collaboration will transform Hal Leonard’s content, including series such as Music Pro Guides and Quick Pro Guides, using Groove3’s online video delivery system and subscription model, while expanding Groove3’s reach beyond the robust community the company has built over the last 10 years, addressing all aspects of the music-making process, including recording, production, engineering, mixing, songwriting, DAW guides and more.

“Groove3 has always had the end user’s best interest in mind and is dedicated to delivering the best tutorials about today’s audio tools and recording and production techniques. Now having the opportunity to partner with Hal Leonard and offer their first-class content along side ours, it’s a match made in heaven for all audio professionals and hobbyists alike around the world,” said Groove3 vice president Antony Livoti.

“Groove3 is the most trusted and time-tested online quality resource for training videos for musicians, and bringing Hal Leonard’s reputable brands and content into their community will be a huge benefit to all musicians interested in learning online,” said John Cerullo, group publisher of Hal Leonard Books.

“Hal Leonard has been a pioneer in offering digital content to active music makers for decades, and the Groove3 partnership, along with our many other recent digital and web based initiatives, will allow us to continue to offer the best in music instruction for years to come,” adds Hal Leonard Corporation president Larry Morton.

Groove3 currently offers more the 850 hours of top-notch online training. The new, exclusive content from Hal Leonard will include product by renowned recording, audio, and music experts from many fields, including the Hal Leonard Recording Method by Bill Gibson, the Bruce Swedien Recording Method, Rikki Rooksby’s series of books on songwriting, and much more.  Additional courses and products will be announced and released in the coming weeks.

In addition, the partnership allows for the development and offering of customized online programs for traditional resellers, such as musical instrument dealers, and licensing programs to audio-trade outlets, secondary and higher educational institutions, and industry organizations.

Hal Leonard Books

Posted by Keith Clark on 09/04 at 01:51 PM
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MOTU Announces Three New Thunderbolt Audio Interfaces

New line offers complementary I/O configurations, A/D/A conversion, 48-channel mixing, DSP effects and AVB Ethernet audio networkin

MOTU has announced three new Thunderbolt audio interfaces with complementary I/O configurations, A/D/A conversion with high dynamic range, 48-channel mixing, DSP effects and AVB Ethernet audio networking for system expansion.

“The culmination of years of R&D, these exciting new products from MOTU usher in a new era in audio interface technology,” states Jim Cooper, director of marketing at MOTU. “The new 1248, 8M and 16A audio interfaces offer a breakthrough combination of superb audio quality, large-scale mixing, powerful DSP and system expansion through the IEEE’s new AVB Ethernet audio networking standard.”

Based on a new, shared technology platform, the 1248, 8M and 16A differ only in their analog I/O configurations. As the most versatile of the three, the flagship 1248 offers 8 x 12 balanced TRS analog I/O, four mic inputs with digitally-controlled individual preamps, two front-panel hi-Z guitar inputs, two independent phone outs and stereo RCA S/PDIF digital I/O.

Meanwhile, the 8M provides eight balanced TRS analog outputs, plus eight mic/line/instrument “combo” style inputs individually equipped with digitally controlled preamps, 48-volt phantom power, pad and MOTU’s hardware-based V-Limit overload protection. The 16A offers 32 balanced TRS analog connections (16 inputs and 16 outputs).

All three units provide two banks of optical digital I/O, word clock I/O and computer connectivity through either audio class compliant USB 2.0 or Thunderbolt (1 and 2 compatible).

Equipped with the latest-generation ESS Sabre32 Ultra converters, analog performance has been engineered for demanding listening situations. For example, the balanced TRS analog outputs provide a measured dynamic range of 123 dB (A-weighted, 20 Hz to 20 kHz). Analog I/O latency is very low, with round trip performance of only 32 samples (0.66 ms) at 48 kHz. Each unit provides comprehensive metering for all inputs and outputs on a large, backlit 324 x 24 pixel LCD.

The 1248, 8M and 16A are equipped with latest-generation DSPs that drive a 48-channel mixer designed after large-format mixing consoles. With 32-bit floating point precision, the mixer’s 48 inputs can take signal from the physical inputs on the interface itself, audio channels from host software on the computer, audio network streams and mixer outputs.

The mixer provides seven stereo aux buses, three groups, a reverb bus that can alternately serve as a fourth group, a Main Mix bus and a separate Monitor bus that doubles a solo bus. Effects include classic reverb, 4-band modeled analog EQ, modeled vintage compression and gating. A flexible matrix routing grid makes it easy for users to route audio to and from the mixer, the computer and audio network streams, including the ability to split a single input (or input pair) to multiple destinations.

