Studio

Friday, January 24, 2014

TransAudio Group Debuts Bock iFet Condenser Microphone At Winter NAMM

Bock iFet phantom-powered condenser microphone.

Engineer David Bock has yet again captured the unmistakable tone of a cherished vintage microphone. The newly introduced Bock iFet phantom-powered condenser microphone possesses the sonic signature of a classic fet47 mic plus deeper bass and higher highs.

The iFet also features two completely separate amplifier circuits. The first circuit uses one FET and four transformers to beautifully capture high-SPL sources, such as drums and electric guitar. The second circuit uses one FET only and is ideally suited for quieter sources, such as voice.

The two-circuit design effectively doubles the size of the Bock iFet’s sonic palette.

“There is a real art and science to designing and manufacturing a modern microphone with a specific sound, that embodies the sonic characteristics of a loved classic vintage microphone,” said Brad Lunde, president of TransAudio Group, the U.S. distributor of Bock’s entire line of high-end studio microphones. “David Bock understands the fundamentals and nuances of microphone design with tremendous depth, and he is passionate about building great microphones that will stand the test of time in the studio.

“With its dual-circuitry design, the Bock iFet is a great studio vocal mic – with all of the depth, character, and authenticity you would find in a vintage studio condenser many times the iFet’s price – and also a great studio high SPL input instrument and kick drum microphone.

“That should make it especially appealing to individuals who are on a budget yet are unwilling to fall short of perfection.”

The Bock iFet Microphone is now shipping and has a US-MSRP of $2,150.00.

TransAudio Group

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Posted by Julie Clark on 01/24 at 11:12 AM
RecordingNewsProductMicrophoneStudioPermalink

Tech Talk: Meaningful Audio Metering

What is this thing really telling me?

Is your audio gear operating in the optimum part of its dynamic range? Does excessive noise or occasional distortion plague your system? Does it become difficult during a long, loud event to discern the loudness of the system, forcing you to keep a sound level meter handy to avoid excess levels?

All of these problems can be solved with an audio meter. Here’s a look at some of the major meter types as well as what they are for.

A number of standards exist for the response characteristics of the meters used by audio products. The two major types are the Volume Indicator (VI) and the Peak Program Meter (PPM). The VI is often referred to as a VU meter, since the meter indication is the Volume Unit or VU.

In today’s audio world it is getting increasingly difficult to find a meter that adheres exactly to a standard. This doesn’t mean that the meter isn’t useful, it just means that one can’t be completely certain of what they are monitoring when they observe it.

I try to check the characteristics of meters every chance I get, just to get a feel for what is out there in the market place.

Volume Indicator
The reading on a VI tracks the loudness component of the signal, which is approximately related to the Root-Mean- Square (RMS) voltage at the output of the device.

This is a vital piece of information to the system operator, not only because it relates to loudness, but because it also relates to the applied power to the loudspeaker. The visual monitoring of a properly calibrated VI can prevent loudspeaker thermal damage and excessive sound level to the audience.

An analog volume indicator (VU meter).

The zero reference on a VI is typically +4 dBu and the range is -20 dB to +3 dB. Since the meter is blind to peaks due to its slow response time (300 ms), a peak LED may be included. The VI is a live sound operator’s best friend, because it gives them visual feedback as to the loudness of the system that can be trusted when the ears become fatigued.

Peak Program Meter
The reading on a PPM tracks the peaks of the audio waveform, but not quite. The true peaks can exceed the indicated peak by up to 8 dB. The meter’s ballistics were designed to ignore very short term peaks for which clipping would not be audible. PPMs are useful for getting the full, usable audio level out of a device.

A “quasi” PPM.

There are several flavors, each having a different scale, but on sound reinforcement mixers meter zero is typically around 1 VRMS and levels up to +16 dB are indicated. They are blind to loudness due to their short integration time.

True Peak Program Meter
Peak monitoring is especially important when making a recording, where it is necessary to avoid clipping of the signal. True PPM’s usually have 0 dB at the top of the scale, with meter zero indicating clipping of the waveform. The response time is instantaneous (practically) so clipping of even very short term peaks is indicated.

Software WAV editors and digital audio workstations often include True PPMs. While quite useful in recording applications, where the objective is to completely avoid clipping, in the world of live sound some short term clipping is typically inaudible, and even unavoidable at high sound levels.

In-the-Trenches Test
The main thing that one needs to know when they encounter a meter of either type is, “What is this thing telling me?” It’s not always apparent from simply observing the meter, with or without program material.

I’ve devised a number of test tracks over the years to determine meter characteristics, but I recently found a much simpler way to know what a meter is showing. It involves a piece of test gear that most audio people already have or should have.

A Reference
The NTI Minirator is a battery-powered signal generator that produces a number of useful test stimuli for sound system work. These include pink noise, white noise, sine waves at 1/3-octaves and a number of others.

One of the most useful features of this handy device is that the actual output level is indicated on the LCD screen. This voltage can be indicated in volts, dBV, or dBu. I keep mine set to dBV.

A software true PPM.

Now, here’s the important part. When the meter is set to produce say -10 dBV (0.316 Vrms) at its output for a sine wave, the same level is produced if the user switches to other waveforms, i.e.. pink noise.

So, I have two signals that are the same RMS level with different (but known) crest factors, and they’re just a menu selection away from each other.

This is quite handy for quickly determining the properties of a meter. Feed a 400 Hz sine wave into the mixer, and bring the level up to 0 dB.

Make sure your amplifiers are turned off or down, because if they’re maxed out you could damage your loudspeakers by operating the mixer at these levels. With 0 dB established, switch over to the pink noise stimulus.

