AT 1
Meet the SoundField By RØDE DE NT-SF1: NT T-SF1: the most versatile microphone ne in the t world.
ONE MICROPHONE. ALL MICROPHONES. The SoundField by RØDE NT-SF1 is possibly the only mic you u will ever need. Coupled with the SoundField by RØDE Plugin, it can become a urround feed, mono shotgun mic, a stereo g-8 main pair, a complete 5.1 surround a hyper-realistic 7.1.4 with-height audio capture device, or a head-tracked 360° soundscape recorder for virtual reality. Whether you are recording music, broadcasting, capturing location sound or working at the frontiers of w, unlocking 360° video production, the NT-SF1 will enhance your workow, unlimited creative possibilities.
SoundField by RØDE Plugin Experiment with different mic orientations, polar patterns and congurations in post-production.
Based on an all-new, ultra-low noise true-condenser capsule and with RØDE’s legendary sound quality, the NT-SF1 comes with our 10-year industry-leading warranty and all accessories, including cables, shockmount, blimp and furry windshield.
AT 2
Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Assistant Editor Preshan John preshan@alchemedia.com.au Art Direction Dominic Carey dominic@alchemedia.com.au
Regular Contributors Martin Walker Paul Tingen Brad Watts Greg Walker Andy Szikla Andrew Bencina Jason Hearn Greg Simmons Mark Woods Ewan McDonald Guy Harrison Cover Photo David Bichard
Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Sophie Spencer subscriptions@alchemedia.com.au Proofreading Andrew Bencina
AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact +61 3 5331 4949 info@alchemedia.com.au www.audiotechnology.com.au PO Box 295, Ballarat VIC 3353, Australia.
All material in this magazine is copyright Š 2018 Alchemedia Publishing Pty Ltd. Apart from any fair dealing permitted under the Copyright Act, no part may be reproduced by any process with out written permission. The publishers believe all information supplied in this magazine to be correct at the time of publication. They are not in a position to make a guarantee to this effect and accept no liability in the event of any information proving inaccurate. After investigation and to the best of our knowledge and belief, prices, addresses and phone numbers were up to date at the time of publication. It is not possible for the publishers to ensure that advertisements appearing in this publication comply with the Trade Practices Act, 1974. The responsibility is on the person, company or advertising agency submitting or directing the advertisement for publication. The publishers cannot be held responsible for any errors or omissions, although every endeavour has been made to ensure complete accuracy. 19/11/2018.
AT 3
AUSTRALIA’S LEADING SUPPLIERS OF PROFESSIONAL AUDIO EQUIPMENT SINCE 1976 BEST PRICES • BEST SERVICE • BEST ADVICE • LARGEST STOCK • NATIONAL DELIVERY EDUCATION SPECIALISTS – TERTIARY, SECONDARY & STUDENT PRICING AVAILABLE ONLINE STORE www.turramusic.com.au
Expert advice on Education licensing for Institutions, Students and Teachers
Turramurra Music
Celebrates in the Music
42
Industry this Year
YEARS
Thank you to all our loyal customers! We look forward to extending our hand to all of our future customers.
PRO TOOLS HD: Full range of high quality Avid HD interfaces available – enquire now for expert advice on you Pro Tools set up
Software
LATEST VERSIONS OF THE FOLLOWING SOFTWARE AVAILABLE: Steinberg – Cubase, Wavelab + Nuendo, NEW Dorico (scoring and notation program) Avid Pro Tools + Sibelius, MOTU Digital Performer, Ableton Live, Propellerhead Reason, PreSonus Studio One + Notion CONFUSED AS TO WHAT SOFTWARE TO BUY? CONTACT OUR EXPERIENCED STAFF FOR ASSISTANCE.
Virtual Synth Heaven SOLE AUSTRALIAN AGENTS FOR SYNTHOGY IVORY PIANO SERIES
Audio Interfaces & Control Surfaces
ARTURIA / BEST SERVICE / EAST WEST / MOOG / MOTU / NATIVE INSTRUMENTS INCLUDING KOMPLETE 12 & KOMPLETE 12 ULTIMATE / SPECTRASONICS / STEINBERG / TOONTRACK / UVI / VIENNA INSTRUMENTS / & MANY MORE!
THE MOST POWERFUL AUDIO COMPUTER INTERFACES ON THE MARKET ANTELOPE / APOGEE / AUDIENT / AVID / M-AUDIO / MOTU / PRESONUS / PRISM / ROLAND / RME / TC / YAMAHA FOCUSRITE, SCARLETT & CLARETT INTERFACES AVAILABLE PRISM SOUND LYRA 2
Avid S3 – Compact, 16-Fader, EUCON-enabled, ergonomic desktop control surface that offers a streamlined yet versatile mixing solution. KOMPLETE 12 KOMPLETE 12 ULTIMATE
SPECTRASONICS KEYSCAPE OMNISPHERE AND TRILIAN
Plug-ins ANTARES / AVID / CELEMONY MELODYNE / IZOTOPE / LEXICON / MCDSP / SOFTUBE / SONNOX / SOUND TOYS / WAVES / & MANY MORE! SOFTUBE CONSOLE 1
ANTELOPE ORION 32+
FOCUSRITE REDNET 8PRE
DSP Power UAD Satellite Now Available with Thunderbolt
RME FIREFACE UFX II
NEW
UA APOLLO X SERIES
UAD-2 OCTO CARD More DSP power and Includes Analogue Classic Bundle
AUDIENT ASP880
Studio Microphones
APOGEE SYMPHONY 2
AEA / AKG / ASTON / AUDIO-TECHNICA / BEYER / BLUE / DPA / EARTHWORKS / MOJAVE / NEUMANN / RØDE / ROYER RIBBON / SE ELECTRONICS / SENNHEISER / SHURE / SLATE / TELEFUNKEN / & MANY MORE! SE ELECTRONICS TOWNSEND LABS ASTON ORIGIN RUPERT NEVE RNT TL SPHERE L22
SLATE DIGITAL RAVEN MTI
Now you can purchase Pro Tools HD software or HD native Thunderbolt with or without Pro Tools SW HD NATIVE THUNDERBOLT
HD NATIVE PCIE CARD + PRO TOOLS HD SOFTWARE
HD NATIVE THUNDERBOLT + PRO TOOLS HD SOFTWARE
HDX CORE
HD NATIVE PCIE CARD
HDX – when only the best will do. Unrivalled DSP power. THUNDERBOLT HD NATIVE AVAILABLE ENQUIRE NOW PCIe and Thunderbolt versions available with choice of high quality Avid professional interface
PRESONUS QUANTUM
PRESONUS FADERPORT 8
HDX CORE + PRO TOOLS HD SOFTWARE
PRESONUS STUDIO 26
NEW
AVID MTRX
ONE MIC GETS YOU A VIRTUAL UAD MIC LOCKER
PRESONUS STUDIO 1810 SLATE DIGITAL VRS-8 A hybrid microphone and 8 channel preamp modeling system that gives you access to virtually any signal path tone you can imagine. AT 4
CONTACT OUR PROFESSIONAL SALES TEAM FOR MORE DETAILS ON MTRX SYSTEMS, AND PRO TOOLS LICENSING OPTIONS FOR STANDARD AND PROFESSIONAL HD SYSTEMS LATEST VERSION OF AVID PRO TOOLS OR PRO TOOLS HD AVAILABLE
TURRAMURRA PROFESSIONAL STUDIO DIVISION 1263 PACIFIC HIGHWAY, TURRAMURRA, NSW. TEL: (02) 9449 8487 FAX: (02) 9449 3293 WEB: www.turramusic.com.au EMAIL: hitech_sales@turramusic.com.au
Keyboards & Sound Modules AUSTRALIA’S LARGEST STOCKIST OF KEYBOARDS ACCESS VIRUS / AKAI PRO / ALESIS / CME / DAVE SMITH INSTRUMENTS / ELEKTRON / KORG / KURZWEIL / M-AUDIO / MOOG / NORD / NOVATION / ROLAND / WALDORF / YAMAHA EURORACK MODULAR MADNESS INCLUDING:
pittsburgh modular
Metro Modular
Studio Outboard Processors & Effects WE STOCK THE BEST OF THE BEST APHEX / ART / AVALON / DANGEROUS / DRAWMER / DBX / EMPERICAL LABS / FOCUSRITE / GRACE DESIGNS / JOE MEEK / KUSH / LEXICON / MANLEY / NEVE / PRESONUS / RUPERT NEVE PORTICO / SLATE / SMART RESEARCH / SPL / SSL / TC ELECTRONICS / TOFT / UNIVERSAL AUDIO / & MANY MORE! SPL TUBE VITALIZER
MADE IN MARRICKVILLE
Studio Monitors
TIP TOP / CAT FULL OF GHOSTS / SYSTEM 80 / & MORE KORG MONOLOGUE
FULL RANGE OF 500 SERIES STUDIO PROCESSORS AVAILABLE
Studio Monitoring & Summing NEW
YAMAHA MODX8
PRESONUS ERIS SERIES AFFORDABLE STUDIO MONITORING
ANTELOPE AUDIO SATORI
High end monitoring controller. True analog monitoring and summing system.
SPL SMC 5.1
Surround sound monitor controller.
IN STORE NOW & ON DISPLAY FOR PREVIEW BAREFOOT FOOTPRINT 01 ATC SCM25
DANGEROUS MUSIC D-BOX DAVE SMITH SEQUENTIAL PROPHET X NOW AVAILABLE AND ON DISPLAY
HUGE RANGE IN STOCK AND ON SHOW! ADAM / ALESIS / ATC / BEHRINGER / DYNAUDIO / EVENT / FOCAL / FOSTEX / GENELEC / JBL / MACKIE / M AUDIO / PRESONUS / TANNOY / YAMAHA / & MANY MORE!
Analog Summing, monitor control, on-board D/A conversion, talkback, headphone cue, simultaneous input monitoring.
SPL MTC
Monitor & Talkback controller.
Analogue & Digital Mixers
ROLAND SE02
FULL RANGE OF: MACKIE MIXERS / ALLEN & HEATH /SOUNDCRAFT / TASCAM / MIDAS / YAMAHA / BEHRINGER / TOFT – NEW TOFT TRIDENT MIXERS COMING SOON WE CARRY A RANGE OF USB + DANTE COMPATIBLE MIXERS THAT INTERFACE DIRECTLY TO YOUR COMPUTER. YAMAHA TF SERIES NEW PRESONUS SL16 AFFORDABLE QUALITY
Digital Media Recorders ZOOM H1N, H2N, H4N, H5, H6, Q2N ,Q4N, Q8, F4 & F8 / MARANTZ PMD SERIES / FOSTEX FR2LE / ROLAND / TASCAM / & MANY MORE! COMPACT FLASH RANGE & MANY MORE! NEW
TASCAM DR60D
ZOOM F8N
ROLAND VR-730 Complete Mix Control: StudioLive Mixers is a compact, intuitive mixing solution for the StudioLive Mixers rackmount digital mixers and Studio One digital audio workstation.
Wide Range of Midi Controllers NOVATION PEAK
AKAI / ARTURIA / ALESIS / ICON / IK MULTIMEDIA / KORG / M AUDIO / NATIVE INSTRUMENTS / NEKTAR / NOVATION / ROLAND / ROLI / STUDIOLOGIC / WALDORF NEKTAR PANORAMA T4
Headphones AUDIO TECHNICA / BEYER / DIRECTSOUND / FOSTEX / SENNHEISER / SHURE / SONY / & MANY MORE! SENNHIESER HD280 PRO AUDIO TECHNICA ATHM50
IK MULTIMEDIA KEYS I/O HEADPHONE DISTRIBUTION AMPS AVAILABLE INCLUDING: APHEX / ART / AVENSON / BEHRINGER / FOSTEX / GRACE DESIGNS / MOTU / PRESONUS / PURPLE AUDIO / RANE / SAMSON / SPL / ETC.
NORD STAGE 3
Deep instrument control in any DAW ROLI REVOLUTIONARY with VST/AU support. Load patches directly from the keyboard and start controlling parameters immediately.