All three interfaces are equipped with a single AVB Ethernet network port. Developed by the IEEE, the 802.1 Audio Video Bridging (AVB) extension to the Ethernet standard has been engineered to deliver reliable, scalable, low-latency, high-bandwidth audio networking. Users can add a second MOTU interface to their system with a Cat-5e Ethernet cable or build a system of three to five interfaces connected to the 5-port, 1-Gigabit MOTU AVB Switch (sold separately). No special IT expertise or network configuration is required, as AVB is designed to be plug-and-play.

By daisy-chaining additional MOTU switches (or third-party AVB switches), large-scale networks can be installed in multi-room studios, performance venues, houses of worship, university music departments and other similar facilities, using existing Ethernet cabling infrastructure and long cable runs (up to 100 meters) between devices and switches. Multiple computers can be connected throughout the network, and all devices and computers on the network can route audio to/from all others.

Each MOTU interface on an AVB network, or any computer connected through Thunderbolt, can simultaneously send and receive 128 channels of network audio I/O. A MOTU AVB network supports over 512 audio channels, and point-to-point network latency is fixed at only 30 samples (0.625 ms), even over multiple switches, when operating the system at 48 kHz.

AVB provides unified, system-wide, precision clocking and synchronization that is measured in nanoseconds. Users can simply click the “Become Clock Master” button in the control software to immediately resolve all devices on the network to the chosen master device.

Any available network port, including the extra (sixth) Ethernet port on the MOTU AVB switch, can be connected to a standard Wi-Fi router (such as an Apple Airport) or directly to a local Ethernet network, to give the user, or multiple users, complete access to device settings, audio routing features and the 48-channel mixer from web app software running on their laptop, iPad, tablet, and smart phone—even multiple devices simultaneously—from anywhere.

Unlike other audio interfaces, which are controlled from a software application running on a computer, the control software for the 1248, 8M and 16A is a web app served from the unit itself to any web browser running on any web client connected through Thunderbolt, USB, Ethernet, or Wi-Fi.

Through Wi-Fi, for example, each musician in a band could control their own headphone mix from their own iPad, with no computer necessary, even when operating just a single interface such as the 1248. If the network is connected to the internet, users can update the firmware in their MOTU interfaces (and switches) from MOTU’s cloud servers with just a few clicks.

All three interfaces, plus the MOTU AVB switch, are now shipping. Price for each interface is $1,495 USD. Price for the MOTU AVB switch is $295 USD.




Posted by Keith Clark on 09/04 at 01:16 PM
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Wednesday, September 03, 2014

Roland Introduces Mobile UA Compact USB Audio Interface

Provides up to four channels of DSD and PCM audio playback in a package not much bigger than a deck of playing cards

Roland has announced the Mobile UA, a new USB audio interface providing up to four channels of DSD and PCM audio playback in a package not much bigger than a deck of playing cards.

The Mobile UA is one of the smallest USB audio interfaces that support both DSD and ASIO. It offers native playback of DSD audio sampled at 2.8 MHz, and Roland’s proprietary S1LKi audio engine also reproduces traditional PCM audio at rates of 44.1 kHz and above with precision.

Using the same type of 1-bit D/A converter used in DSD, the resulting conversion produces very smooth, unclouded sound, particularly when compared to standard D/A conversion in PCM-based devices.

Inheriting Roland’s VS Streaming technology for low latency and stable operation, the Mobile UA provides up to four simultaneous channels of audio playback. Both USB audio streaming and DSP are processed on a single custom chip, offering users high performance for playing virtual instruments and editing tracks in DAW software. ASIO and Core Audio drivers are available for compatibility with popular Windows and Mac audio applications.

The Mobile UA is equipped with quality TRS mini-jacks on each side of the unit, providing two stereo audio outputs for monitoring with headphones or connecting to external sound systems. The main audio signal can be sent to both outputs, allowing two people to monitor the same sound at one time with headphones. Alternately, a musician can use one jack for sending the main audio signal to studio loudspeakers or a PA, and use the other for monitoring a click track and/or cue mix via headphones or stage monitors.

The Mobile UA features a high-performance headphone amplifier with 158 MW + 158 MW output at 40 ohms. This enables the use of a wide variety of headphone types and provides plenty of volume for monitoring in situations where isolation from ambient sounds, nearby performers, and crowd noise is preferred.


Posted by Keith Clark on 09/03 at 02:12 PM
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Tuesday, September 02, 2014

ATC Launches SCM20ASL Pro (V2) & SCM20PSL Pro Reference Nearfield Monitors

Active model replaces previous-generation SCM20ASL Pro while passive model is all new, providing an entry point into ATC at a lower price point

ATC has introduced the new SCM20ASL Pro (V2) active and SCM20PSL Pro passive reference nearfield monitors. The active model replaces the previous-generation SCM20ASL Pro, while the passive model is all new, providing an entry point into ATC studio monitoring at a lower price point.