The virtual Minirator runs on PC and uses the sound card output, available via their website. The physical Minirator shown has been updated, and I suggest the MRPro version because it stores and plays WAV files.

Peak or RMS?
A VI will produce the same indication for a 0 dBV sine wave as it does for 0 dBV of pink noise, since it is tracking the approximate RMS level of the signal.

A PPM will read approximately 9 dB higher for 0 dBV of pink noise than it does for a 0 dBV sine wave, since it is tracking the signal peaks.

The NTI Audio Minirator MR-Pro.

A True PPM will read 12-14 dB higher for pink noise than for the sine wave, since it is tracking the true peaks in the waveform.

So, with this simple test, you immediately have a pretty good idea of the type of meter that you have - VI, PPM or True PPM. The most useful meters are those that indicate both, or are selectable between several meter ballistics.

Audio meters aren’t just glorified “signal present” or “clipping” indicators. They allow you to see the level of the audio signal so that you can achieve the best performance from your system.

Spend some time with your audio gear (especially the mixer) to understand what is indicated by the metering. This will allow you to get the best signal-to-noise ratio from the device, and provide some much needed assistance to fatigued ears when mixing live.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world. For more information go to synaudcon.com.

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Posted by Keith Clark on 01/24 at 10:18 AM
Live SoundFeatureBlogStudy HallDigitalEducationMeasurementMonitoringSignalStudioSystemPermalink

Thursday, January 23, 2014

iZotope and BT Release BreakTweaker

iZotope, Inc. and GRAMMY-nominated composer and technologist BT, have combined creative forces to bring you a drum sculpting and beat sequencing environment that blurs the line between rhythm and melody.

“We are a company that loves to make innovative products, taking the best of the past and forging into the future. We were able to collaborate with BT’s forward-thinking vision on BreakTweaker, and we’re proud to offer something incredibly unique and brand-new to the world of drum machines,” says iZotope Product Manager, Jack Cote.

Powered by three distinct modules, the Sequencer, the Generator, and a futuristic MicroEdit Engine, BreakTweaker is a creative rhythmic instrument that can be used with any DAW and MIDI controller. Perfect for anyone looking to create truly original and dynamic beats, it’s a new platform for rhythmic composition.

Key Features

  Manipulate audio at a molecular level: control pitch, rhythm, and texture at the finest resolution available
  Escape traditional drum grids: create complex polyrhythmic beats with unique isorhythm and playback speed settings
  Get over 2 GB of professional, royalty-free content: explore presets, drum samples, and wavetables designed by today’s top musicians and DJs, including BT
  Craft your own drum sounds: blend drum samples with robust synthesis features to generate compelling hybrid sounds
  Take control of your beats: Easily trigger and sequence complex patterns and samples using any MIDI controller

Following their joint release of the live re-mixing plug-in, Stutter Edit, BT and iZotope are now back again with BreakTweaker. Featuring BT’s patented micro edit technology based on pioneering rhythmic sound design research, BreakTweaker aims to change the way we think about rhythm and pitch.

“I’ve always been intrigued by the way humans perceive rhythm, particularly the threshold point of where the ear perceives rhythm as pitch,” describes BT,  “The idea of exploring and exploiting this threshold inspired BreakTweaker, a tool where I could finally realize rhythmic possibilities that I once imagined but had never before been able to hear.”

View the video demonstration:

iZotope
BreakTweaker

For more information, visit the BreakTweaker website:  http://www.izotope.com/breaktweaker and watch the BreakTweaker overview video: youtu.be/ErVxMz5tVek

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Posted by Julie Clark on 01/23 at 04:09 PM
Live SoundRecordingNewsEngineerSoftwareStudioPermalink

Apogee Announces New MiC 96k Professional Microphone For iPad, iPhone And Mac

Apogee Electronics introduces MiC 96k digital microphone for iPad, iPhone and Mac.

Apogee Electronics is pleased to introduce MiC 96k, a professional digital microphone for iPad, iPhone and Mac.

The MiC 96k can be used to record vocals, voice overs, acoustic guitar, piano, drums or anything in-between. Inspired by the most revered and classic microphones in history, MiC 96k is designed to sound amazing and be easy for anyone to use with their iPhone, iPad, iPod touch or Mac.

Introduced in 2011, the original MiC has become the mobile microphone of choice for both aspiring and professional artists looking for that big-studio sound. The new MiC 96k, which features the same look and portable form factor as the original, now provides the ability to make higher fidelity recordings – up to 24-bit/96kHz – and includes an iOS Lighting cable as well as a microphone stand adapter in the box.

Like its predecessor, MiC 96k also includes an iOS 30-pin cable, Mac USB cable, and table-top tripod stand.

MiC 96k Highlights:

  PureDIGITAL connection for pristine sound quality
  Designed for voice and acoustic instrument recording
  Studio quality cardioid condenser microphone
  Up to 96kHz, 24-bit analog-to-digital recording
  Works with iPhone, iPad, iPod touch and Mac
  Includes iOS Lightning cable, iOS 30-pin cable, Mac USB cable
  Simple setup, you can start recording in minutes
  Apogee engineered microphone preamp with up to 40dB of gain
  Control knob allows easy input level adjustment
  Multicolor LED for status indication and input level monitoring
  All metal construction
  Microphone stand adapter included
  No batteries or external power required
  Compatible with GarageBand, Logic Pro and Pro Tools
  Made in the U.S.A.

Apogee Electronics

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Posted by Julie Clark on 01/23 at 11:12 AM
RecordingNewsProductDigitalMicrophoneStudioPermalink

In The Studio: My Top 10 Mic Preamps

This article is provided by Bobby Owsinski.

 
Mic preamps are an essential part of recording, and many of us tailor our tracking and overdubs around what’s available.