AT 5
COVER STORY
Parkway Drive’s Gamble: Recording with Their Live Engineer
22
ISSUE 52 CONTENTS
18
Producer Profile: Tristan Hoogland
View from the Bench: Audio Level Meters
38
Steinberg UR-RT4 RND-Transformer Interface
SPL Crimson 3 Audio Interface AT 6
Kamasi Washington: How Hip Hop Saved Jazz
62
56
Studio One 3 DAW
Sennheiser Evolves Its Wireless to G4
28
58
28 48
Musos Corner Celebrating 50+ Years in the Music Industry
Pro Audio Interfaces
Outboard
PreSonus Studio 192 Audio Interface
BAE 10DCF Single Channel Filter Compressor
Steinberg UR-RT4 Interface Rupert Neve Design
BAE 1073MPF Dual Channel Mic Pre with Filter
Control Surfaces
Icon QCON ProX USB DAW Controller
Antelope Audio Zen Tour - Thunderbolt & USB Interface Focusrite ISA430 MKII Producer Pack Focusrite Red 4 Pre Thunderbolt Interface with Dante
RME Fireface UCX Audio Midi Interface
Universal Audio Apollo 8 Duo Thunderbolt 2
Converters
AVID Pro Tools Dock Media Controller AVID S3 - Pro Tools Control Surface
Avalon VT-737SP Silver 737 Channel Strip
API 2500 Stereo Compressor Rack Unit
API 3124+ Four Channel Preamp
SSL Nucleus 2 Dark DAW Controller with Mic Pre’s Monitor Control & Dante Yamaha Nuage Control Surface
Neve AMS 1073DPA Dual Mic Pre
500 Series
IGS Audio iQ. Inductor Equalizer
Focusrite RedNet HD32R Pro Tools HDX Dante Bridge
IGS Audio ONE LA 500 (1LA500) “LA2A Clone”
SSL Super Analogue XRack
IGS Audio RB 500 ME Stereo Pultec style EQ IGS Audio 576 Blue Stripe Rev A - Fet Compressor Warm Audio TB12-500 Tone Beast 500 Mic Pre & DI API 550B 4 Band EQ for 500 Series SSL Solid State Logic VHD Pre Module - 500 Series SSL Solid State Logic Stereo Bus Compressor Module (500 Series) G Comp
AVID HD 8x8x8 I/O
Summing
SSL Stagebox SB i16 Mic Pre Network IO Dante
Shadow Hills Equinox Mic Pre & Summing Mixer SSL Sigma Delta - Analogue Summing Mixer Neve AMS 8816 Summing Mixer SPL MixDream XP Analog Summing Mixer 2591 Antelope Satori Monitoring Controller + R4S Remote Bundle
50 YEARS WWW.MUSOSCORNER.COM.AU
Musos Corner
Neve AMS 1073LB Mono Mic/Pre 500 Series Neve AMS 2264ALB mono limiter/compressor Rupert Neve Designs 535 Diode Bridge Comp Meris Mercury7 500 Series Digital Reverb Radial Engineering Jensen Twin Servo Mic Pre
www.musoscorner.com.au
Newcastle West NSW 2302
02 4929 2829 AT 7
GENERAL NEWS
APOGEE JAM+ Apogee has announced the all new Jam+ portable USB instrument input and output. Jam+ connects your electric guitar, bass, keyboard or any acoustic instrument with a pickup, directly to your iOS device, Mac, or Windows PC. You can also connect a dynamic microphone using an adapter that’s sold separately. In 2010, Jam was introduced in Apple stores worldwide as the first studio quality digital instrument input to connect guitars to GarageBand on iOS devices and Mac computers. Jam+ continues the evolution in the series by bringing the latest advances in
Apogee’s design and engineering to an all new metal chassis with enhanced PureDIGITAL circuitry for true-tone audio resolution. Other features on Jam+ include low latency monitoring with a Blend control, stereo output for connection to headphones or powered speakers, a Drive input mode for overdrive tones, a rugged metal body and three multicolour LED indicators. Sound Distribution: (02) 8007 3327 or www.sounddistribution.com.au
PRESONUS ATOM Presonus jumps into a whole new product category with the release of its new Atom production and performance pad controller. Atom has 16 RGB-backlit pads on it and comes bundled with Studio One Artist production software to let you both create and perform. If you’re not a Studio One user, Atom is compatible with most other music production and performance applications. The little performance powerhouse offers flexible control of virtual instruments and real-time triggering of samples and loops using 16 full-size, velocity- and pressure-sensitive RGB
AT 8
pads that can send polyphonic aftertouch, channel pressure, or MIDI CC messages, plus 20 buttons, four programmable rotary encoders and eight pad banks that can be assigned in multiple ways. It also includes MVP Loops content custom-designed for Atom. The best part? A very reasonable asking price. Link Audio: (03) 8373 4817 or info@linkaudio.com.au
MORE NEWS AT www.audiotechnology.com.au
SHURE BLUETOOTH 5 EARPHONE Shure’s new high-resolution Bluetooth 5 Earphone Communication Cable is the next generation in Shure’s foray into Bluetooth connectivity, allowing you to ‘go wireless’ with your choice of Shure’s detachable SE Sound Isolating earphones. Featuring a dedicated, high-performance headphone amplifier, the earphone accessory provides lower noise and distortion while accurately maintaining frequency response, and also supports popular audio codecs, including Qualcomm aptX audio, aptX HD, aptX Low Latency, AAC, and SBC. You get an in-line remote
and mic control and the new cable leverages the latest version of the Bluetooth wireless communication standard. Retaining a wireless range of up to 30ft, its battery capacity provides up to 10 hours of play time with up to 350 hours of standby time. The cable will be available for purchase through authorised Australian retailers. Jands: (02) 9582 0909 or www.jands.com.au
AVANTONE PRO AMPS UP Avantone Pro teams up with mix engineer Chris Lord-Alge to birth a new studio reference amplifier. The CLA-200 amp is a full-range amp built the old school way with generous VU meters, high-quality Japanese components, and a toroidal transformer linear power supply. Housed in a standard 2U rackmount chassis, the convection-cooled amplifier can generate over 200W per channel into an 8Ω load. The specs include a quoted 20Hz-20kHz ±0.5dB frequency response. Connections include XLR input and link sockets and Speakon outputs. Commenting
on the amp, Chris Lord-Alge says, “Every good pair of monitors needs a powerhouse of an amplifier to give you clear results. So now, thanks to Avantone Pro and the ears of CLA, we’re making available for the first time the return of a classically-designed amplifier for your studio.” Audio Chocolate: (03) 9813 5877 or www.audiochocolate.com.au
MORE NEWS AT www.audiotechnology.com.au
AT 9
LIVE NEWS
DPA’S TINIEST MIC CAPSULE YET DPA Microphones introduces its smallest ever high end pro audio microphone capsule as an addition to its d:screet and d:fine ranges. The d:screet CORE 6060 and 6061 Subminiature Microphones and the d:fine CORE 6066 Subminiature Headset Microphone are just 3mm in diameter – 2mm smaller than DPA’s existing 4000 series. All three incorporate DPA’s Core microphone technology that reduces distortion and increases dynamic range. The new 6000 series capsules are omnidirectional and have a frequency range of 20Hz-20kHz. The d:screet 6060
lavalier has a noise floor of 24dB A, while the d:screet 6061 lavalier and d:fine 6066 headset have noise floors of 26dB A. The d:fine 6066 also has a redesigned lightweight, one-size-fitsall headset with a spring mechanism to help it grip below the ears and to the side of the head for added security. All three are available in black and beige with more colours to be added. Amber Technology: 1800 251 367 or sales@ambertech.com.au
BLUETOOTH-EQUIPPED ZLX Electro-Voice adds two new models to its ZLX portable powered PA speakers series — the ZLX-12BT 12-inch two-way and ZLX-15BT 15-inch two-way — both of which are Bluetoothequipped for audio streaming from any mobile device. Both models share the same features as other loudspeakers in the ZLX range including QuickSmart DSP with sub/top system-match, two-band EQ, five user-programmable presets, visual monitoring of limiter status, input level control and meters, and more. The loudspeakers are powered by a 1000W Class D amplifier
AT 10
which delivers up to 127dB peak SPL. The EV-patented Signal Synchronised Transducers (SST) waveguide design provides precise and consistent coverage, and a three-handle design is integrated into a rugged composite enclosure for an optimal combination of portability and durability. Bosch: 1300 026 724 or stsales@au.bosch.com
MORE NEWS AT www.audiotechnology.com.au
BRING YOUR CREATIVE PROJECT TO LIFE EARN YOUR MASTER OF CREATIVE INDUSTRIES ALONG THE WAY
FIND OUT MORE
jmc.academy/masteraudio AT 11
POWERSOFT ARMONÍAPLUS Powersoft has radically redesigned ArmoníaPlus with a new workflow approach designed to be free from sub-menus, with every function available via either a right-click or just a hotkey. The ArmoníaPlus interface presents a single window with four main views: Workspace, System List, Operator View, and Events. Workspace is where all the main action takes place – a natural environment where system engineers can plan and work. The Workspace is divided into two parts, with amplifiers occupying the left-hand space and loudspeakers sitting on the right. You can move through four main steps: Design, Config, Tune, and Show,
each offering a variety of functions. Amplifiers can be added directly to a rack, while loudspeakers can be grouped together, vertically and horizontally as you would expect them to be in the physical realm. A special Headroom meter adapts its LEDs based on the limiters of the speaker itself, showing an at-a-glance view of how hard the speakers are been driven. Production Audio: (03) 9264 8000 or www.pavt.com.au
WAVES EMOTION LV1 V10 Waves introduces a new version of its eMotion LV1 live mixer which brings a number of new features and improvements to performance and stability. One of the most useful features v10 introduces is the ability for both monitor and FOH engineers to share the same stagebox between two LV1 mixers while setting separate gain control modes, allowing each engineer to set their nominal input levels safely and conveniently without affecting
AT 12
each other. eMotion LV1 is available in 64-/32-/16-channel varieties and allows up to eight SoundGrid-compatible plug-ins to be run directly on each channel. A number of bug fixes and minor enhancements have been implemented with v10. Sound & Music: (03) 9555 8081 or www.sound-music.com
MORE NEWS AT www.audiotechnology.com.au
STUDIO MONITOR KH 310
THE EAR OPENER
»
It is extremely detailed and revealing, but remains unfatiguing, even after many hours of listening.
« »
sound on sound magazine
»
The broad horizontal dispersion is very impressive and ensures a virtually unchanging sound image across the entire stereo panorama.