Both models incorporate ATC’s drive units, hand-built in its UK facility. The new SH-25-76S 1-inch soft dome tweeter is the first to be designed and built by ATC,the result of six years of research and development by managing director Billy Woodman and R&D engineer Richard Newman.

“The tweeter is designed and built with the same no-compromise philosophy as all other ATC drive units,” Newman notes. “The design takes notes from the highly-regarded ATC midrange dome by utilizing a dual-suspension design, negating the requirement for Ferro-fluid, and avoiding the detrimental effects of this drying out over time, a feature considered to be of utmost importance for longterm consistency.”

The large neodymium motor with heat-treated top plate is optimized to ensure an extended frequency response (-6 dB @ 26 kHz) and low non-linear distortion. The geometry of the waveguide is designed for optimum dispersion and made from a precision-machined alloy so that the entire structure is rigid and free from resonances.

The bass/mid driver used in both loudspeakers is a proprietary 6-inch device, constructed with a 3-inch voice coil and a short-coil, long-gap topology, it combines high power handling and low power compression. Unique to the drive unit is ATC’s Super Linear technology, which by employing specialist materials in the magnetic circuit, reduces third harmonic distortion in the lower midrange.

The electronics in the active design have also had considerable development time invested in them, resulting in reduced noise and distortion (a further -10 dB @ 10 kHz) and a reduced operating temperature for improved reliability. The amplifier design is a revised version of ATC’s discrete MOSFET Class A/B design with 200 watts and 50 watts continuous power available for the bass and high frequency sections, respectively.

The user controls have also been improved over the previous generation with more flexible input sensitivity controls and a revised low frequency shelf control to help achieve good balance in difficult acoustic conditions. The amplifier includes protection circuits for both DC offset and thermal overload.

The cabinet has been restyled to more closely follow the larger monitors in ATC’s professional range and is constructed from heavily-braced MDF. Highly damped, elastometric panels are bonded and stapled to the cabinet’s inner walls to suppress cabinet panel resonances, while the enclosure’s front panel is heavily radiused to reduce cabinet diffraction, improving the frequency response and imaging.

The loudspeaker can be wall mounted via a K&M 24120 Wall Mount (available separately). Note that the cabinet requires modification to accept the “top-hat” mount.

The SCM20PSL Pro passive high-performance loudspeakers carry a UK RRP of £2,083.00 GBP (plus VAT) per pair; the SCM20ASL Pro (V2) active high-performance loudspeakers carry a UK RRP of £3,647.00 GBP (plus VAT) per pair. Both are available worldwide from any authorised ATC stockist.


Posted by Keith Clark on 09/02 at 02:28 PM

In The Studio: 10 Steps For Mixing With Mastering In Mind

This article is provided by Bobby Owsinski.

One of the things that mastering pros complain about is that so few mixers actually think about how the things they do while mixing might affect mastering.

Mastering is the final creative place in the production process where a mix can be altered, but the mix won’t necessarily be improved unless you help the mastering engineer out by following some very simple tips when you’re mixing.

Here are 10 steps for mixing with mastering in mind compiled from the latest 3rd edition of The Mastering Engineer’s Handbook. The tips apply not only to online distribution, but CD and vinyl as well.
Regardless of if you master your final mixes yourself or take them to a mastering engineer, things will go a lot faster if you prepare for mastering ahead of time. Nothing is so exasperating to all involved as not knowing which mix is the correct one or forgetting the file name.

Here are some tips to get your tracks “mastering ready.”

1. Don’t Over-EQ When Mixing: A mix is over-EQed when it has big spikes in its frequency response as a result of trying to make one or more instruments sit better in the mix. This can make your mix tear your head off because it’s too bright, or have a huge and unnatural sounding bottom. In general, mastering engineers can do a better job for you if your mix is on the dull side rather than too bright. Likewise, it’s better to be light on the bottom end than have too much.

2. Don’t Over-Compress When Mixing: Over-compression means that you’ve added so much mix bus compression that the mix is robbed of all it’s life. You can tell that a mix has been over-compressed not only by its sound, but by the way its waveform is flat-lined on the DAW timeline. You might as well not even master if you’ve squashed it too much already.

Hypercompression deprives the mastering engineer of one of his major abilities to help your project. Squash it for your friends, squash it for your clients, but leave some dynamics in the song so the mastering engineer is better able to do his thing. In general, it’s best to compress and control levels on an individual track basis and not as much on the stereo bus except to prevent digital overs.

3. Having The Levels Match Between Songs Is Not Important: Just make your mixes sound great, because matching levels between songs is one of the reasons you master in the first place.