I’ve been asked a lot lately what my favorite mic preamps are, so I thought that today’s post was a good time to do that.

1. Hardy M-1—My all-time favorite, you can’t beat the clarity and transparency with just a hint of color. When I think of the Hardy, the word “power” comes to mind in that everything recorded with it sounds powerful.

2. API 212/3/12/512 —In my mind, API is the sound of rock. It can be gritty and aggressive sounding in a good way. A big favorite on drums and guitars.

3. GML 8302/04—When I want clean, deep and round, this is the one I’d reach for. I was lucky enough to do several albums on a GML console. The sounds were almost instantly great without working too hard.

4. Shadow Hills GAMA—A very versatile amp with lots of different sounds, it can almost be a combination of the above three.

5. Great River MP-2NV—To me the MP-2NV is like a better sounding Neve 1073. When you want that sound, this is a good way to go.

6. AMEK Angela/2500—Very underrated audio gear in general, AMEK preamps sound like a more modern English console (which they are), but better than the typical SSL. Big and round with a touch of aggressiveness. The EQs were great too.

7. Universal Audio 8110—It’s a shame this one isn’t made any more. It’s eight channels of versatility in that you could make it sound a number of different ways, but it always sounds great.

8. PreSonus M80—Here’s another one that’s no longer made. The M80 is eight channels of very high quality sound that’s more on the level of the previous preamps than you’d normally expect. PreSonus says there are parts in it that they can’t get anymore. What a shame.

9. Trident A-Range—The “Trident” sound to me is sort of like a British version of the API. Very rock and roll and aggressive. Other Trident consoles like the TSM and Series 80 had good sounding preamps with the same general sound, but the A-Range is the best of the bunch. I’ve had the pleasure of working on several of them over the years.

10. Golden Age Project Pre-73—Talk about bang for the buck, the Pre-73 gives you a reasonable facsimile of a Neve 1073 for a lot less money (about $350). I bought one as a gift for an assistant and was so impressed that I bought one for myself.

There are plenty of others that I’ve tried and liked, and lots more that I haven’t tried yet. I’m still of the mind that there are other factors that are more important when getting sounds (like the player, the instrument and mic placement), but having a great preamp can certainly make your job easier.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. Get the The Recording Engineer’s Handbook, Third Edition here.

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Posted by Keith Clark on 01/23 at 09:28 AM
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PreSonus Now Shipping New Temblor T10 Studio Subwoofer

Designed to complement full-range studio reference monitors

PreSonus is shipping the new Temblor T10 subwoofer, designed to complement full-range studio reference monitors, including the PreSonus Eris and Sceptre

The T10 is intended for serious personal studios and professional music-production environments, offering a tight low end due to fast and accurate transient response and providing user controls not normally available in this price range.

The T10 incorporates a 10-inch glass-composite woofer, driven by 250 watts of Class AB power; a front-firing, bass-reflex acoustic port; optimized, resonance-suppressing internal bracing; and internal damping and heat sink. Frequency response ranges from 20 Hz to 130 Hz.

The new subwoofer sports left and right, balanced XLR and 1/4-inch TRS and unbalanced RCA main inputs; if both XLR and TRS inputs are connected, the TRS input takes precedence. The T10 has left and right balanced XLR and 1/4-inch TRS outputs and an extra subwoofer output for connecting a second T10.

User controls include input gain (-30 dB to +6 dB, continuously variable) and a continuously variable low-pass filter (50 Hz to 130 Hz) that allows users to create a seamless crossover transition between their full-range studio monitors and subwoofer for a more accurate listening environment.

A switchable high-pass filter removes frequency content below 80 Hz from the full-range signal sent from the T10 outputs to the main monitors, avoiding destructive cancellation and reinforcement in the T10’s upper range.

The T10 also offers a momentary footswitch (included) that bypasses the subwoofer, high-pass filter, and Sub Out. This allows the audio source signal to pass directly through the Temblor T10 to the full-range studio monitors, enabling users to compare mixes with and without subharmonic frequencies, helping to ensure mixes will sound good on a wide variety of systems.

Polarity invert and a ground-lift switches are also provided. The T10 measures 15.75 x 15.75 x 12.6 inches and weighs 39.5 pounds. The enclosure has integrated, gravity-calibrated rubber feet for stable placement.

The Temblor T10 is available immediately. Expected MAP/street price is $399.95.

PreSonus

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Posted by Keith Clark on 01/23 at 09:15 AM
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Allen & Heath Introduces Qu-24 To Its Compact Digital Mixer Range

Offers total recall of settings, intuitive touchscreen, integrated multi-track recorder, remote I/O, multi-channel USB streaming to Mac, control app, and more

Allen & Heath has introduced the 30-input/24-output Qu-24 to its Qu Series of compact digital mixers, which also includes the rack mountable Qu-16. (Allen & Heath is distributed in the U.S. by American Music And Sound.)

Qu-24 offers total recall of settings (including 25 motorized faders and digitally controlled preamps), an intuitive touchscreen, Qu-Drive integrated multi-track recorder, dSNAKE for remote I/O and personal monitoring, multi-channel USB streaming to Mac, Qu-Pad control app, and iLive’s renowned FX library to deliver class-leading audio quality.

Qu-24 features a dedicated fader per mic input channel, 24 mic/line inputs, 3 stereo inputs, 4 FX engines with 4 dedicated sends and stereo returns, 20 mix outputs including 2 stereo matrix mix outputs and 2 stereo groups with full processing, patchable AES digital output with a further 2-channel ALT output, dedicated talkback mic input, and 2-track output.