« »
soundcheck magazin
»
To sum up, the Neumann KH 310 is an excellent loudspeaker (...) it should perform well under a wide variety of conditions. resolution magazine
« »
Neumann’s KH 310 is spot-on, definitely. These monitor speakers are convincing from A to Z. xound.com
Its workmanship and feature set are first class, and it will fulfill all professional criteria. bonedo.de
The workmanship, measurements, and listening impression are all completely convincing. sound & recording magazin
« »
This 3-way design is poised to open a lot of eyes – and ears. recording magazine
«
« « Also available with digital input and delay function AT 13
SOFTWARE NEWS
NASHVILLE JOINS OUTPUT ARCADE Output Arcade is a unique loop synth instrument with new content delivered everyday. It even works with your own loops, allowing you to perform and manipulate them in real time. An Arcade kit is curated from presets of 15 loops. White keys play loops, black keys modify them and sliders change character. Output has made a cool addition to Arcade called Nashville. It brings a host of riffs on dirty fuzz tones and layer sliding pedal steel textures all played by first class Nashville artists. Output has a rep for augmenting the natural tones of sampled instruments
and turning them into self-sufficient instruments in their own right — Nashville is no different. You just may find yourself using a lap steel sample in a hip hop track, or a quacky country guitar in a pop tune. “We’ve created a sandbox and everyone’s gonna play in it a different way,” says Rob O’Block, one of the producers behind the sounds of Nashville. Output: www.output.com
WAVES’ FIRST COMPUTER Waves announces Axis One — a standardised computer customdesigned and optimised to run Waves audio applications via SoundGrid connectivity. Axis One can act as the ‘brain’ of a SoundGrid system, seamlessly interfacing with SoundGrid I/O devices and servers such as the eMotion LV1 live mixing desk. Use it at front-of-house, at monitor position, or in your broadcast AV rack – or add it to your studio for tracking, mixing and mastering. Packed in a roadworthy half-rack 2U case, Axis One is a turnkey solution for live sound, broadcast, and the
AT 14
studio. Under the hood, Axis One is powered by an Intel i5 8600 processor with 16GB DDR4 RAM, 256GB SSD storage, two HDMI ports, one DiplayPort, and nine USB ports (one of which is USB-C). It runs on Windows 10 and interacts with your SoundGrid device via the single EtherCon connector. Waves Axis One will ship early 2019. Sound & Music: (03) 9555 8081 or www.sound-music.com
MORE NEWS AT www.audiotechnology.com.au
AT 15
REVIEW
SOUNDTHEORY GULLFOSS Intelligent Equaliser Plug-In Is Gullfoss the ultimate decision maker or an unwanted back seat driver? Review: Preshan John
I make no secret of the fact that I dislike ‘intelligent’ plug-ins which make mix decisions on my behalf. Same goes with algorithmic mastering. No thanks, I have ears for a reason. Still, here I am reviewing a plug-in that sits squarely in that ‘intelligent’ camp. Perhaps I’m just more resigned to the fact that AI is going to be a larger part of my life — whether it’s predicting my email responses or my EQ curves. Gullfoss by Soundtheory is a plug-in that — as much as I hate to admit it — does the latter extremely well. Pitched mainly as a corrective tool that identifies and fixes frequency issues in a mix or instrument, it’s the plug-in’s ability to breathe life and definition into virtually anything that impressed me. Let me tell you how, then you can download the 14-day trial and hear it for yourself. VIRTUAL EARS
First and foremost, all plug-ins of the ‘I-can-doyour-job-for-you’ nature need to have a good listening algorithm — like the trained ear of a mix engineer. Using Soundtheory’s “computational auditory perception model”, Gullfoss breaks down AT 16
what it hears into two primary components: dominant (frequency content that’s most prominent in the source material), and dominated (content that’s suppressed relative to dominant frequency areas). These form the basis of how audio is then treated. The key here is that when Gullfoss says ‘dominant’, it’s primarily talking about stuff that gets in the way of clarity. A woofy bass drum, a tooaggressive filter, or an out of control resonance. Gullfoss does not allow you to set your own EQ curve. The graph is only there to show you how the plug-in is behaving in real-time as it reads and responds to incoming audio. The control you’re given over Gullfoss’ algorithm is akin to operating an automatic transmission. You can set it to Drive, Neutral or Reverse, and pump the accelerator, but the car is going to be shifting all the gears for you. The four controls on the surface of Gullfoss take a little getting used to. Recover and Tame break up the sound into those dominated or dominant parts of a sound. While Recover will generally boost the elements that are getting obscured, Tame will pull them down. They’re not binary opposite controls;
PRICE $250 CONTACT Soundtheory: www.soundtheory.com
they each affect the sound in different ways. They’re also not a simple boost or cut; both controls will raise and lower areas of the frequency spectrum. However, the net effect of turning them both up will be enhanced clarity and separation. Each control goes from 0-200%, but anything above 100% is usually taking it too far. ‘Bias’ decides whether Gullfoss leans toward favouring the Recover (positive) or Tame (negative) control for frequencies they both want to get their hands on. Brightness goes from -100% to 100% and affects the overall tone of your Tame and Recover settings; turn it up to skew the results towards a brighter tonality, or down to make everything darker. Boost is set in dB (-50dB to +50dB), and is designed to simulate low end ‘loudness’. It’s a perception control that allows you to adjust the balance of low and mid frequencies (the scoop), to compensate for changes in level. You can limit the plug-in’s working bandwidth by dragging in the red markers from either end of the spectrum, or pull them across each other to treat just a selected portion of frequencies.
F1 FIELD RECORDER + LAVALIER MIC PROFESSIONAL AUDIO. UP CLOSE.
The Zoom F1 Field Recorder + Lavalier Mic is the perfect solution for videographers who need to capture the nuance of every word with professional quality audio. Superior Sound: Newly engineered LMF-1 lavalier mic+ (up to) 24-bit/96kHz recording to micro SD card. Low Profile: Attaches neatly on belts, waistbands, or slipped concealed into a pocket. One-Touch Controls & Display: Instant access to record levels, limiter control, lo-cut filter and volume output + sunlight viewable LCD.
(top) You can limit Gullfoss' affected area by pulling in the red markers from either end, or (left) overlap them to only adjust the lows and highs. (above) This instance is biased to focus more on recovering 'dominated' areas of your sound source.
Videographer Essentials: Record Hold function, onboard limiter, battery powered (or power adapter), transfer files via card reader or directly via USB.
SELECTIVE HEARING
On the master bus, the accuracy with which Gullfoss is able to distinguish between dominant and dominated elements of a mix is incredible. A shaker tucked underneath a wall of guitars and synths would classify as dominated, and turning up Recover brings it nicely into auditory view. On the other hand, the kick and claps in a techno tune were too dominant, and the Tame control affected those quite independently. Remember, Gullfoss is using EQ curves, so if you’re using it on a master, your mix might be better served by simply turning up or down the individual fader for those dominant or dominated tracks. In other words, don’t overdo it. Gullfoss works just as well on individual tracks — electric guitars, piano, sax, strings, you name it. In my tests, the plug-in never failed to bring clarity, focus and accentuation of detail. If like me, you’re struggling with the idea of handing over control to the machines, there’s a way of thinking about Gullfoss that makes the transition more palatable. See, the real power behind Gullfoss, is not in its ability to make decisions for you, but its dynamic response. For a while now, engineers have warmed to multiband dynamics as a way to EQ, based on the amplitude of frequency content at a given time. Often using it to occasionally tame a certain area of the frequency spectrum without killing the source’s tone by notching it out with an EQ. Gullfoss can play a similar role, just 100 times more in-depth. As a tool for fixing problematic tracks, it’s invaluable. With Gullfoss, I will gladly surrender my rights to a row of Gain, Frequency and Q knobs in exchange for its intelligent ability to treat audio in an utterly musical fashion. Soundtheory’s got a winner here and I’ll be using it often.
Proudly distributed in Australia by Dynamic Music
dynamicmusic.com.au
AT 17 Dynamic_Zoom-F1.indd 1
19/3/18 4:02 pm
STUDIO FOCUS:
PUZZLE FACTORY A river cuts through rock, as they say, because of its persistence. And Flood is persistent. “Flood looked me in the eyes and said, ‘You should move to London.’” recalled Australian producer Dax Liniere, who at the time was tossing around the idea of moving overseas when he mentioned it to the famous UK producer. “I was saying how expensive it was. He looked me in the eyes again, pointed at me and said, ‘You should move to London.’ A few months later, it clicked. It was like a cheesy movie, I could hear his voice and see him pointing.” Up until a few years ago, Dax Liniere was comfortably living the self-employed studio owner life in Canberra. That all changed when Dax AT 18
managed to snag a Churchill Fellowship which sent the producer around the world on a fact-finding mission. The idea was to embed himself in various professional environments to glean insights he could bring back to his homeland. It hasn’t quite worked out according to plan. While Dax did learn a whole lot from his time with Alan Moulder and Flood at their Assault & Battery studio in the UK, a few years later we lost Dax to the UK. After the initial Fellowship trip, he went back on another fact-finding mission in November 2015. This time to see whether there’d be enough work for him to make a move to Europe. At the time he’d set his sights on the much more affordable city of Berlin when Flood derailed those plans.
A few months later, Dax had packed everything he owned into a shipping container and sent it to the other side of the planet. “It took a gruelling seven months to find the right place,” said Dax. “In the initial visit I checked out three or four properties with some different commercial agents. They said the market usually gets flooded just after Easter, because they take the break to pack up their offices and move out. Apparently it wasn’t going to be any trouble finding a £15/square foot lease. I arrived just after Easter and, lo and behold, there was nothing. That was the first of many disappointing road blocks.” 18 months later, he’s finally finished building Puzzle Factory in Tottenham, just five miles from
the centre of London. “I lucked out,” said Dax, who eventually found a spot at the intersection of three different train networks in the largest (180sqm) unit of three in a big warehouse. There are five studio rooms in Puzzle Factory, all spaced out in a row. Control Room 1 is Dax’s personal mixing and mastering suite. Next in line is Control Room 2, adjoining the main 5.5 x 4.7m live room, which connects to a smaller 3 x 2.7m live room, that backs onto Control Room 3. There’s also a machine room, kitchen/lounge, and a 6.5m by 3.5m workshop, to give Dax space to build his own gear. Jeff Hedback designed the structure and acoustics of the studio, while Dax undertook the job of building it to his spec and fitting out the
internal acoustic treatment. Control Room 1 is Dax’s own space. Dax is a die-hard in-the-box mixer, so his setup looks more like a mastering suite than a big studio mix room with no console and Duntech Princess monitors reaching to the ceiling. The focus was on designing great rooms, then creating an impeccable signal path all the way from the source to the monitors. The outboard is mostly preamps, and the only thing on his desk besides a keyboard and ball mouse is a single fader Alphatrack for balancing the mix. “The room was designed specifically for those speakers and you can tell,” said Dax. “The monitors are flat down to 38Hz and you can hear everything. We were nearing completion of Control Room
1 and it was time to do some measurements. Jeff designed it to the millimetre, and I built it to the millimetre. I ended up getting the Sonarworks calibration mic, set it up how Jeff asked, captured the measurements and sent them off. He came back and said, ‘I’m not totally sure what I’m seeing here. It’s possible there’s something wrong with the mic.’ So I repeated the process with another setup; calibration, capturing responses. He came back and said, ‘well the results are basically identical and incredible. You have no low frequency resonances, your left to right matching across the spectrum is amazing, and the low frequency matching from left to right is within 3.5dB.’ I was pretty excited to get that email.” AT 19
FEATURE
MIXER PROFILE
TRISTAN HOOGLAND When Sing Sing marched its gear South, resident engineer/producer Tristan Hoogland established a new position at Fall Back Studios. From his new custom-built mix room, he shares steps to enhance your indie synth pop mix. Story: Mark Davie
When the main Sing Sing studio in Cremorne shut down and Kaj and Jude squeezed the biggest and best pieces of its inventory into the smaller Sing Sing South, there wasn’t a whole lot of space left for engineer/producer Tristan Hoogland. He and Aaron Dobos had been sharing a makeshift production/mixing space there, and they figured it was the right time to look for a new place to work. “We were looking for months, and it was really hard to find anything around that was decent,” said Hoogland. Eventually they happened on to their dream scenario, Fall Back Studios, an upand-coming spot in Ascot Vale owned by Daniel Neidermeyer and built by he and his musician brother Stefan. Fall Back is a new co-share studio facility with custom-built studio pods dotting both floors. Each AT 20
pod is slightly different, reflecting the needs of each long-term tenant. For Dobos and Hoogland, that meant a dedicated mixing space, with enough room to record vocals and guitars. Acoustic consultants Marshall Day helped design the pods, which — with their angled noses — look like virtual reality starship cockpits dotted around the building. The arrangement at Fall Back is very rare, Dan was footing the bill, but allowed Hoogland and Dobos to specify exactly what they wanted from the build. Hoogland had been involved in a few studio builds up in Brisbane, and learnt a lot from the Queensland University of Technology’s Gasworks studio build, which paid dividends at Fall Back. “I was employed as a technician there and learnt the ropes by watching people,” said Hoogland. “When I came here, I helped Dan choose the right materials. All the structure and
logistics, Dan is across because he also owns Sideways Studios, which is a rehearsal space and small recording studio.” The arrangement between Dobos and Hoogland is pretty flexible. “We basically halve the month. It works out pretty well because most of my work is mixing, whereas Aaron does about 50-60% recording.” Though he still mixes the occasional indie band like a Fountaineer record that Matt Neighbour produced, Hoogland’s clientele has shifted more towards electronic music, specifically synth pop artists like Woodes, Golden Vessel, and Huntly. Hoogland says it’s a completely different experience to the over-the-shoulder preciousness of young rock musicians new to the mixing process. Most of Hoogland’s clients are happy to let him get on with the job. “Most of it comes in over email, I would
7
ART 7
STILL THE FIRST CHOICE OF AUDIO PROFESSIONALS THE WORLD OVER, RCF INTRODUCES THE ART 7 MK4 SERIES. THERE HAS NEVER BEEN A BETTER TIME TO HEAR THE DIFFERENCE FOR YOURSELF. PRODUCTION GRADE BUILD QUALITY. SEVERAL SYSTEM CONFIGURATIONS (UP TO 4” VC). PROPRIETARY FIR-PHASE TECHNOLOGY. FIVE YEAR AUSTRALIAN WARRANTY. MADE IN ITALY.