4. Getting Hot Mix Levels Is Not Important: You still have plenty of headroom even if you print your mix with peaks reaching -10 dB or so. Leave it to the mastering engineer to get those hot levels. It’s another reason why you master.

5. Watch Your Fades and Trims: If you trim the heads and tails of your track too tightly, you might discover that you’ve trimmed a reverb trail or essential attack or breath. Leave a little room and perfect it in mastering where you will probably hear things better.

6. Make Sure To Print The Highest Resolution Mixes You Can: Lossy formats like MP3s, Windows Media or Real Audio, and even audio CDs, won’t cut it and will give you an inferior product in the end. Print the highest resolution mixes possible by staying at the same resolution as the tracks were recorded at. In other words, if the tracks were cut at a sample rate of 96 kHz/24 bit, that’s the resolution your mix should be. If it’s at 44.1 kHz/24 bit, that’s the resolution the mix should be.

7. Alternate Mixes Can Be Your Friend: A vocal up/down or instrument-only mix can be a life-saver when mastering. Things that aren’t apparent while mixing sometimes jump right out during mastering and having an alternative mix around can sometimes provide a quick fix and keep you from having to remix. Make sure you document them properly though.

8. Check Your Phase When Mixing: It can be a real shock when you get to the mastering studio and the engineer begins to check for mono compatibility and the lead singer or guitar solo disappears from the mix because something in the track is out of phase. Even though this was more of a problem in the days of vinyl and AM radio, it’s still an important point since many so-called stereo sources (such as television) are either pseudo-stereo or only stereo some of the time. Check it and fix it before you get there.

9. Know Your Song Sequence: Song sequencing takes a lot of thought in order to make an album flow, so you really don’t want to leave that until the mastering session. If you’re cutting vinyl, remember that you need two sequences—one for each side. Remember, the masters can’t be completed without the sequence. Also, cutting vinyl is a one-shot deal with no undo’s like on a workstation. It’ll cost you money every time you change your mind.

10. Have Your Songs Timed Out: This is important if you’re going to be making a CD or vinyl record. First, you want to make sure that your project can easily fit on a CD, if that’s your release format. Most CDs have a total time of just under 80 minutes. When mastering for vinyl, cumulative time is important because the mastering engineer must know the total time per side before he starts cutting. Due to the physical limitations of the disc, you’re limited to a maximum of about 25 minutes per side if you want the record to be nice and loud.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and go here for more info and to acquire a copy of The Mastering Engineer’s Handbook.

Posted by Keith Clark on 09/02 at 02:20 PM
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Second Edition Of “Small Signal Audio Design” By Douglas Self Now Available From Focal Press

Provides an extensive repertoire of circuits that can be put together to make almost any type of audio system.

Focal Press has just released the second edition of Small Signal Audio Design by Douglas Self, providing ample coverage of preamplifiers and mixers, as well as a new chapter on headphone amplifiers. The handbook provides an extensive repertoire of circuits that can be put together to make almost any type of audio system.

Essential points of theory that bear on practical performance are lucidly and thoroughly explained, with the mathematics kept to a relative minimum. Self’s background in design for manufacture means that he keeps a wary eye on the cost of things. The book also includes a chapter on power-supplies, full of practical ways to keep both the ripple and the cost down, showing how to power everything.

The book also teaches how to:

—Make amplifiers with apparently impossibly low noise

—Design discrete circuitry that can handle enormous signals with vanishingly low distortion

—Use humble low-gain transistors to make an amplifier with an input impedance of more than 50 Megohms

—Transform the performance of low-cost-opamps, how to make filters with very low noise and distortion

—Make incredibly accurate volume controls

—Make a huge variety of audio equalizers

—Make magnetic cartridge preamplifiers that have noise so low it is limited by basic physics

—Sum, switch, clip, compress, and route audio signals

The second edition is expanded throughout (with added information on new ADCs and DACs, microcontrollers, more coverage of discrete op amp design, and many other topics), and includes a completely new chapter on headphone amplifiers.

Author Douglas Self studied engineering at Cambridge University, then psychoacoustics at Sussex University. He has spent many years working at the top level of design in both the professional audio and hi-fi industries, and has taken out a number of patents in the field of audio technology. He currently acts as a consultant engineer in the field of audio design.

Find out more and get Small Signal Audio Design, 2nd Edition here.

Focal Press

Posted by Keith Clark on 09/02 at 12:36 PM
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New PreSonus StudioLive RM-series Rack-Mount Digital Mixers Offer Recallable Touch Control (Video)

Based on the StudioLive AI-series engine and controlled with UC Surface software for Mac, Windows, and iOS

Based on the StudioLive AI-series engine and controlled with UC Surface software for Mac, Windows, and iOS, the new PreSonus StudioLive RM16AI and RM32AI 32x16x3 rack-mount Active Integration digital mixers are scalable, compact, and recallable.