The mixer is packed with considerable processing capability. High speed dual core DSPs provide extensive channel and FX processing, with ample room for future processing updates and functionality. Five latest generation ARM core processors run in parallel to efficiently deliver startling performance.

Qu-24 is equipped with a high resolution (800 x 480) full-color touchscreen featuring the easy to drive ‘Touch Channel’ access to channel processing, the FX racks and all the setup and system management controls. In keeping with the iLive user interface, the SuperStrip provides control knobs for a selected channel’s key processing parameters, such as gain, HPF, parametric EQ, gate threshold, compressor threshold and pan.

Qu-Drive, the mixer’s integrated 18-channel USB recorder, works with an external USB drive to record and playback multi-track and stereo audio .wav files. The USB interface can also be used to store scene and library data for archiving and later recall. If using the Qu-24 in the studio, there is also a USB audio 32 x 30 streaming interface for record and playback to Mac DAW systems.

A&H’s proprietary dSNAKE low latency audio connection enables the mixer to connect over a single Cat-5 digital snake to a remote audio rack, such as the AR84 or AR2412, and is compatible with the ME personal mixing system.

Motorized faders provide total recall of mix levels and switching between layers, which allows instant access to all channels and masters or the Graphic EQs. To customize the fader layout to suit certain applications, a third user definable layer is also available.

There is a free QuPad app giving wireless control of the mixer’s key parameters and settings, enabling the user to tweak the PA, adjust the monitors on stage, and even mix the show from the audience. The app connects to the mixer via a Wi-Fi router plugged into the Ethernet control port.

The 24 mic/line inputs feature crystal clear AnaLOGIQ total recall pad-less preamps, optimised for transparency and low harmonic distortion. In keeping with the excellent audio quality, the Qu-24 is equipped with many of the iLive pro touring series’ FX emulations, used by many engineers in place of top-end plug-ins and external FX units, including classic reverbs, gated reverbs, delays and modulators.

Rob Clark, A&H R&D director, states: “Building on the key features of the hugely successful Qu-16, Qu-24 is an exciting follow up. During the design process we aimed to retain a compact format with all mic inputs on one layer, with extra matrix, groups and FX sends to suit larger applications.”

The new Qu-24 ships in February, 2014.

Allen & Heath
American Music And Sound           

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Posted by Keith Clark on 01/23 at 08:10 AM
AVLive SoundRecordingChurch SoundNewsProductAVConsolesDigitalMicrophoneSound ReinforcementStudioPermalink

Manley Debuts CORE Reference Channel Strip

Manley Labs today announced the CORE, an analog channel strip.

Manley Labs announced the CORE, an analog channel strip.  CORE is an innovative and affordable mic preamplifier, compressor, equalizer, and limiter combo-unit that combines the greatest hits of the Manley product line with fresh technology. 

The intuitive design incorporates musical and forgiving circuitry that allows the user to concentrate on performance rather than be lost in a sea of knobs.  No other channel strip at this price point offers higher headroom or higher end sound than the CORE, which is, like all Manley products, handcrafted in Southern California.

“As more musicians are contributing to a project remotely, coached by the recording engineer over the telephone, we saw the need to provide an affordable and easy-to-use, excellent sounding recording channel for these guys. They aren’t engineers, they are musicians!” commented EveAnna Manley, president of Manley Labs.  “The CORE is feature-laden without being confusing. Its whole purpose is to give the musician the tools he needs to turn in a great sounding track.”

Front panel controls include 48V Phantom power switch, 120Hz High Pass Filter switch, Phase Invert switch. Input Attenuator (Variable Pad), Mic Pre Selectable Gain 40 dB or 60 dB (Total Gain >70 dB) and Line Amp Selectable Gain 20 dB or 40 dB,

The Mic-Line preamplifier features a hand-wound Manley IRON® input transformer with nickel laminations in a mu-metal can, Class A tube amplifying stage circuit topology (similar to the Manley VOXBOX® and Manley Dual Mono & Mono Microphone Preamplifiers), all-triode high voltage vacuum tube circuit and regulated 300 Volt B+ supply.

A quarter-inch direct input is similar to the DI in the Manley SLAM! with all-discrete solid-state circuit and 10 Meg Ohm input impedance (ideal for guitars, bass, keyboards, etc.).

The on-board compressor utilizes the ELOP technology also found in the Manley VOXBOX and is placed before the mic preamp making it virtually impossible to clip. It offers a ratio of 3:1, continuously variable Attack, Release, and Threshold controls and a silent bypass switch.

The equalizer has low and high Baxandall shelves (80 Hz and 12 kHz) with ± 12dB range and a sweepable midrange bell EQ (100 Hz – 1 kHz) or (1–10 kHz) with ±10dB range.

A fast attack FET brickwall limiter offers continuously variable threshold and release controls, a Peak Limit LED indicator and 10dB range output gain control.

A large illuminated analog display provides a 3-way meter select to read Compressor gain reduction, Mic Preamp Output level, and Main Output audio levels.

Inputs and outputs include balanced XLR mic inputs and line inputs, a front panel direct instrument 1/4” input, insert point between Mic Preamp/Compressor and EQ/Limiter via 1/4” TRS jack, balanced XLR direct output (after Preamp/Compressor section) and balanced XLR main output.

Availability and Pricing: The Manley Core is available Q2 2014 at a MSRP of US $2250.00.

Manley Labs

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Posted by Julie Clark on 01/23 at 06:41 AM
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Tuesday, January 21, 2014

In The Studio: Takes Vs. Punching (Video)

Article provided by Home Studio Corner.

 
In the following video, Joe Gilder talks about various styles of capturing a recording.

Specifically, do you prefer takes or punching? And what works best for the particular artist and session?

Takes, of course, are usually the longer form, working to get the track down in extended sessions.