TO FIND YOUR NEAREST RCF PROFESSIONAL DEALER, VISIT: WWW.RCFAUDIO.COM.AU
AT 21
say 95% of the sessions I mix are unattended,” he said. “I have two clients who live around the corner from here, and they don’t even come down.” Hoogland puts it down to trust in the process, but also that most electronic musicians are comfortable with their listening environments at home: “Unlike bands, who might spend a lot of time in rehearsal rooms, and hearing live instruments in pre-production. For an electronic artist, the entire environment from writing to recording and mixing a song is coming out of the same speakers. They’re very used to that sound.” Hoogland is a fan of his new working mode. “It’s better when they don’t attend, because there’s freedom to move around between projects, and try things you wouldn’t otherwise try. “On average, it might take me an hour to make my way through a list of requests. When they’re in the room, it could take three to four hours because they have to listen to the song over and over to make sure the 2dB bump is exactly what they wanted.” When he’s behind closed doors, Hoogland isn’t afraid to roll his sleeves up and dig deep into a mix. He broke down four of the key points you should think about when mixing indie electronic music. Tip 1: Edit down the bottom end “Getting rid of a lot of bottom end is probably the biggest step in a lot of this music. Because indie electro is largely self-produced, you become more of an editor. A session may incorporate three or four bass lines, which the laws of physics make really difficult to fit on a typical speaker. I tend to start by editing the arrangement and minimising the overlap of lines. Some people can layer them really well, and it works, but more often than not the relationship with the kick drum doesn’t work. AT 22
A common one is to get both a subby and fuzzy bass. Oftentimes you can achieve a similar result by blending in an instance of Sountdoys Decapitator or Thermionic Culture Vulture. Some sessions will also have four or five kicks, and a lot of lo-fi phone recordings, so everything has a lot of irrelevant mess in the bottom end. The low end is really important to electronic music; you want it to hit really hard.” Tip 2: Get aggressive with processing to even out anomalies “Because a lot of artists record at home and don’t necessarily have great microphones. You have to get rid of a lot of anomalies in the recording, which means you have to get really aggressive with EQ. I also run things through outboard like the Retro StaLevel or UA 1176 compressor, and tape saturators, to help even out some of those anomalies.
Tip 3: Parallel distortion over compression “Mixing rock music, you use a lot of parallel compression, especially to make drums sound explosive. With a static drum sample in electronic music, that technique doesn’t quite give the same result. Parallel distortion really helps. A lot of the times tracks come in really clean, so distortion can inject a little bit of character back in. It also helps sounds cut on small speakers.” Tip 4: Make sure you can hear the low end “The Barefoot monitors really help me judge the low end, which is a big emphasis in that kind of music. It helps you get rid of a lot of the unwanted stuff you wouldn’t otherwise hear. I probably spend 80% of my time on the Yamaha NS10s, but the 20% on the Barefoots is really important.”
AT 23
FEATURE
AT 24
Parkway Drive took a ‘gamble’ by asking their live sound guy to produce their last record. The success of George Hadjichristou and his brother Dean’s meticulous, hard-hitting approach turned a gamble into a sure thing. Story: Paul Tingen Photos: Chady Awad & David Bichard
Three years ago Byron Bay metalcore band Parkway Drive took what they called “a gamble”; instead of continuing to work with big-name producers, they decided to record their fifth studio album, Ire, with two relatively unknown names. Well, one name, two people. Installed at the helm as the album’s producer George Hadjichristou, and his brother Dean took to the controls as the recording engineer. While the Hadjichristou name might not ring a bell for most, it has a longstanding affiliation with the Parkway Drive camp. George is the band’s live front-ofhouse engineer. “We thought it was more important having someone we are super comfortable with, knows our sound, and knows what works live,” commented drummer Benjamin Gordon, at the time. “Getting super-comfortable” is not typically a recipe for excellence, but the “gamble” paid off, and Ire became Parkway Drive’s most successful album at the time, reaching number one in Australia. Last year, when it came to making the band’s sixth studio album, working with the Hadjichristou brothers was no longer a gamble, but the obvious thing to do. Once again, the chemistry worked, as Reverence turned out to be even more successful. Reaching number one in Australia, but also the third spot in Germany and number 14 in the UK. Reverence is sonically adventurous, stepping out of the confines of metalcore to incorporate musical influences like folk and orchestral. George Hadjichristou didn’t only produce, but brought a collection of talents to the table like piano, string composition and arrangement. The album features some real strings, as well as synths and all manner of samples, atmospherics and other non-standard sounds in the metalcore genre. Going full loop, bringing George’s FOH perspective to the albummaking process has delivered enormous-sounding albums that not only cement the band’s ascendancy in Australia’s heavy music scene, but helped them climb to the top of many live festival bills around the world. HADJICHRISTOU’S ANALOGUE EDGE
Unfortunately, while George was eager to talk, he had just started the band’s European tour, and couldn’t find any space in the midst of his 24/7 schedule. It was up to Dean, the elder of the two,
to spill the beans for the Hadjichristou brothers. Firstly, that name. “It’s Greek!” explained Dean. “We were born and raised in Greece, and 24 years ago my parents moved to Canada, wanting to make a fresh start here. I was 14 at the time, and George 11. We have been musicians all our lives. As we grew up, we were always in bands, with me playing drums. With regards to audio engineering, I started in radio broadcasting, and then studied Music Recording Arts at Fanshawe College, in Ontario. That was at the beginning of this century, and we still had to work with two-inch tape and a full console for a year before we were allowed to touch Pro Tools! I also did a postgraduate program at the college, which focused on post-production for films and mastering.” Hadjichristou says his analogue education at Fanshawe gave him “an edge. Kids today are really spoilt working with non-linear DAWs and plug-ins, and gain structure not being as important as it was then. However, most of what I’ve learned has been the result of working in my studio on my own, every day, in the real world for the past 10 years. There’s that saying about having to learn the rules before you can break them, and that’s what I did. These days plug-ins have become really good and I’m mostly in the box, but I still occasionally miss analogue. There’s something magical about tweaking an EQ on a console, and compression is the one thing I think still doesn’t work as well in the box.” SURFING THE WAVE
Hadjichristou’s “real world” studio in Ottawa, Canada is called All Buttons In. It’s where Parkway Drive recorded Ire and Reverence, a million miles away from the surf of Byron Bay. Dean began the studio back in 2001, but today there’s a full Pro Tools system with a Control 24 control surface, JBL LS305 and PreSonus Sceptre S8 monitors, a small collection of outboard like a UA 1176N compressor and API 512c preamp, and quite a few microphones. The list of visiting artists is extremely varied and includes quite a bit of folk and country. “When you run a commercial studio you don’t choose the genres you’re working with,” remarks Hadjichristou. “I’ve worked on many country records, on my own, and with my brother. Recently, we work together less frequently, as he’s moved to Arizona, and is often on the road with Parkway Drive. Still, heavy bands gravitate towards recording here, and I love that!” To date, Parkway Drive is the biggest name that has come through the doors of All Buttons In. Hadjichristou avidly remembers the first time they arrived in the beginning of 2015. “The band knew that George and I work on records together, and after a South American tour they came here, to record a demo for a song called Vice Gripe, which became the lead single for Ire. We had such a great time that they came back to do the entire album, and subsequently, Reverence. This despite notching up record-breaking temperatures of -45°C while we were recording Ire! The Australians did not like that! Luckily, the weather was beautiful when we recorded Reverence.”
THE PROCESS OF MAKING THE NEW PARKWAY DRIVE ALBUM INVOLVED DEMOING ALL SONGS, NOT ONCE, BUT TWICE. While some artists prefer to avoid demo-itis by limiting their demos to Garageband sketches, apparently recording advanced demos during preproduction is quite common in metal. Hadjichristou explained the process: “Jeff Ling, the band’s lead guitarist, starts a lot of the music, and he records his ideas at a small studio in his house. Ben, the drummer, and Winston [McCall], the singer, will also get their ideas down at his place. Over a year their demos kept evolving and, when they were advanced enough, George and I flew out to Australia. This was in July 2017, and we spent a month at a studio called Byron Hive, in Byron Bay. It was a comfortable and easy place to meet up, where George would work with them on the final arrangements, while I recorded everything in Pro Tools. Plus the band could go surfing during time off!”
DRUMS FOR DAYS
By the end of the time at Byron Hive, everyone knew “exactly what we were going to do,” said Hadjichristou. “We then recorded the band in Canada, one member at a time. The singer doesn’t need to be there while we are getting snare drum tones, for example. We did two weeks in August recording drums and piano at the Warehouse Studios in Vancouver, which is Bryan Adams’ studio, and then spent the next 10 weeks at my studio, recording the rest of the material. The order we recorded things in was drums, bass, rhythm guitars, lead guitars, vocals, group vocals and then all the support things like strings, anything MIDI, and synths pads and effects. It was a revolving door of rock stars coming in and out of my place! It would be my brother and I and one band member at a time in the studio. Everyone trusted us to go with the game-plan and get the right tones, so there wasn’t a lot of communication with the band members who weren’t there. We only asked for approval from the others if something changed.” Recording one instrument at a time affords the Hadjichristou brothers a clarity of focus producers and engineers don’t typically get when recording an entire band at once. Dean began breaking down the detail that went into recording each instrument, starting with the drum sessions at The Warehouse. “We used the pre-production scratch tracks as backing for the drummer to play to in Vancouver,” he began. “We had already been experimenting with different recording techniques. On two of the songs we wanted to record the drum shells separate from the cymbals, and we also wanted a lot more room and space around the drums. WE RECORDED THE DRUMS IN STUDIO 2, WHICH HAS A NEVE 58X24X32 CONSOLE ORIGINALLY COMMISSIONED BY GEORGE MARTIN AND GEOFF EMERICK IN 1977 FOR AIR LONDON STUDIOS. It
has 31106 EQs inside, and it’s my favourite console to work on. It adds just the right amount of colour to drums, so all the mics went through that. “Because we had the time, we spent three days setting up, AB-ing every microphone and drum kit you can imagine! I ended up with an Audix D6 on the kick inside, and an RCA 77-DX on AT 25
Making a metal record is very similar to doing a hip-hop record, because both are all about phrasing and cadence and less about melody The Hadjichristou brothers (Dean - left, George - right) take a break while working at The Warehouse Studio 2.
the outside. Because I knew the sessions would go to an external mix engineer, I wanted to give him some different snare options. We taped together a Beyerdynamic M201, Shure SM57, and AKG C451 so the capsules were phase aligned. There were six toms, each with an AKG C414 condenser, and I also brought my own CAD M179 mics for the upper toms. The ride and hat mics were Neumann KM54s, and the overheads were Manley Golds — we tried the Telefunken ELA M 251, but didn’t like them. “The rooms was where the fun happened, with Coles 4038 mics going through UA 1176 compressors, and Neumann M149s as the far room mics going through an amazing Pye compressor. I also sent a mono hall mic through a Distressor. PLUS, WE SET UP A PA IN THE ROOM AND BLASTED THE KICK AND SNARE THROUGH A BIG SUB, WHICH BEEFED UP THE LOW END. The Warehouse Studio 2 live room sounds
very aggressive and bright, which can be amazing, but we wanted to steer it a bit more towards the sounds of the kick and the snare. While we were at The Warehouse, George also played piano parts on the Yamaha C7 piano there, which I recorded over the hammers with AKG C414s set to omni, and the same room mic we had used for the drums.”