New UC Surface control software was specifically designed to be an interface for live mixing but is also suited for a studio environment. The layout of the mixing features provides quick, intuitive access to key parameters.

The 3U rack-mount RM16AI provides 16 locking XLR inputs with recallable XMAX Class A preamps, 8 XLR line outs, and 3 main outs (left, right, and mono/center), 32 internal channels and 25 buses, a 52x34 FireWire 800 recording interface, 96 kHz operation, and extensive signal processing.

The 4U rack-mount RM32AI offers 32 inputs with recallable XMAX preamps and 16 line outputs but otherwise has the same features as the 16-input version. Both mixers offer individual 48-volt phantom power on all inputs, and a 48-volt meters button displays phantom-power assignment on the input meter grid.

The RM-series mixers’ Active Integration technology includes direct Wi-Fi and Ethernet networking and integrated Capture recording software with Virtual Soundcheck mode and Studio One Artist DAW for Mac and Windows.

Fat Channel signal processing is provided on all input channels and all buses, including a 4-band parametric EQ, compressor, gate, limiter, and more. However, unlike StudioLive AI mixers, the RM series’ recallable XMAX preamps mean that every parameter on the mixer can be saved and recalled.

The front panel provides input signal-present and clip LEDs (16 for the RM16AI, 32 for the RM32AI) and provides stereo RCA tape inputs, a headphone output with volume knob and source-select buttons, a mute all button that temporarily mutes all inputs and outputs, and a USB type-A jack that hosts the included Wi-Fi LAN adapter.

The rear panel contains an option slot that comes with two FireWire 800 ports, an Ethernet control port, and S/PDIF I/O. The slot also accepts the same option cards as the StudioLive AI-series digital mixers; with Dante, AVB, and Thunderbolt cards coming soon, StudioLive RM-series mixers can easily be updated to keep up with the latest networking technologies. Also on the rear are a power jack and switch, MIDI I/O, and a mirror of the line outputs on a DB25 connector for connecting to wireless in-ear systems.

In addition to UC Surface, the integrated software bundle includes PreSonus Capture live-recording software with true Virtual Soundcheck and Studio One Artist DAW. StudioLive RM-series mixers also work with PreSonus’ free QMix-AI aux-mix control software for iPhone/iPod touch, enabling musicians to control their own monitor mixes. An extensive library of tutorials and downloads completes the package.




Posted by Keith Clark on 09/02 at 07:43 AM
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Thursday, August 28, 2014

FabFilter Unveils New FabFilter Pro-Q 2 Equalizer Plug-In

Major update to the Pro-Q equalizer plug-in includes new features such as Full Screen mode, unique Natural Phase processing, Auto Gain, and more

New FabFilter Pro-Q 2 provides a major update to the Pro-Q equalizer plug-in, including new features such as Full Screen mode, unique Natural Phase processing, Auto Gain, Spectrum Grab, Gain-Q interaction, and Slope Support for all filter types.

For Pro-Q 2, FabFilter has completely redesigned the internal filter engine, improving the existing Zero Latency and Linear Phase processing modes and also introducing the Natural Phase mode. Matching the magnitude response of analog EQ’ing, Natural Phase also closely matches the analog phase response without introducing noticeable pre-ring or a long latency.

In addition, Pro-Q 2 is more than twice as CPU-efficient as its predecessor.

Very steep filter slopes of up to 96 dB/octave are now available, and Pro-Q 2 lets users change the slope of any filter type, not just the usual low/high cut filters. This makes it possible to create super-narrow bell filters, very steep or gently sloping shelves, and more. In addition to the existing bell, low/high shelf, low/high cut, and notch filters, Pro-Q 2 also offers new Band-Pass and Tilt Shelf filter types.

Pro-Q 2 also has an upgraded user interface. Spectrum Grab enables users to directly adjust peaks in the real-time spectrum analyzer display. It also offers flexible interface resizing, in addition to a Full Screen mode that makes it easier to do very precise adjustments. EQ Match lets users match the spectrum of another track in just a few seconds.

Pro-Q 2 provides the top features of the original Pro-Q as well—up to 24 EQ bands, innovative interface with multi-band selection, per-channel EQ’ing in L/R or M/S mode, real-time spectrum analyzer, intelligent solo mode, double-click text entry, stereo/mono plug-ins, Pro Tools hardware surface support. Also included are fine-tuned knobs and controllers, GPU-powered graphics acceleration, interactive MIDI Learn, undo/redo and A/B switch, Smart Parameter Interpolation for smooth parameter transitions, an extensive help file with interactive help hints, sample-accurate automation, advanced optimization, and more.