On the other hand, there’s punching, which is a much shorter “stop and go” format. It can work to speed the process and help quickly fix blemishes and problem spots.

Which is best? As with most things audio, it likely comes down to a combination that works right for your style and needs. Check out Joe’s overview on the issue.

 

 

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.

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Posted by Keith Clark on 01/21 at 04:14 PM
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Hal Leonard Publishes “Musical iPad: Performing, Creating & Learning Music On Your iPad”

A guide through the most popular and productive apps for the iPad 2, iPad (3rd or 4th generation), or iPad Air and iPad

Hal Leonard has just published “Musical iPad: Performing, Creating, and Learning Music on Your iPad,” a guide through the most popular and productive apps for the iPad 2, iPad (3rd or 4th generation), or iPad Air and iPad mini running iOS 6.

It’s also a comprehensive approach to learning and making music on Apple’s popular tablet devices. Musical iPad will help readers learn the most appropriate ways to configure their iPad for music creation, connect it to other musical devices, and suggest powerful apps for all musical needs.

Musical iPad provides guidance for using the best iPad music apps and demonstrates how to apply them in musical life. The authors, experienced in the creation of music technology textbooks, training, and courses, maintain a companion website that includes useful video tutorials and updates. With Musical iPad: Performing, Creating, and Learning Music on Your iPad, you’ll learn how to:

—Use musicianship apps to help you stay in tune and keep your voice or instrument in shape
—Use cloud storage to share music and data files with other devices
—Turn the iPad into a tuner, metronome, and practice aid
—Emulate a host of acoustic and electronic instruments
—Use your iPad as a virtual sheet music resource for all your performance and practice needs
—Learn to play an instrument with your iPad
—Compose and share music on your iPad

And much more…

“Musical iPad will help you turn your mobile device into a powerful amplifier for your creativity – and turn your modest investment in a tablet device into an extremely valuable tool for learning and making music”, says David Mash, senior VP for Innovation, Strategy, and Technology at Berklee College of Music. “The well-written, easy-to-follow instructions and descriptions will get you up to speed in no time and will help you make the most of your Apple iPad.”

About the Authors
Thomas E. Rudolph, ED.D., is a an educator, performer, composer, and author of several music-related books. He develops and teaches courses for the Berklee College of Music online school, Central Connecticut State University, and Villanova University. Dr. Rudolph is co-founder of TI:ME (Technology in Music Education, http://www.ti-me.org) and, as a composer/arranger, has works published by Neil A. Kjos and Northeastern Music Publications. His articles have appeared in the Music Educators Journal, theInstrumentalist, and Downbeat Magazine. He lives in Philadelphia.

Vincent Leonard, along with Thomas E. Rudolph, is co-author of Recording in the Digital World, Finale: An Easy Guide to Music Notation , and Sibelius: A Comprehensive Guide to Music Notation. His works as a producer and composer have premiered nationally and internationally. He is also widely known as a copyist and arranger, having worked with Peter Nero, the Philly Pops Orchestra, Doc Severinsen, the London Symphony Orchestra, Chuck Mangione, and Leslie Burrs and in musicals by Duke Ellington, Alan Menken, Kurt Weill, and Mitch Leigh. He lives in Philadelphia.

Musical iPad
$19.99 (US)
200 pages, including DVD ROM

Learn more about the book and order it here.

Hal Leonard

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Posted by Keith Clark on 01/21 at 12:44 PM
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Shure Opens New Sales & Marketing Office In Miami

Company’s first location dedicated exclusively to sales, marketing, and business development in Latin America

Shure Incorporated has opened an office in Miami, the company’s first location dedicated exclusively to sales, marketing, and business development in Latin America.

As the gateway to Latin America, Miami has become the central hub for companies with business in the territory. Today, more than 300 Fortune 500 multinational corporations have regional or worldwide headquarters there.

“Latin America is a strategic region for the development of Shure business overseas, and we understand that being physically closer is key to augment our presence with clients,” says José Rivas, managing director for Shure Latin America.

Shure associates in the Miami office will work directly with existing and potential distribution centers, dealers, partners and key users in Latin America, providing faster and more personalized service and support. The new office includes business development, sales, marketing communications, artist relations and market development teams.

“Having specialized, dedicated Latin American sales, marketing, and market development Associates is critical for us to remain competitive and maintain Shure as the most trusted audio brand in Latin America,” continues Rivas. “We are now even more connected to customer needs, industry trends, and market opportunities.”

“We feel that with this new office, we’ve taken a big next step toward sustaining continued growth and leveraging our resources in the region,” notes Mark Humrichouser, general manager of Shure’s Americas Business Unit. “Shure has long valued our business relationships in Latin America. We are committed to our customers in the region, and we will continue to work to exceed their expectations.”

The office is located at the Wells Fargo building in the heart of downtown Miami at the following address:

333 Southeast 2nd Avenue, Suite 2000
Miami, FL 33131

The office will be managed by Rivas. Additional Shure Associates now located in Miami, include Ernest Russo, sales manager for Central America and Caribbean markets; Gabriel Benitez, product marketing manager; Catherine Ptasinski, marketing communications manager; and Denise Nino, marketing communications specialist. Helio Garbin, sales manager for South American markets, and Igor Del Ventura, market development manager, will report to the Miami office, but are located in Brazil.

Shure

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Posted by Keith Clark on 01/21 at 12:29 PM
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Mr. Bonzai Hosting Bold “Producers” Panel At Upcoming 2014 NAMM Show

Some of today's most creative producer/engineers will they reveal the inside story of their success

Award-winning photographer/music journalist Mr. Bonzai hosts the slashing-edge panel “Producers” in the Winter NAMM H.O.T Zone (Hands On Training) on Friday, January 24 at this year’s show in Anaheim.