ALL BUTTONS IN
After the sessions in the Warehouse were complete, the Hadjichristous decamped to Ottawa for the bass overdubs. “We used a Kemper amp simulator for both the bass and electric guitars, so there were AT 26
no mics involved. I recorded the bass DI via the A-Designs REDDI, which is a great red box with a tube inside, so we got a bit of character from that, and we used a UA 1176 on the way into Pro Tools. We split the DI signal to the Kemper for a cab sound, and that signal was split again to an analogue Sansamp for some distortion. I prefer analogue distortion! So I ended up with three bass tracks. I also AB-ed a Burl Audio B2 converter with my Apogee Rosetta 800, and for some songs we chose the Burl. Once we found a bass sound we were happy with, we stuck with that for the entire album. We did that with most sounds on the album. “We sent the Kemper line output into my API 512c mic pres to get some of that mid-range, and the lead guitars went through a Neve 1073 or a Tube-Tech CL1B, to give them a bit more glue. During demo-ing we’d created loads of guitar amp profiles in the Kemper, and also bought some profiles. We basically took four days sculpting the sounds of different amps on the Kemper and how to blend them together. I made a bunch of bounces of a guitar part of a certain song, with the different amp options unlabelled, so the band didn’t know what was what, and then we all listened on headphones to see which we liked best. Once we decided on a blend, that’s what we used at my studio. I GREW UP MIKING CABINETS, AND I STILL LOVE DOING THAT, BUT I HAVE TO SAY THESE KEMPERS SOUND PRETTY GOOD!
“Recording the vocals was fun, because it was back to my grassroots… analogue style. For most
of the record we used a Neumann U47 FET, because it can take Winston’s volume. The chain was U47 FET to a Neve 1073 mic pre, to a UA 1176LN compressor, then a CL1B, then the Burl converter into Pro Tools. On the previous record we spent many hours getting the right attack and release settings and gain structures, and I kind of replicated those for this record. We sometimes switched microphones, which obviously changed the gain structure, but the attack and release settings remained the same. The other microphones we tried were the Peluso 2247LE for what we called the ‘wizard vocals’; basically anything that wasn’t very loud, mostly spoken word. When we were doing vocal stacks for choir-ish sounds, I used all sorts of microphones — like the Avantone CK7 — to get different textures and keep them out of the same sonic space as Winston’s main vocal. “We recorded a small string section in my studio, which has worked pretty well in the past. I had to make the sound a bit bigger using digital reverb, which is pretty convincing. In a track like Shadow Boxing, we reinforced the strings with individual violin and cello parts, and some MIDI strings. I recorded the strings with the Neumann U47 FET on the cello, and Peluso CM6s on the violins, plus the Avantone CK7 as a stereo pair, and a Peluso 2247LE in the middle for a centre image. These mics went through Neve 1073 and API mic pres, and I only compressed the centre Peluso. The rest went into Pro Tools without compression or EQ.” During the entire recording process,
The brothers spent three days trying every mic combination, on all the available drums, to dial in the perfect tone. This is what they ended up with.
Hadjichristou had also been mixing. “It’s a tough balance as a recording engineer, because you always try to keep your system running as smoothly as possible while recording. At the same time, EVERYONE WANTS TO HAVE A CLEAR IDEA OF WHAT THE FINAL RESULT WILL SOUND LIKE AND WHEN THINGS SOUND MORE SLICK AND FINISHED IT INSPIRES EVERYONE. The
entire band came over to Ottawa when we recorded the final stages, to make sure they were happy with the additional layers we put on, like strings and keys, and also to judge the final rough mixes. “Then everybody left, leaving me with a million files to prepare, backup, comp, and send to the mixer, Josh Wilbur! I made sure I sent him the blends we had decided on, and printed any processing we wanted to hear. The recording session for Shadow Boxing would have been 150 tracks, but I reduced that to 65-odd. The string session alone would have been 30 tracks, which I mixed down to two. We’d recorded 20 channels of ‘war drums’ in Vancouver, and a mixer doesn’t want to deal with all that, so I also bounced those to stereo. I MIX OTHER PEOPLE’S SESSIONS, AND I HATE IT
WHEN EVERY SINGLE DECISION IS LEFT UP TO ME. I DON’T NEED 16 TRACKS OF ONE GUITAR PASS. DECIDE THE BLEND, AND JUST SEND ME TWO TRACKS!”
DIGITAL OVERLOAD
A few thousand kilometres south-east from Ottawa, in his home studio near Los Angeles, engineer, producer and mixer Josh Wilbur talked through his mix process for Reverence, which he says was the
easiest he’s ever done. “They sent me exactly what they wanted to hear, which made the entire mix process very smooth. It was an unusual way to work, though. Normally I have tons of communication with the producer, and sometimes the band, but I did not talk very much with them. I did one trial mix, they reacted to it the next day, which was the fastest response ever, saying, ‘your mix is on the record,’ and then I mixed the rest of the album using that first mix as a template. When I was finished they sent me very detailed and organised comments, which I incorporated, and that was it.” Wilbur is a studio professional of impressive pedigree. Born on the US East Coast, he played drums in bands as a teenager and obtained an associate’s degree in audio engineering from New England School of Communications. Following this he moved to New York, where he worked at Soundtrack studios assisting the legendary Andy Wallace, and also with hip-hop production team Full Force. For a few months he even worked with pop production icon Max Martin. This wideranging set of experiences shows in his credit list, which includes LCD Soundsystem, P!nk, Avril Lavigne, Steve Earl (Wilbur won an engineering Grammy Award for Earl’s Washington Square Serenade album), Leona Lewis and many others. However, a substantial part of Wilbur’s credits are in the metal arena, with names like Avenged Sevenfold, Gojira, Trivium, Megadeth, and Lamb of God (which netted him three Grammy nominations). “I like everything from pop to hip-
hop to R&B to metal,” remarks Wilbur. “Working with Full Force, Max Martin and Dr Luke, taught me a lot of about song writing and production, while I obviously learned a lot about engineering, mixing and production from Andy. If truth be told, making a metal record is very similar to doing a hip-hop record, because both are all about phrasing and cadence and less about melody.” Getting married and having kids pushed Wilbur to live in Los Angeles, where he has a studio in a building behind his house. It has a substantial amount of outboard, which he only uses for tracking. When he mixes he’s all in the box, using Pro Tools, Genelec 8351 and Yamaha NS10 monitors, and the Dangerous Music Monitor ST, and an Apollo I/O. Contrary to many engineers and mixers who claim that moving into the digital-only realm was plain sailing, Wilbur admits the changeover — which he made around 2010 — was “rough.” “The last full album I mixed on a desk was Lamb of God’s Wrath (2009), and when I went in the box I didn’t realise how much I had been relying on analogue circuitry,” he explained. “When you’re mixing on an SSL E- or G-series you always see the channel lights overloading. You’d boost things from there, and it’d sound great. When I went in the box I’d boost EQ on, say, the snare, and it’d hurt my ears, despite me making the same moves I was used to making on the SSL. I think it had a lot to do with gain staging, which I had to learn to control in the digital realm, and I also learned to mostly apply subtractive EQ, rather than the boosting I was used AT 27
to. I had to learn to focus on removing sounds I don’t want to hear, rather than change what’s there. That relates to a more general issue as a mixer, which is that you sometimes just have to accept there are certain sounds you may not like. Rather than try to change it, you need to leave it alone, and just balance it and make it fit.” WILBUR FORCE
Wilbur’s mixes for the Reverence album didn’t contain any sounds he had issues with. The ease of his mix process was not only due to well-organised sessions, but also because they sounded great. “When I start a mix, I first listen to the rough and with Parkway Drive they just sounded killer! The guitars in particular sounded fantastic, and at that point rule number one is not to ruin the production! They had done a great job tracking, and all I had to do was make sure you can hear everything and find other things to improve. Again, that was often a matter of using subtractive EQ to make space, so everything is more audible. “IN GENERAL, MIXING METAL IS A MATTER OF MID-RANGE MANAGEMENT, BECAUSE EVERYTHING LIVES THERE. I love deep bottom end, and have a reputation for really deep sounding records, but the focus in metal is on the mid-range and especially on the guitars. When people ask me how I approach the bass when mixing metal I say that I’m trying to get it to sound like the left hand of a piano. The low keys of a piano are my reference, and I try to get the same shimmer at the top, so it’ll cut through in the mix. “I also always start my mixes from scratch. The only template I have is a bunch of delays and reverbs and doubler aux tracks that I start with at the bottom of a session. It’s similar to the way I used to work on a desk, when I had my effect tracks on the right. I still run Pro Tools very similar to a console. I prefer to use sends for my reverb returns, and not to put them on the actual audio tracks. When I get sessions to mix there’s often a reverb and delay on every single vocal track, which seems unnecessary to me. I tend to take these off, then I can dip into my aux tracks to see what the vocals, and other tracks, need.” DRIVE TO THE FINISH
“When I received the Parkway Drive sessions I started with the guitars, which is unusual for me, because I generally start with the drums. I liked the guitar tones they had, so it made sense to use them as a foundation to work from. I applied some EQ to the guitar tracks, using the stock Avid EQ3 7-band, which is still my favourite EQ, as it doesn’t have a large graphic representation. You don’t want to EQ based on what your eyes see, which is the reason I switch off the graphical representation on the FabFilter Pro-Q2, my second go-to EQ. The guitars had the EQ3 and a Waves LA3A, because they make the guitars chop. It makes them hit harder on the entrances. “The Metric Halo Channel Strip is my go-to for the drums. It sounds more like an SSL than any plug-in modelled on an SSL! I used EQ and compression on all the individual tracks. I ALSO ADDED SOME DRUM SAMPLES, USING THE MASSEY DRT TO CREATE MIDI FROM THE AUDIO, AND THEN I USE THE STEVEN SLATE TRIGGER TO TRIGGER THE AUDIO. I do it that way AT 28
so I can audition sounds very quickly. Some guys use the same drum samples on every record, but I don’t work like that. The snare sample I added on the Parkway Drive album was never used on another record! I then print the samples. There’s a lot of reverb on the snare, by request, mostly from the Valhalla Room, one of my favourite plug-in reverbs, and also a little bit of Lexicon chamber. “I tend to use distortion on the bass to get it to cut through, though that is a delicate balance. I often receive a fuzz bass track, and that’s so difficult to work with. The type of distortion is key. I’LL OFTEN COPY THE CLEAN BASS TRACK AND SEND THE COPY THROUGH THE SANSAMP PLUG-IN, TO GET SOME GRIND. I THEN SUCK SOME MIDS OUT OF THE CLEAN TRACK, AND ONLY USE THE MIDS FROM THE DISTORTED TRACK.
Parkway actually sent me a distorted bass track, so I regularly used that. It comes in and out in the songs, depending on where it’s needed. I ride the volume on the distorted bass sound as well and will EQ both bass tracks differently for different parts of the song.” “On the vocals, I had the EQ3 7-band, FabFilter De-Esser, Waves 1176, and on some tracks the SoundToys Decapitator. I use the latter as a tape simulator, to do what the console used to do. I don’t crank the drive on it, I just use the stock setting with A pressed in, which I believe stands for analogue tape modelling. I do that quite often. It’s that or the UAD Studer A800 plug-in. Reverbs on the vocals were dependent on the song or the part, and would often be the UAD EMT plate. I also often used the SoundToys Echoboy for delay and the SoundToys MicroShift. These are all part of my standard settings, as is the Eventide H3000 Factory. I then find combinations of them that work.” Wilburn said mixing the Parkway Drive albums was, “a pleasure. I wish all mix projects were like that!” For Hadjichristou, meanwhile, it was “an extra bonus to have worked on a high-profile record, as it gives me a bit more credibility. George and I have worked together for 15 to 20 years, and I’ve been busy for 15 years in a business nobody is supposed to survive in, so it is not defining my work. Still, I’m very happy to have worked on these Parkway Drive albums, as they up my profile a bit, and bands I work with have more trust in me.”