FabFilter Pro-Q 2 is now available for EUR 149, USD 199, or GBP 124, supporting both Windows and Mac OS X in VST and VST 3, Audio Units, AAX, RTAS and AudioSuite plug-in formats. Bundles with FabFilter Pro-Q 2 and other FabFilter plug-ins are also available at:

Existing FabFilter customers can purchase or upgrade to Pro-Q 2 with very attractive discounts by logging in to their online user account at

System requirements on Windows are either Windows 8, 7, Vista or XP and a VST 2/3 host, or Pro Tools, or Mac OS X 10.5 or higher with Intel processor and an Audio Units host, VST 2/3 host, or Pro Tools. Both 32-bit and 64-bit hosts are supported.


Posted by Keith Clark on 08/28 at 10:53 AM
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Mark Rahilly Joins Shure U.S. Sales Team

Covering the retail market throughout the U.S. east and southeast

Mark Rahilly has joined Shure as a senior sales manager. Based in Boston, he is covering the retail market throughout the U.S. east and southeast.

Previously, Rahilly worked with Harman Professional as a territory manager, and also worked for 10 years with Shure sales rep firm Richard Dean and Associates. He has accolades and awards of excellence for his efforts in management and sales across various brands.

“Mark has an enthusiasm for Shure and a passion for the industry,” says Abby Kaplan, national sales director, U.S. Retail Group at Shure. “He has a successful track record working with independent rep firms and developing outstanding customer relationships. I am proud to welcome Mark’s enthusiasm and expertise of Shure and look forward to his contribution to our team.”


Posted by Keith Clark on 08/28 at 09:24 AM
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Wednesday, August 27, 2014

Full Sail University’s API Vision Great Match For Students

New and advanced students at Full Sail University benefit from working on the API Vision console.

Full Sail University’s flagship recording arts degree program commissioned a fully-loaded, 64-channel API Vision analog console that was placed in Studio B of the university’s extensive studio complex just over a year ago. Since then it has been used for hundreds of session recording courses.

In addition to providing students with clear-cut examples of signal flow, the console provides the classic analog sound that continues to be revered as a benchmark of excellence in professional audio.

“The Vision a great match,” said Darren Schneider, session recording course director at Full Sail University. Schneider teaches students about signal flow, taking advantage of the Vision’s comprehensive signal path to illustrate his lessons.

“Signal flow is easy to ‘see’ on the Vision,” he said. “We run it in-line, and every section falls in order – from the preamp, to the compressor, to the EQ, to the assignments. API also built us a custom switch which allows the compressor to insert pre- or post-EQ, which is also instructive.”

Schneider notes that the Vision’s ability to simultaneously mix in stereo and surround also lends itself to new educational perspectives.

“Both our new and advanced students benefit from working with API’s undeniably great sound,” said Dana Roun, education director of audio arts at Full Sail University. “Just walking into the room is an experience for first-timers.

“As more and more students come in with exclusively digital experiences, the sight of the console inspires them. The sound is something most of them have never experienced before.”

Reliability is a big deal for Full Sail University, which operates 24/7. The Vision has been used nonstop since it was installed and there has not been a single problem, notes Roun. 

“Exposure to API gives our students real life experience with the analog sound that everyone in the industry talks about,” comments Schneider. “It enriches, and often changes, their perspectives.”


Posted by Julie Clark on 08/27 at 01:19 PM

Tuesday, August 26, 2014

Blue Announces Mo-Fi Self-Powered Headphones

Onboard 240 milliwatt amplifier takes the burden of power off of mobile devices

Blue Microphones has introduced Mo-Fi headphones (, outfitted with a built-in amplifier matched to high-powered precision drivers

The onboard 240 milliwatt amplifier takes the burden of power off of mobile devices, improving their performance in the process and helping to bridge the gap between mobility and high fidelity.

“When studio monitors transitioned from passive to active, everyone benefitted, and now we’re bringing that innovation to headphones,” says John Maier, CEO of Blue. “With a built-in amp and high-powered drivers, Mo-Fi delivers astounding accuracy, unrivaled fidelity, and detailed imaging, no matter what source you’re plugged in to. Mo-Fi frees you from the confines of the studio, enabling you to take studio-quality listening everywhere and create mixes that translate across all systems.”

Mo-Fi provides three amp settings—On, On+, and Off. When switched to On, Mo-Fi activates the amp to deliver powerful, detailed sound. On+ engages the amp’s analog low-frequency enhancement circuit to enhance bass performance. Off puts Mo-Fi in passive mode for connecting to high-output studio gear. The settings are completely analog, no DSP involved.

Instead of using a fixed “spring loaded” headband structure, Blue invented a multi-jointed headband design that provides adjustability for a variety of head shapes and sizes. The earcups—shaped like ears—stay parallel at all times, creating a tight seal for solid bass response, improved isolation, and reduction of bleed.