Some of today’s most creative producer/engineers will they reveal the inside story of their success and the pitfalls they have encountered. Mr. Bonzai moderates, with Michael Bradford, Billy Bush and Ken Jordan.

Day: Friday, January 24, 2014
Start Time: 01:30 pm
Duration: 1 hour 30 min
Room: The Forum (203 A-B)
Presenter: Mr. Bonzai

Michael Bradford
Michael Bradford lives in L.A., but his heart and soul were made in Detroit, where he was born, and where he first learned to play music. One of the most versatile musicians and producers in music, Bradford has worked with a stunning variety of top artists. He has written, produced and engineered records for Kid Rock, Uncle Kracker, Stevie Nicks, Anita Baker, New Radicals and Beth Hart, in addition to creating dozens of orchestral arrangements for records, film and TV.

Bradford has worked extensively in film and TV, notably with composer Paul Buckmaster (Murder In Mind, The Maker) Terence Trent D’arby (The Fan), Uncle Kracker (Osmosis Jones, American Pie 2), and Miley Cyrus (Hannah Montana). Bradford is the co-writer and producer of Uncle Kracker’s #1 smash “Follow Me,” and the producer of his #1 cover of Dobie Grey’s classic “Drift Away.” He has also coproduced Kid Rock’s triple-platinum LP, “The History of Rock,” and engineered the New Radicals classic “Maybe You’ve Been Brainwashed Too.”

Bradford has recently produced and written songs for Jem, John Mellencamp, Deep Purple and Travis Tritt. He has performed live as the music director for artists Dave Stewart, Kid Rock and Ringo Starr.  Another recent highlight for Bradford was Stevie Nicks’ new album “In Your Dreams”, where he played bass on various tracks, and co-produced the song “Cheaper Than Free”. Bradford’s latest work can be heard in the Film “About Time”, where he co-produced a new recording of Ben Folds’ song “The Luckiest”, featuring Folds and a full orchestra. Bradford has also recently produced music for the film “The Getaway”, as well as the upcoming film “Life Of Crime”, based on a novel by the legendary Elmore Leonard. “The Long Night” is the first full-length album, coming soon from Michael Bradford, writer, producer and musician. A mix of rock and ambient, with some trip-hop influences, it is perhaps the missing link between Massive Attack and Pink Floyd.

Billy Bush
Record producer, engineer and mixer Billy Bush is well known for his extensive work with multi-platinum rock band Garbage.  In addition to producing, engineering and mixing records for Garbage, Bush joins the band on tour to help reconcile their technological needs with their live performance. The result has pushed the boundaries of and blurred the lines between live performance and recorded music. As a mixer, Bush’s credits include The Naked & Famous’s Passive Me, Aggressive You (Fiction), Snow Patrol’s single “In The End” from Fallen Empires (Polydor), and Neon Trees’ Picture Show, including the single “Everybody Talks” which reached #6 on the Billboard 200.

Also an accomplished producer, Bush produced, engineered, and mixed Fink’s Perfect Darkness (Ninja Tune), as well as French band Superbus’s Sunset (Polydor France) and The Boxer Rebellion’s Promises (Absentee).  Recently, Bush engineered and mixed The Naked and Famous’s song “Following Morning,” which will appear on the Dallas Buyers Club soundtrack. He also recently completed mixing Los Angeles band NO’s forthcoming album, as well as English singer and songwriter Jake Bugg’s forthcoming Shangri La (Mercury), produced by Rick Rubin. He is currently mixing the forthcoming album from Eastern Conference Champions, and is set to produce the follow up to Fink’s critically acclaimed Perfect Darkness.

Ken Jordan
Celebrated producer and songwriter Ken Jordan is one of the founding members of the Grammy-nominated, platinum-selling electronic music duo The Crystal Method. Originally formed in 1993 in Las Vegas, NV, The Crystal Method has been heralded by the Village Voice as “one of the best live dance acts on Earth.” Together with production partner Scott Kirkland, The Crystal Method have been known for over a decade for their enduring dance floor anthems (“Now Is The Time,” “Keep Hope Alive”), airwave smashes (“Trip Like I Do”) and a willingness to collaborate with an array of talent-including rock’s elite like Scott Weiland, Matisyahu, New Order’s Peter Hook, Emily Haines of Metric and Filter’s Richard Patrick.

TCM has dominated the remix, film soundtrack, television, gaming and advertising worlds, most recently helping Victoria’s Secret drop jaws with music for a TV commercial campaign and collaborating with soundtrack heavyweight Danny Elfman for several contributions to Hollywood blockbuster Real Steel.  Their platinum-status debut album Vegas (released in 1997) is one of the biggest-selling electronic albums of all time, landing them in the top five of best-selling electronic acts in America.  TCM scored the film London, as well as the themes for TV shows “Bones” and “Third Watch,” and were the first act to work with Nike for their running soundtrack series.

In his spare time, Jordan serves as a board member of the Electronic Music Alliance (EMA)-a public charity, non-profit organization and global membership alliance uniting the electronic music industry and community to be the “Sound of Change,” cultivating, collaborating and celebrating social responsibility, environmental stewardship, community building and volunteerism-an avid hockey player, environmentalist and electric car driver.

Mr. Bonzai
Award-winning photographer, filmmaker and music journalist Mr. Bonzai has written over 1,000 articles for magazines in the U.S., Europe and Asia. His photos and stories have appeared in Rolling Stone, Billboard, Mix, EQ, Pro Sound News, Keyboard, Daily Variety, Hollywood Reporter, Los Angeles Times, and the New York Times.  He has authored seven books, including “Faces of Music” (Cengage, 2006) “Music Smarts” (Berklee Press, 2009) and “John Lennon’s Tooth” (BookBaby 2012).