Mix engineer Josh Wilbur: "In general, mixing metal is a matter of mid-range management, because everything lives there." (Photo: Julen Esteban-Pretel)
AT 29
FEATURE
Touring with Snoop at 18, breaking boundaries on Kendrick’s To Pimp A Butterfly, providing the groundwork for Flying Lotus’s albums. How Kamasi Washington and hip hop saved jazz. Story: Mark Davie
AT 30
Kamasi Washington doesn’t want to save jazz. He just wants you to give it a chance. In the same way that the perception of country music is saddled with layers of exaggerated twang, jazz has been pigeon-holed into the living rooms of audiophiles with squillion-dollar hi-fis, and quiet dinner-date background music. “Musicians in the past suffered from the word jazz having this bad reputation,” figured the saxophonist, who draws a lineage of jazz from early big band, through James Brown — “not the name, but the music, absolutely,” — to hip hop. “People putting words on jazz music and call it hip-hop. That’s been doing well,” he said. “Ask most R ’n’ B musicians and producers what they like, and they all love Herbie Hancock and Stevie Wonder. It’s jazz, the word, that’s been having a hard go of it, and it’s affected a lot of musicians who strongly associate themselves with that word. When people give the music a chance it’s such a freeing, spontaneous, soothing-to-your-soul music.” Washington himself doesn’t like all jazz: “If you get introduced to John Coltrane, you’re not automatically going to like Count Basie. Same thing with hip-hop; you hear Kendrick and love it, but you may not like a DJ Luke song. I don’t know that my music can all of a sudden make people like jazz, what I hope it does is make people more open-minded so when they come across something labelled as jazz, they’re no longer saying, ‘I’m not listening to that!’ That’s what happened to jazz for a long time.” Washington has lived on this bridge spanning jazz and hip hop for his entire career. “Hip hop was there as an option for me before jazz was,” said Washington, whose first tour after graduating high school was with Snoop. “I would go on tour with Snoop and come home and we’d have our own gig at 5th Street Dick’s, playing with people like Gerald Wilson,” said Washington. “Integration between jazz and hip hop was always there. I grew up in an area called Leimert Park, in Los Angeles. There was the World Stage, which was Billy Higgins’s jazz club, as well as 5th Street Dick’s, and around the corner was Project Blowed, where Freestyle Fellowship, Pharcyde and all those guys hung out. We would play Low End Theory which is a straightup DJ club. There’d be four DJs, then an eight-piece band on a little, itty-bitty stage that could barely fit the DJ. The drummers had to play with a two-piece drumset, and share a crash cymbal! The people loved it. Jazz can exist in the same places all other music does.” While the decades old confluence of jazz and hip hop was obvious to Washington, it really gained mainstream attention when Kendrick Lamar dropped the genre-hopping smash hit, To Pimp A Butterfly, to which Washington, and childhood friend and bassist Thundercat (Stephen Lee Bruner), contributed heavily. “Kendrick put it in the foreground, but people like Dre and Battlecat and cats like J. Dilla and Q-Tip, and A Tribe Called Quest, they were all using jazz,” said Washington. “Snoop’s whole band were jazz musicians, so they’ve always been in hip hop, just more behind the scenes. It just wasn’t really talked about, because
hip-hop was so new, the story was about the new music more so than the music it integrated.” While Washington knows the fight to reclaim jazz is an uphill battle, he’s comfortable carrying that weight on the shoulders of his dashiki. The opening track on his 17-song debut album, The Epic is literally called Change of the Guard, about a real dream he had of a young warrior rising to battle an established ageing guard. You can’t get more overt. RECORDING FIRST ALBUMS
Washington still lives in Inglewood — one of the suburbs on the doorstep of LAX — in his father’s house. Rickey Washington was a successful woodwind player in his own right, but chucked it in to break a cycle of fatherlessness and help raise Kamasi. He now travels on many of Kamasi’s tours and guests on the occasional song. Out the back of their house, Kamasi still has his garage studio where he first began recording. “Pops is not into technology — we had to teach him how to text message — but he was always cool to let me have my space there. He likes having us around all the time, and that energy. My studio was definitely lo-fi back in the day. I had like a Digi 002, Behringer mic pres and copies of Neumann mics, but it had a vibe. It was called The Shack. “I started off making beats with just an MPC and a Yamaha Motif keyboard. Then I got some advice from Tony Austin — my drummer and a really great engineer — and Terrace Martin, who had his own studio, about what I needed to record a full song. “I bought some mics, then I bought the computer, and a lightpipe extender so I had 16 in. We’d all be in the same room — piano and drums in there. The L-shaped room provided a natural divide, and I bought some individual dividers to help the quieter instruments. It was still pretty tight; you could definitely hear everything in every mic, but there was a vibe to it and it was cool because we could use it whenever we wanted.” THUNDERCAT’S AWAY
The biggest hit to come out of Washington’s backyard was Thundercat’s breakthrough album, The Golden Age of Apocalypse. He released it on Flying Lotus’s [Steven Ellison] Brainfeeder label, another good friend of the pair. It was the first of many albums to come out of their local tribe of musicians, some of whom are in Washington’s band, The Next Step. “We had been talking about it for years, that we had a sound,” said Washington. “When Thundercat came out with The Golden Age of Apocalypse it cemented the idea that we had something special. We would all be on different tours, but when we were in town we would make sure to come together and play. Even if it was to no-one, for no money. We didn’t care because we found this sound and approach that was different than what was going on in the rest of music. “We each had our own version of this sound. When Thundercat’s album did so well, we thought we should all buckle down and record, but we were all working! I made up my mind that I was going to
AT 31
It was a like a musical sweatshop with a rotating boss. You’d change project and the next person comes in full steam ahead!
Washington writes and conducts all his orchestral arrangements, recorded at Henson with renowned mixing engineer, Russell Elevado, in the foreground.
SWINGIN’ SWEATSHOP
CHARTING PROGRESS Washington finds it easy to chart his music out on paper, “creating it is the hard part,” he said. “I spend a lot of time just thinking. I can be sitting still at the piano, but I’m actually working; imagining the song. I’ve got a real fast car, all I need is the directions. If I don’t have the directions then I’ll usually sit and wait because otherwise you’re gonna drive the wrong way. “Some songs take a long time to figure out what they are, because I didn’t really know what the feel should be or what the chords mean. For two or three years, a song from The Epic existed on this little Post-It note with just ‘The Rhythm Changes’ written on it. I knew it in my head but that was the only representation of it. “One thing I learned from this last record, Heaven & Earth is that the album itself wants to be something too. There was one song we used to play a lot called The Conception. It was one of those songs we would play if we were having a bad show. It would always be good. “We were in the studio and thought it would be a good song for the album, but somehow we could not play it right. We tried it two or three times and it just wasn’t happening. Then I don’t know why but I just decided to play this other song called One of One. I wrote it maybe 10 or 12 years ago. It’s a really rhythmically difficult song and the few times we’d played it at gigs it was a train wreck. That was the first time we ever recorded it and it sounded great.”
record, which was when Lotus asked if I wanted to make a record for Brainfeeder.” Washington knew he wanted to record his own version of jazz, but his studio was “pretty limited. When you’re working in your own studio, you’re the engineer. Even when you hire an engineer, when something goes wrong and breaks, everyone is eventually going to look at you to fix it. It’s not the most creatively conducive environment to record. That’s when we decided to go to a different studio.” He hatched a plan. Studio day rates can be fairly cost prohibitive for an independent musician looking to record an album, but they will generally do a deal on a longer term. Washington figured if he could get a few of the crew together that also AT 32
wanted to record their own album, they could split the cost of a month’s hire and it would work out cheaper for everyone. “If all eight of us pitched in and bought the days we needed for all of us, we’d get it at a much cheaper price than if we did it individually,” said Washington. “Plus, we’d get the benefit of being there with each other.” They found a studio, King Size Soundlabs in Los Angeles (home of Rob Schnapf ’s MANT Studios), which had an “old Neve board, a pretty good mic cabinet, and some great instruments like a B3 organ, some cool synths, old Rhodes and Clavs. It was kind of ’70s and real old school. It was definitely not known for jazz, more like indie rock.”
They devised a schedule, sometimes setting aside a complete day for one project, other times switching between a couple in a single day. “Each of us got maybe three full days, and everyone was supposed to be there the whole time every day,” said Washington. “It was a like a musical sweatshop with a rotating boss. You’d change project and the next person comes in full steam ahead! ‘Alright, let’s do it! Here we go!’” Everyone approached their session differently, according to the sound they were trying to capture. Some recorded live, some overdubbed, some completely improvised, others wrote out charts. Washington goes both ways; charting out the tunes, with an ‘anything goes’ view of the actual recording. “My songs are unorthodox so I have to explain what’s going on and build a chart,” he said. “I was trying to think of the players and what I could get out of them for each song. If I had a full day I would pick four songs, if it was a couple of hours I would pick one or two songs. I’d give it to the musicians and sit with them individually, and go over the kinds of things they could do and the kinds of places they could go. “Then when we record, everyone knows I’m wide open. At the end you can do whatever you want to do with my music. If it’s cool, it’s cool. I’ll just go with the flow. If a song that was supposed to be a ballad starts getting played fast, Imma just go with it. Other guys would come in with an idea or rhythm and just jam it out. Music takes these weird twists and turns and my thought process was to record as many songs as I could and then try to find the album within that.” Because he had everything written out,
THE POWER BEHIND YOUR PERFORMANCE F1 Flexible Array loudspeakers. Inspired by arena audio technology. Touring line arrays, with their flexibility and coverage control, were the inspiration for the Bose F1 Flexible Array loudspeaker system. Small in size but not performance, the F1 Model 812 allows you to control your coverage in nearly any space. Four flexible array options shape your sound, ensuring every seat in the house enjoys an optimum listening experience. Add to that 1000 W of full-range power plus the portable, high performance F1 Sub and the result is a system with unmatched clarity, range and punch only Bose can deliver. LEARN MORE AT BOSE.COM/F1. AUS 1300 368 436 | sales_australia@bose.com NZ 0800 705 500 | sales_newzealand@bose.com AT 33
Vox Recording Studios is Washington's dream studio. From the custom Deledio console built with a collection of vintage Universal Audio and API modules, to the floor-to-ceiling racks of Ampex tube preamps. (below) The instrument collection at Vox is staggering. These keyboard racks barely scratch the surface of what's on offer, and all those drawers are full of pedals.
WASHINGTON’S SAX CHAIN Washington: “An RCA 44 ribbon mic going into an Ampex, between the bell and neck, about six to eight inches away. That was my signal chain for this last record, and it’s been the best mic to really capture my sound.”