“To design Mo-Fi, we began by studying how and where people make and enjoy music,” says Mitch Witten, VP of Product at Blue Microphones. “We set out to completely reimagine the construction, comfort, and fit, without any preconceived notions about what headphones are ‘supposed’ to look like. The result feels less like a headphone, and more like a high-quality instrument you can wear. The listening experience is so immersive and enveloping that it makes you want to revisit your favorite recordings and see what details you’ve been missing.”

For height adjustment, Mo-Fi’s pivoting-arm design allows a wide range of motion and a personalized fit. The earcups float into place or fully extend out of the way when worn around the neck. A headband adjustment knob that allows users to select their preferred pressure and tightness while also further bossing isolation and reducing ambient noise.

The rechargeable battery, charged via micro USB, provides 12-14 hours of actual play time. Mo-Fi senses when the headphones have been removed and automatically turns off to save power. If the battery runs out, Mo-Fi will continue to play in the Off setting.

Mo-Fi (MSRP $349.99) is available now at Guitar Center, Musician’s Friend, Sweetwater, and Amazon, and is coming soon to retailers worldwide. For more information, go to

Blue Microphones

Posted by Keith Clark on 08/26 at 09:58 AM
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With Meyer Sound, AIDA Cruises Ensures Accurate Translation From Rehearsals To Stage

The self-powered Meyer systems eliminated the need to place equipment racks in the rehearsal spaces on the cruise ship.

With nearly 500 Meyer Sound loudspeakers already installed aboard its ships, AIDA Cruises of Rostock, Germany has recently added Meyer Sound UPA-1P loudspeaker systems in four performance rehearsal spaces in Hamburg. In addition to exceptional sound quality and reliability, the design team has named accurate sonic translation as a primary reason for the system choice.

“Since Meyer Sound’s goal with all products is neutrality, performers will hear the same sound on the shipboard stage as they hear in rehearsals,” says system designer Malte Polli-Holstein of Hamburg-based Three-in-One Entertainment & Consulting. “Also, all of AIDA’s ships have Meyer Sound systems in their theatres, so it only makes sense to build the rehearsal systems to offer the same high level of sound quality.”

The largest rehearsal room, AIDA 1, features four UPA-1P loudspeakers, two CQ-1 and four UPM-1P loudspeakers, two 600-HP subwoofers, and four UM-100P stage monitors. The AIDA 2 and 3 rooms each feature four UPA-1P loudspeakers, while the lofty, 10-meter high AIDA 4 space uses eight UPA-1P loudspeakers. All rooms have Galileo® loudspeaker management systems, with one Galileo 616 processor in AIDA 1 and one Galileo 408 processor in each of the other rooms. Hamburg-based Amptown System Company supplied and installed the systems.

Logistics and reliability also played a role in loudspeaker selection, according to Polli-Holstein.

“The self-powered Meyer systems eliminated the need to place equipment racks in the rehearsal spaces, since there was no available equipment room,” he says. “The robustness and reliability of the systems was critical as well. The loudspeakers are running 10 hours per day, six days a week, and any downtime would disrupt the tight rehearsal schedules.”

Meyer Sound systems are used throughout AIDA Cruises’ new Sphinx class ships, most notably in the extravagant “Theatriums.” These theatre atriums feature main systems of M1D line array loudspeakers and 700-HP subwoofers, extended by MM-4 and UPM-1P loudspeakers and M1D-Sub subwoofers for fills and delays on the various deck levels. Theatres in the Cara class ships feature CQ-1 and UPM-1P loudspeakers along with 500-HP subwoofers. The AIDAcara ship uses the legacy UPA-1C conventionally powered loudspeakers installed in 1996.

Meyer Sound

Posted by Julie Clark on 08/26 at 09:48 AM
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New Launch Control XL Hands-On Controller For Ableton Live Now Available (Video)

Similar to Launch Control but provides twice as many buttons, an extra row of knobs, and eight 60 mm faders for high-precision level control

Novation is now shipping Launch Control XL, a new hands-on controller for Ableton Live. It’s similar to Launch Control but provides twice as many buttons, an extra row of knobs, and eight 60 mm faders for high-precision level control in integrating seamlessly with Live.

The knobs are laid out in three rows of eight, just like Ableton’s mixer interface. They also have multi-colored LEDs that illuminate to distinguish between sends, EQs or any other device.

Buttons, knobs and faders can be assigned to any parameters within Ableton, allowing users to make their own layouts. Knobs can also be customized by users with custom colors, and effortlessly switch between their own mappings and Live’s built-in functionality.

Combined with Launchpad S, it provides hands-on control over everything in Ableton at once: session view, mixer, effects and instruments. It’s bud powered, with no drivers needed.