Go here for more information.

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Posted by Keith Clark on 01/21 at 12:13 PM
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Rupert Neve Designs Launching 551 Inductor EQ At 2014 NAMM Show

Includes three bands of EQ inspired by Neve’s most prized vintage designs

At the 2014 NAMM Show, Rupert Neve Designs is launching the 551 Inductor EQ, the first and only equalizer for the 500-series designed by Rupert Neve.

The 551 includes three bands of EQ inspired by Neve’s most prized vintage designs, along with custom-wound inductors, transformers and class-A gain blocks, it brings the thick, powerful lows and sweet highs of Rupert’s classics to the 500-series format.

The 551 echoes Neve’s classic 3-band EQ feature set, with a custom inductor, switched frequencies and a HPF. Traditional transformer-coupled inputs and outputs designed specifically for the 500-series are used for both technical performance reasons and optimum musical reproduction.

The 551 will begin shipping in late January 2014 with a U.S. list price of $950.

The 3-band, custom-tapped inductor EQ on the 551 was inspired by RND’s favorite portions of Rupert’s vintage EQ designs. The low frequency band is designed to produce a creamy, resonant bass response – however, unlike the vintage modules that inspired it, the LF band on the 551 can be used as either a shelf or a peak filter, adding punch, dimension, and control to your low end.

The 551’s inductor midrange band is ideal for sweetening vocals and instruments while bringing them forward in a mix, and its proportional “Q” response makes it well-suited for minimizing problematic frequencies.

The high-frequency band is a hybrid vintage/modern design, blending inductor circuitry with capacitor-based topologies to achieve vintage tones with enhanced control.

The high-pass filter is a 12 dB/octave design with a fixed 80 Hz frequency, and can be used in tandem with the low frequency EQ to add low-end presence without clouding the source material.

As Neve originally intended with his most prized classic designs, each EQ section uses low-feedback, class-A discrete electronics to prevent low-level artifacts and harshness from detracting from the tonal shaping.

However, the updated EQ circuit of the 551 is a decidedly modern design using techniques and components that were simply not available 35 years ago, and should not be considered a “clone”.

Both the high and low band can be switched from shelf to peak curves and offer 15 dB of boost or cut. The high band can be switched from 8 kHz to 16 kHz, and the low band can be selected at 35 Hz, 60 Hz, 100 Hz or 220 Hz.

The inductor-based mid band offers six center frequencies; 200 Hz, 350 Hz, 700 Hz, 1.5 kHz, 3 kHz and 6 kHz. The mid band also has a “Mid Hi Q” switch to narrow the bandwidth (increase the Q) of the filter.

“While creating functional 500-series modules is relatively simple,” Neve states, “designing those modules to equal their non-500-series counterparts with the current, voltage and space restraints is quite challenging. In creating our own 500-Series modules, we experimented with a number of different transformer and circuit designs to achieve the same presence and sweetness found in the Portico Series of modules.

“The result of these efforts is that outside of the slightly lowered headroom, our 500-series modules are nearly indistinguishable from standard Portico Series modules, and are perfectly suited for studios of the highest caliber.”

Another element unique to the 551 is the custom-wound inductor in the EQ circuitry. Inductors are wires wound around a coil that provide a form of frequency-dependent resistance. When they saturate, they bring out beautifully musical harmonics that give tracks the smooth, polished sound that has made Neve’s consoles and equipment so desirable for over 50 years.

With an extreme attention to detail towards variables like the winding and core materials in relation to the surrounding EQ circuitry, the 551’s custom inductor helps the EQ capture the vitality of Neve’s vintage modules, while still retaining its own sonic signature.

Rupert Neve Designs

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Posted by Keith Clark on 01/21 at 05:51 AM
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Monday, January 20, 2014

SSL Console Matrix2 Making Its NAMM Debut At 2014 Show

Matrix2 hardware device inserts can now be loaded directly from the console hardware controls with an intuitive new interface

Solid State Logic is showcasing the Matrix2 console at the 2014 NAMM show in Anaheim (booth 6900).

Since its 2008 launch, the SSL Matrix console has established a high-profile clientele who value its unique capabilities as a genuinely hybrid production platform.

Like its predecessor, the Matrix2 features a combination of SSL analog summing, streamlined integration of boutique analog outboard mic pres and processing (via its software-controlled analog patch system) and an advanced DAW control surface.

“Matrix provides production power and flexibility like no other console,” says Piers Plaskitt, CEO of Solid State Logic. “This is a significant upgrade to Matrix, which introduces a cool collection of features that new customers will enjoy and existing customers can access free of charge. This new version should see Matrix continue to provide the most elegant and streamlined way to create a true hybrid audio production environment for years to come.”

Based on feedback from customers, the new Matrix2 provides a collection of new features. One of the unique features of Matrix has always been the integrated software controlled patching of analog channel inserts. Matrix2 hardware device inserts can now be loaded directly from the console hardware controls with an intuitive new interface that facilitates loading individual processors, A/B comparison of different processors and building processor chains.

This was previously only accomplished via the remote browser software, which has now also been re-designed to provide a new “drag and drop” style interface for loading processors and building chains.

A “fader linking” system has also been added to the console, which allows two or more faders to be grouped, to facilitate stereo or 5.1 channel control or subgroup style mixing.

The A-FADA (Analogue Fader Accesses DAW Automation) summing system used in Duality, AWS and the new SSL Sigma rack has been introduced to enable the analogue faders of Matrix2 to be driven by automation data from a user’s DAW.