Washington managed to get the most songs recorded of anyone, about 45 in all. Out of that he picked 17, which became his first album The Epic. It would be natural to assume that with so many changeovers, they’d stick with one recording setup that adequately served everyone. They weren’t going for adequate though. Each artist was trying to execute their specific vision, so the setup would be altered to match. To handle the changeovers, they went down to the Hollywood Musicians Institute and grabbed a bunch of interns to help wrangle leads, instruments and mics. “Tony Austin was running the sessions,” said Washington. “Everyone’s approach to the recording setup was different — in one scenario the bass might go through the Ampex, in another scenario I get it — so we had the interns moving gear around and re-patching every session. My approach was to have everyone playing at the same time, so my setup of the room was chock full. Only the orchestra and choir would be overdubbed later on, whereas others overdubbed everything.” BIGGER THAN A CALIFORNIA KING
King Size Soundlabs is an amazing studio, but for his Harmony of Difference EP, Washington found a new favourite joint, Vox Recording Studios. Washington described it as “like King Size Sound on steroids.” Vox was the famed Electrovox studios, where the Wrecking Crew stationed themselves for a time, and jazz luminaries like Charlie Parker and Dizzy Gillespie recorded. It went out of business right around the time composer and guitarist Woody Jackson came into a pot of gold from scoring Grand Theft Auto V. “He used to work at a store called Black Market Music which was the best analogue recording equipment store in L.A,” recalled Washington. “He made a bunch of money scoring Grand Theft Auto and basically bought out Black Market Music.” AT 34
There’s an entire storage area full of vintage equipment, with racks of synths — “every Moog and two vinyl-based Mellotrons, which I’d never seen” [Jackson used to be a Mellotron technician – Ed] — rows of amps, and walls of drums and guitars. It’s a musician’s paradise, “like a kid in a candy store,” said Washington. On the engineering side, there’s an incredible array of analogue gear. At its core is a custom Deledio Console, built for Wally Heider, that has 24 channels with Universal Audio 1108 preamps, Universal Audio EQs, and API EQs on each. It was used on the Beach Boys’ Smile sessions. There’s also a Neve 5442 suitcase, and Neumann 18x4 console from Sunset Sound lying around. On the outboard side are all variety of Urei compressors, two Neve 2254e compressors, and a Fairchild 670. Pultec, Lang and Universal Audio EQs are also in the racks, flanked by a stack of Ampex 351 preamps and tape machines. Washington has a particular affinity for old Ampex tape machines, and not having to choose which channels to send through the old tape was a huge selling point for him: “Most studios have one or two. He has like 30 of them.” He then recorded the strings at Henson, which “doesn’t have some of the custom stuff but the room is so beautiful,” said Washington. “Then I got infamous mixing engineer, Russell Elevado [D’Angelo’s mixing engineer – Ed]. He shipped a bunch of his gear over from New York to Henson, and just brought a whole other level to it.” MODERN SEPARATION
For Washington, going to the trouble of finding a place stocked with vintage gear is not about emulating the hey-day of jazz. “It’s beyond jazz. The heyday of recorded music was the time just before we went digital,” said Washington. “There’s a warmth and a natural feeling that happens with
analogue that you seem to lose as you use more digital components. On the flipside, there’s a clarity you get with digital that’s cool. I try to mix it up. To me, it’s all about what sounds the best. I’m not too philosophical with it. Even though we recorded to Pro Tools, we always mixed down to 1/2-inch tape. Russ, who mixed the last two records [including Washington’s latest Heaven & Earth], didn’t use any plug-ins, it was all analogue. Old mics and old pres with natural reverb. We mixed it at Henson and they had six different natural plate reverbs and four echo chambers. We were keeping it out of the box as much as possible.” Kamasi doesn’t really have a template for the kind of music they’re recording, so it’s all about what sounds good to his ears. “A lot of the records I use as references aren’t jazz; like Beatles records, and King Crimson records,” said Washington. “TO ME IT’S MORE ABOUT FINDING THE CHARACTER OF THE SONG THAN TRYING TO MAKE IT SOUND LIKE A CLASSIC JAZZ RECORD. I try to take it from scratch because there’s no real good reference for a lot of stuff we’re doing. There’s no record that also has two drummers and two percussionists and upright bass, a whole load of keyboards, a grand piano and a whole orchestra — strings, winds, brass and a big choir. All playing intensely. There are jazz records that have orchestras, but it’s usually alongside a small jazz band playing softly. “When I look at the pictures of how old records were made, by comparison we play with much more separation. We had each of the drummers in their own room. The horns, bass and piano were all in the same room, but it was big, so they were spread out. We used a lot of pedals and effects, we were re-amping things. We weren’t on a quest to emulate the sound of anything. It was more like, ‘What’s the sound of us?’”
AT 35
TUTORIAL
Techno pioneer Dave Clarke explains his secrets to perfect bass. Tutorial: Dave Clarke
From the beginning of my career I had to concern myself with bass. When vinyl was the predominant format for DJs, we learnt how to control the low frequencies. We had to. If you went too hard, you’d risk the needle skipping. (Incidentally, that’s why the bass drum sounds comparatively thin on those vinyl pressings from the ’90s.) Now bass is a fascination for me. Making sense of the low end is about how best to deliver the energy in your mix. I’ll spend the time using EQ to carve space for the bass drum to come through with its second, third and fifth harmonics. I’ve also spent time refining my monitoring so I can adequately understand and appreciate bass in a way that’s clear and reliable. Once you’ve got the fundamentals right then eventually dealing with bass becomes an intuition. EQ SURGERY
Artist: Dave Clarke Album: The Desecration of Desire AT 36
The bass drum and the bass line are always the challenge in electronic dance music but fortunately our DAW’s EQ is now so surgical, it’s easy to do the work — everyone has the tools at their disposal. Years ago you might only have a low or high tone control and it really was about taking the edge off what was most objectionable. Now you can precisely carve out the space you need, such is the surgical nature of the EQ. Which frequencies? That’s easy as well: every DAW has a real-time frequency analyser; it’s easy to see where the energy is. I like the bx_digital V3 plug-in as a surgical tool. I have a variety of different EQ tools that I like to use for a quick shift to reveal what’s going on. Sometimes I’ll use a Tone Tilt EQ — where the EQ
simultaneously boosts the HF and attenuates the LF content. That can be a very good way of finding out why your bass line and bass drum aren’t gelling — shift it away, see what it isn’t working with; make some tweaks; then shift it back. COMPRESSION
I’ll generally start off any track with the bass drum and/or bass line or maybe a drum loop. I work out a drum loop and then work around that and then add body to it. I have my favourite hardware compression for different types of bass lines and sometimes I sum a couple of bass lines together so I can compress them as one. I don’t overdo the compression. I’m normally in the 2 to 3:1 ratio range, occasionally 4:1. I’ll use my ELI Distressor on the drum bus, normally at a ratio of 4 or 5:1. I don’t nuke my drum bus, and in so doing I get the interplay between the compression of the bass drum with the compression of the bass line. They will go through different compressors and then they get fed into the master mix through a Cranesong STC compressor — very minimal, just enough to rein the levels in a little bit. You can, of course, do something similar with plug-in compressors.
It doesn’t matter if you’re setting up your first remote live stream session, DJing your hundredth party, or mastering your twelfth studio album—the Studioseries has got you covered. All of these interfaces boast as much I/O as we could
Studio 24: 2x2 USB-C audio interface
squeeze into their sturdy chassis. But robust hardware is only half the package. The included Studio One® Artist
Studio 26: 2x4 USB 2.0 audio interface
DAW lets you record and mix with unlimited tracks and effects in an intuitive UI that stays out of your way. And the Studio Magic Plug-in Suite expands your sonic palette with powerful VSTs from the best developers in the business.
Studio 68: 6x6 USB 2.0 audio interface
Meet the family.
Regardless of what your musical passion is or where you make it happen—we’ve got the right interface for both your workflow and workspace. Learn more at www.presonus.com.
STUDIO-SERIES RECORDING SYSTEMS
Studio 1810: 18x8 USB 2.0 audio interface
Studio 1824: 18x18 USB 2.0 audio interface
192 kHz, USB 2.0 Recording Systems Studio 192: 26x32 USB 3.0 audio interface and command center
ALL STUDIO-SERIES USB INTERFACES INCLUDE
Ph: 03 8373 4817 www.linkaudio.com.au
OVER $400 VALUE INCLUDED FREE AT 37
EQ Surgery: I like the bx_digital V3 plug-in as a surgical EQ tool to precisely carve out frequency space for my kick drum.
Over Easy: The ELI Distressor is one of my favourite drum bus compressors. I don’t nuke my drum bus, and in so doing I get the interplay between the compression of the bass drum with the compression of the bass line. A ratio of around 4:1 ought to do it.
I like working with the compression of the kick drum and bass lines while in that workshopping stage. If they’re working together cohesively then getting everything else to work with them becomes easier. I’ve learnt a lot from listening to reggae. The bass is the star, it’s where the energy is. MONITORING
My monitoring setup plays a big role in getting my bass right. My ATC SCM50s handle bass so well. I’m not using a sub in my studio, it’s just coming out of the three-ways. I’ll then flip to my smaller Neat Acoustics monitors to see if the mix is sounding okay or if any distortion is being introduced. I also use my trusty DK Audio meter for secondary visual feedback so I can see what it’s looking like. As a final ‘translation’ check, I have a corner in my studio which isn’t treated, where I can sit and feel what it would be like in a corner of a nightclub. Confession time: I do crank it. I know I shouldn’t. I often go into the studio with the best of intentions: ‘yes I’ll keep it calm today, I’ll keep it quiet.’ But, no, I turn it up because I crave the urgency. AT 38
That said, I’m only pushing levels a little too hard in the final mix stages. Prior to that, when I’m writing and arranging, I’ll happily keep things at a more sensible level. The problem with monitoring at too high a level is your ears will experience a ‘squashed ear’ compression. In other words, after extended exposure you’re not actually hearing the mix properly and any decisions you make will be compromised. BASS-APELLA ANYONE?
I spend a lot of time concentrating on bass energy in my productions. It doesn’t always work out. Sometimes it sounds really horrible. Sometimes I have a really beautiful bass line but I can’t get it to sit with a bass drum and then you just almost wish you were doing a ‘bass-apella’ version of the track. Bass is fascinating and sometimes infuriating but it’s always so crucial. Dave Clarke’s first release was in 1990. His most recent album, The Desecration of Desire, was released last year, produced from his boat-borne studio in Amsterdam. And, yes, it has perfect bass.
DAVE CLARKE’S MONITORING SETUP Dave Clarke: My ATC SCM50s, the Towersonic speaker stands, the Cranesong Avocet with the DK Audio metering for my monitoring is just so sorted now, that it’s totally right. I don’t second guess it. The Bryston amp… again it’s all that boring stuff you do once and do properly and it keeps repaying you in spades. And the cables. I know I shouldn’t talk about cables but they’re sexy… they have a tonality as well. Some cables are silver, some are copper and I use those wisely depending on the application.
D E S I G N
M E E T S
I N T U I T I O N
SAY GOODBYE TO YOUR BULKY ANALOGUE MULTICORE! The TF Tour Pack offers a total of 32 mic inputs and includes a plug & play Dante stagebox solution with 16 inputs and 8 returns.
T O U R P A C K
+ N Y6 4 - D D a nte Ca r d
+ TIO16 0 8 - D S t ag ebox TF1 Mi xe r
4999
TF TOUR $ PACK
RRP* FOR A LIMITED TIME ONLY — OFFER ENDS 31 DEC 2018
TERMS & CONDITIONS This promotional offer is part of a national sales program conducted by the distributor, Yamaha Music Australia Pty Ltd and is valid from 1 November 2018 at participating TF Series dealers. The promotion will end on 31 December 2018. Final purchase price is determined by the participating dealer. *The prices set out in this advertisement are recommended retail prices (RRP) only and there is no obligation for Yamaha dealers to comply with this recommendation.