The new Launch Control XL comes with Ableton Live Lite and a library of loops and samples from Loopmasters. And, it isn’t limited to Ableton Live, also capable of controlling other major music software, or any MIDI-compatible iOS software.

Hardware Specifications
Knobs: 24 rotary pots
Faders: Eight 60mm Faders
24 assignable buttons
2 template switch buttons
26 LEDs in the buttons and 24 LEDs under the knobs
Kensington security slot

Mac OS X 10.9 Mavericks, Mac OS X 10.8 Mountain Lion
Windows 8.1, Windows 8, Windows 7
iOS 7, 6
USB-MIDI class compliant
Live 9.1.3 or later required (As with all new products with Ableton Live support, Ableton provide support only for the latest version of the latest generation of Live)



Posted by Keith Clark on 08/26 at 09:27 AM
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In The Studio: Automation Tip—How To Get Away With Fake Strings (Video)

Article provided by Home Studio Corner.

In this new video, Joe focuses on implementing “fake strings” within a song, and in particular, how to do it using volume automation.

It’s not appropriate for every song and needs to be applied carefully, with proper use of volume automation helping in getting it integrated appropriately into the mix. In other words, serving good purpose in the song without being too prominent—or, well, fake.

The video also serves as an overall primer on volume automation during the recording phase, as opposed to waiting until the end of the song. And as always, Joe provides examples, utilizing a track he’s currently working on to reinforce the discussion.


Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

Posted by Keith Clark on 08/26 at 07:55 AM
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DPA Microphones Expands Sales Force In U.S., Canada & Latin America

Three new area sales manager, general manager promoted

DPA Microphones has appointed Christopher Spahr, Pedro Rocha and Leonardo Romero as area sales managers for the eastern U.S., western U.S. and southern U.S./Latin America/Canada, respectively. The company has also promoted Shan Siebert to general manager of the Longmont, Colorado-based office.

“This announcement comes at a very exciting time for DPA Microphones’ U.S. operations,” says Eric Mayer, president of DPA, Inc., the U.S. branch of DPA Microphones. “Not only does this serve as an example of how our brand recognition has expanded, but it also means that we’ll be able to further grow our fan base throughout the Americas. In addition, these appointments will allow me to perform my day-to-day strategy, sales and marketing responsibilities more effectively, giving us the opportunity for even greater success.”

Fluent in English, Spanish and Portuguese, Romero has more than 20 years of experience in the U.S., Latin American and Middle Eastern markets. In addition to joining DPA, Inc. as area sales manager for southern U.S., he will also oversee distributors in Canada and Latin America. He comes to the company from Teldyne Reson, manufacturer of high-tech sonar and hydrophone devices, where he served as Latin America sales director.

With a bachelor’s degree in economics from the University of California, Irvine, Romero has held sales positions for a variety of companies across the oil, printing and software markets, including Lub-Line Corporation; OneVision Software, AG; Esko-Graphics; Budde International, and Newgen Systems. He will build on these experiences to help with DPA’s presence in the church, theme park and cruise ship markets.

With a bachelor’s degree in business administration from Barry University and a recording arts associate’s degree from Full Sail University, Spahr’s 20-year career spans the music, recording arts and pro sound industries. He comes to DPA as area sales manager for eastern U.S. from RTW, where he oversaw U.S. sales and operations.

Prior to that, he spent seven years as a market development manager and certified U.S. RF expert for Sennheiser, in both the installed sound and professional channels. He has also served as staff engineer at Criteria Studios in Miami and performed live sound work for various concerts, corporate functions and theater applications.

A former film sound instructor with The Los Angeles Film School, Rocha joins DPA as area sales manager for the western U.S. In addition to his bachelor’s degree in electronics and telecommunications engineering from the Santo Domingo Institute of Technology, he also earned a certificate degree in audio engineering from the Musician’s Institute and attended Full Sail University’s online entertainment business master’s program.

Prior to his teaching position, Rocha served as an Avid Pro Tools marketing specialist and as a pro audio and musical instrument sales representative for a variety of retailers, which will serve him well for his current position. In addition to his responsibilities of west coast sales, he is also be tasked with overseeing growth in the film and music industries.

Building on his five-year tenure with the company, Siebert has been promoted to general manager from his role as service manager, which he has held since 2010. In this new position, Siebert is overseeing the day-to-day operations of the U.S. office, including distribution, technical assistance and loan facilitation.

Prior to joining DPA, Siebert served as a radio installer in the United States Army for four years, learning to solder and repair electrical equipment, and later earned a pastry degree that led to his own dessert company, where he learned the intricacies of business management.

DPA Microphones

Posted by Keith Clark on 08/26 at 06:59 AM
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