A-FADA enables channel automation to be performed entirely in the analogue signal path, but with the advantages of DAW automation data editing. The addition of A-FADA to Matrix2 facilitates project portability between different SSL products. The previously optional 5.1 output card will now be included as standard pre-fitted in all units.

A collection of smaller new features have also been added including “partial TR setup save and import,” which allows selected parts of the console setup to be saved and imported as setup templates, as well as new preset insert matrix “scenes,” preset insert naming tools, an “automatic dB readout” for Pro Tools users that allow the scribble strip to automatically display fader values upon touch, modifier key “press and hold” functionality for Cubase/Nuendo users, and new DAW templates for Presonus Studio One and Ableton Live.

Priced at $23,999 (£14,749 and €18,487 + tax), Matrix2 is now shipping. All of the new features (excluding the 5.1 output card) are available to existing customers as a free upgrade.


image


Solid State Logic

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Posted by Keith Clark on 01/20 at 04:51 PM
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In The Studio: How To Create Width, Height And Depth In A Mix

This article is provided by the Pro Audio Files.

 
The hallmark of a great recording/mix for me is one where the music all lives within a tangible, dimensional world. The exception being songs that call for a two-dimensional or more lo-fi approach.

In general, a recording that has width, height, and depth creates for a compelling sound.

And truthfully I think part of the magic of a song is pulling the listener into a different world — creating the illusion of that space only adds to that effect.

What is width, height, and depth? Just like in film or paintings we can create the illusion of three dimensions on a two-dimensional surface. In film, the surface is the screen, in paintings it’s the canvas, and in music it’s the stereo field.

The only difference is that with a screen or canvas, the height is already given simply by its existence, whereas with the stereo field we have to create an illusion of “top-to-bottom” dimension.

Height
So let’s start with height.

It’s a strange and interesting phenomenon that we hear high-pitched frequency content as coming from above, and the low tones coming from below.

Partially this is due to suggestion. We subconsciously equivocated “high” pitch with “high on a vertical scale”. Partially, this is due to common tweeter placement with speaker woofers most often being lower than the tweeter in vertical alignment.

Partially, this phenomenon is caused by the way low-frequency tones project. The wider dispersion of low tones allows them to reflect off the nearest surface such as your desk. Higher tones are more directional and will reach your ear without as much near reflection over short distances.

For these reasons and probably others, we tend to hear high harmonic content as “up” and low harmonic content as “down.” By creating contrast in the extremes of the frequency spectrum we can make a mix sound “tall.” If just one naturally bright element like a bell or hi hat is a touch brighter, and one low element like a kick or bass is a touch subby-er, the whole mix will expand.

Width
Width is also about contrast.

If two sounds are exactly the same and play at exactly the same time from each speaker, we perceive it as coming from a center point between the two speakers. This is often called the “phantom mono” or “phantom center.” The key here is similarity. As soon as the sounds become different or the timing becomes different they start to spread across the stereo field.

It stands to reason that two sounds that are easily localized (meaning our ear can clearly hear where the sound is coming from like a woodblock, triangle, or glockenspiel) played at different times will sound very wide. The greater the contrast between what’s happening in the left speaker versus the right speaker, the wider the image.

While this seems fairly simple, remember that in a dense mix its very easy to get harmonic stew. Sounds can begin to run together pretty easily.

A prime example is doubled guitars. If you double a guitar part four times and pan two doubles to one side and two doubles to the other, the end result is often not as wide as desired.

The key here is to create as much contrast as possible: use different guitars, amps, and/or mics and mic placement for the doubles. Create contrasting tones. This will allow the ear to hear more separation when they are panned apart.

On the subject of width, I’m also generally a proponent of using natural panning, rather than relying on chorusing, haas delays, or doublers. Two different takes of a guitar line is going to sound wider than one take sent through a doubler.

Depth
Lastly we have depth. It’s a bit tricky. There are three things to remember in terms of defining front-to-back placement:

Louder sounds closer
Brighter sounds closer
Less reverb sounds closer

And all of the converse is true as well.

Start with level. It’s so much easier to mix when you have an idea as to where something needs to live in terms of volume. Keeping in mind your front-to-back image while setting levels will make things fall into place very quickly.

In terms of tone, high-frequency sounds loses their energy faster than low frequency sounds over distance. So as a general rule of thumb you can roll off some highs to help shift things to the back. It should also be noted though that when sounds get very very close to us we tend to hear the low midrange more predominantly as well.

That “in your ear” sound is more characterized by low-mid forwardness than by brightness. It’s a bit counterintuitive but once you get a feel for it, finding the right tonal balance will make sense.

Then there’s the reverb. Reverb is a whole subject unto itself, but, suffice to say that generally the more reverb a sound has the further back it will fall.

Going further: the shorter the pre-delay, the farther the element will seem. And, reverb that is higher in late reflections rather than early reflections will also be indicative of a sound that is further away. Level, pre-delay, and early vs. late reflections — those things all work in conjunction to form a realistic spatial sound.

Of course, getting natural space at the recording stage is best. If you know you want your drums to sit back a bit, place the overheads a little further from the kit. And record the room capture from further away too.

Contrast
The linchpin to all of this is contrast. In order to make something sound very close, something else needs to sound far away. In order to make something feel high in the speakers, something else has to feel low.

This isn’t the easiest stuff when it comes to engineering records — it takes a long time to get spatiality down. But once you do, you quickly find your records having that special hard-to-place quality that just sounds “like a record.”

 
Matthew Weiss engineers from his private facility in Philadelphia, PA. A list of clients and credits are available at Weiss-Sound.com. He’s also the creator of the new Mixing Rap Vocals tutorial series.

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

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Posted by Keith Clark on 01/20 at 04:49 PM
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