FIND A DEALER au.yamaha.com
EVENTS AND PROMOTIONS yamahabackstage.com.au
LIKE US ON FACEBOOK facebook.com/yamahabackstagepass
FOLLOW US ON INSTAGRAM instagram.com/yamahabackstage
WATCH VIDEOS
youtube.com/yamahaaustralia AT 39
TUTORIAL
VIEW FROM THE BENCH
Meters are supposed to be the arbiters of truth, so why isn’t there just one standard? Tutorial: Andy Szikla
It’s 1861, and James Clerk Maxwell (whose name now describes magnetic flux) and his chums take a break from zapping frogs’ legs, influencing Einstein, and shooting lightning between big Frankenstein Balls, to sit themselves down in front of the ancestor of all audio level meters — a galvanometer. The British Association for the Advancement of Science has deputised them to come up with the first proper set of electrical units. Among his experiments, Maxwell measures grams of magnetic force produced by different quantities-per-second of electricity flowing through his AT 40
galvanometer — exactly the same forces which cause deflection of the needle in a VU meter. Maxwell’s definitions, published in 1863, are the roots from which more or less all of today’s electrical units stem, though not without occasionally branching off. Before long, Maxwell’s unit of resistance is thought too small, so it gets multiplied by a billion, and re-branded as the Ohm. Likewise, electromotive force is multiplied by one hundred million, and called a Volt. 20 years later the French not only stopped the British making the Volt and the Ohm 10 times bigger again, but also changed the name for current
60
20 0
–
40
DC FULL-WAVE RECTIFIER
3,600Ω AC AUDIO
80
VU
1
100 %
2 3+
20
3 2 1 0
7 5
10
CAPACITOR
+ –
DC CURRENT
N MAGNET
S RETURN SPRING
GALVANOMETER-TYPE MOVING COIL VU METER Galvanometer-Type Moving Coil VU Meter
original and expensive ones today, the empties meterwhich windings about the AC audio is convertedAC to aaudio DC level by the fullis converted to a DC level by thethrough full-wavethe rectifier, thenatcharges the capacitor at adevices, rate governed by capacitor sizethe andcapacitor circuit resistance. In effect, is the charge on resistor this capacitor that the needleresistor winds up When is following. external and everything else is built into same rate.itWith the 3600Ω illustrated, wave rectifier, which then charges the incoming voltage decreases, the capacitor empties through the meter windings at about the same rate. With the 3,600 the meter housing. Galvanometers are also used a steady 1kHz sine wave at +4dBu (1.228V RMS) at a rate governed by capacitor size and circuit ohm resistor illustrated, a steady 1kHz sine wave at +4dBu (1.228vrms) should produce deflection to 0vu in 0.3 seconds. PPM and gain reduction should produce deflection to 0vu in 0.3 seconds. resistance. In effect, itByis using the charge on this different resistors the sensitivity may be changed to correlate higher levels to 0vu. for In the original devices, and meters, employing expensive ones today, the resistor is and everything elsethe is built into the may meterbe housing. Galvanometers are also to further vary their different input electronics Byexternal using different resistors sensitivity capacitor that the needle ends up following. When used for PPM and Gain-Reduction meters, employing different input electronics to further vary their alignment and ballisalignment and ballistics. changed to correlate higher levels to 0vu. In the the incoming voltage decreases, the capacitor tics.
from the Weber to the Ampere. Sacré bleu! So it carried on for the next 100 years, with a network of tweaks and inconsistencies that can leave amateur historians out on a limb. The truth is that while electricity is a force of nature, electronics is not. It is an industrial idea conceived to exploit those forces, by people whose job it is to make up new stuff as they go along. Pro audio is a side-branch of the electronics industry, and audio level meters are one of its many offshoots. Meters are a great metaphor for the development of electronics; they accommodate multiple interpretations and often deflect towards change despite wearing a facade of seemingly absolute values governed by a permanent order. In reality, that order has re-defined audio level meters many times — not only in terms of how they work, but also what information when you look at them. In essence, audio level meters are visual indicators used to assess the amplitude of a signal, and to align signals passing between equipment. They provide information for an audio technician, but the exact nature of that information has never been completely agreed upon. HIGH VOLUME DISTRIBUTION
Prior to 1920, audio level control was no more sophisticated than turning up the volume until you heard distortion, then backing it off a bit. This worked as long as the ins and outs weren’t too complicated; a single source into one amplifier. On Armistice Day in 1921, a ceremony for the burial of the Unknown Soldier in Arlington Virginia was heard simultaneously on public address systems at Arlington, New York, and San Francisco, with high-bandwidth (16Hz~5kHz) telephone lines used as the distribution medium. Since too much level
might cause overloading of the phone network amplifiers or ‘repeaters’, and the technician at Arlington couldn’t hear the P.A. in the other cities, the designers concluded that a separate means of input monitoring was required. Their solution was to create a device called a ‘Volume Indicator’, which consisted of a valve diode working into a DC milliamp meter. The amplification of the valve was adjustable so users could calibrate the needle’s deflection. Thereafter, an operator could keep tabs on audio levels without having to hear the result. Moving needle meters quickly caught on, especially in helping prevent over-modulation of radio signals, and saturation of telephone repeater amplifiers, where headroom clipping could infect adjacent wires. The problem was: everyone had their own version! There were different schemes measuring peak or RMS voltage, with slow, medium or fast pointer speeds, system impedances based on either 500 or 600Ω, and reference levels set to 1, 6, 10, 12.5 or 50mW. Scales might indicate decibels from –20 to +6, or –10 to +3, or percentage saturation to 60%, or both, with zero usually calibrated at the mid deflection point. Confusion reigned, and if you had to connect one brand of equipment to another, good luck. IN LOUD AGREEMENT
Common sense collided with history in 1940, when some guys from NBC, CBS, and Bell Telephone got together to agree on a new standard volume indicator and reference level, and what they invented was the VU meter as we know it today. Intended to display comparative loudness or power in ‘volume units’, their aim was to create a meter that would indicate identical levels for a range of dissimilar sources which an average listener would consider to
be of equal loudness. They assembled focus groups and played them male speech, female speech, piano, brass band, dance orchestra, and violin recordings, asking them to pick the level for each one where the sound system appeared to be on the threshold of distortion. Since all of these sources contain different ratios of peak to overall RMS power, a peak-reading scheme was deemed undesirable. RMS, or Root Mean Squared, is what you get when you take an undulating AC waveform and squish it into a flat block of equivalent density. Imagine the waves at the beach are peaking all over the place while you’re trying to work out the average depth of the tide. If you had a football-field sized piece of glass, you could squish the waves down until the sea is flat and then easily measure the volume of water. RMS tells you the power density under each wave, and the VU guys saw volume in a similar way. Their new meter would measure average power density against time. They chose 0.3 seconds as their preferred sample depth, rigging an internal capacitor to take that long to charge or discharge — a process governing each full-scale deflection and recovery. The slow timing was chosen partly because it would make a moving needle comfortable to read. The VU meter was calibrated so that 0vu — aligning with ~71% deflection — would correspond to a level produced by 1mW of power in a 600Ω system. It was intended that its sensitivity might, via the addition of an external resistor, be easily adjusted to any standard reference level, say +4 or +8vu, or to indicate 100% modulation of a broadcast signal at whatever level that happens. Both uses were accommodated on the VU scale, marked in dB from –20 to +3vu, with the 100% mark aligned with 0vu. AT 41
dBFS 0
+24 +22
3
+20 +18 +16 +14 +12
+dB 12
7
8
6
4
5
TEST
4
+dB 12
+10 +8
+dB 5
6
+6 +4
0
6
3
9
6
12
9
15
12
18
15
+2 0
dBFS 0
5
TEST
18 20
–2 4
3
8
2
12
1
–4
–8
25
–10 –12 –14 –16
–20
35
18 –dB
EBU PPM
5 7
30
10
35
20
–dB
–18
dBu Level In/Out
30
12
0 3
25
10 6
–6
20
+dB 3
BBC PPM
NORDIC PPM
–dB
DIN PPM
–dBFS
EBU DIGITAL
40
20 –dB
–dBFS
SMPTE DIGITAL
COMPARATIVE METER LEVELS These are examples of the most common co-existent PPM scales, with VU thrown in for comparison (shown here in its usual alignment where 0vu is equal to professional line level +4dBu). The dBu scale on the left shows the amplitude of the analogue signal you would expect to see at any input or output being monitored, relative to the levels expressed by each meter.
+4dBu VU
Comparative Meter Levels Diagram shows examples of the most common co-existent PPM scales, with VU thrown in for comparison (shown here in its usual alignment where 0vu is equal to professional line level +4dBu). The dBu scale on the left shows the amplitude of analog signal you would expect to see at any input or output being monitored, relative to the levels expressed by each meter.
STANDARD ISSUE
The 0vu standard was overwhelmingly accepted in all corners of industry, and later referred to as 0dBm (dB milliwatts). A 1mW sine wave into a 600Ω load produces a level of 0.775vrms, and today we disregard the load and just call the voltage level 0dBu. The ‘u’ originally meant ‘unloaded’ (ie. we don’t care what the load is, only the voltage level) but that’s all so long ago, now we simply regard dBu as dB ‘units’. Some other people decided they didn’t like a zero reference of 0.7745966692… volts RMS, and upped it to 1 Volt even, calling it 0dBV. Today our domestic hi-fi gear uses –10dBV as its 0vu reference point (around –8dBu), whereas most professional systems use +4dBu, and some gadgets give you the choice of both. SCALING THE EUROPEAN PEAKS
By 1950, the dBm was a smash hit world-wide, but the VU meter was not so successful outside the USA. Those pesky Europeans were less interested in subjective comparisons of sound sources, and more interested in knowing exactly how signals were being handled by actual equipment. In Germany, Scandinavia, and England, engineers had been separately developing peak level meters with the principal aim of monitoring broadcast modulation. Nobody saw the benefit in using an RMS-type device to do this monitoring, since it is the program peaks which cause over-modulation. Engineers at the BBC were even unkind enough to refer to the AT 42
VU meter as the ‘Virtually Useless’ meter. Peak-reading meters were around before VU, but to this day there has never been a single standard scale, timing or reference level to which all Peak Program Meters (PPM) abide. However, they do all calibrate to so many dBu, and to the naked eye all appear to sort of behave the same way-ish, with fast attack times (to catch the peaks) and usually a slow decay. In the days of analogue valves, overloading produced distortion that was not terribly harsh, and the feeling was if it lasted less than a few milliseconds nobody would notice. A milliamp meter could deflect to full scale in 1mS, so it became standard practice to slow down the attack slightly to remove these aurally meaningless peaks — the Brits chose 4mS and the Germans 10mS. With today’s digital equipment, hitting full-scale makes a loud bang, so quicker attack times can be useful. PPM decay might last anything from 1.5 (EBU) to 2.5 (BBC) seconds, mainly to make things easy on the eye. In time, it became fashionable to include a peak-hold of between one and three seconds, which is helpful if you have to look away momentarily, and don’t want to miss what happened. The front panel scales and alignments, however, are largely dreamt up. The BBC used a scale which was simply numbered one to seven, where four corresponded to 0dBu, and six to their maximum permitted signal of +8dBu. The EBU scale was marked in dB from –12 to +12dBu, with 0dBu marked as ‘Test’. There were many others — Diagram 2 provides a taste.
Engineers at the BBC were even unkind enough to refer to the VU meter as the ‘Virtually Useless’ meter
PUNISHING VU LEVELS Max Albiston, Chief Engineer at Channel 7 in the 1980s, told me about the days when TV commercials started appearing with heavily compressed audio tracks, resulting in viewer complaints that some sounded much louder than the shows they were interrupting. Max and his colleagues devised an ingenious remedy. They observed that compressed commercials caused a VU meter to peak many more times per minute than uncompressed ones, and by counting the peaks they could index the main offenders and code them for routing through an audio expander. This worked splendidly for the viewers and was applauded by management, but sponsor complaints quickly trumped the viewer ones and after only two weeks the scheme was axed!
FREQUENCY FOUND™ Audix Performance Series Wireless Audix 60 Series wireless offers 64 MHz of tunable bandwidth, true diversity receivers and an impressive operating range of 450 feet at a price point that is unmatched in the market today. Choose R61 single channel or R62 dual channel systems, available in multiple configurations including handheld, lavalier, headworn and instrument microphones. Learn more at audixusa.com.
Over 30 Years in Business To find your nearest Integrator/Reseller, please visit
www.pavt.com.au and click on ‘Where To Buy’
Production Audio Video Technology Pty Ltd 4/621 Whitehorse Road, Mitcham 3132, Victoria PH: 03 9264 8000 sales@pavt.com.au NEW ZEALAND OFFICE PH: +64 21 410 050 tim@productionaudio.co.nz AT 43 www.audixusa.com | 503.682.6933
DC AUDIO DC AUDIO LEVEL IN LEVEL +Vref IN
DC AUDIO DC AUDIO LEVEL IN LEVEL IN
+Vref
etc.
COMPARATORS COMPARATORS VDC #7
VDC #3
VDC #2
VDC #1
0VDC
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
VDC #6
VDC #5 VOLTAGE DIVIDER VDC #4
Ref
Sig
VDC #7
VDC #6
VOLTAGE DIVIDER
Ref
VDC #5
VDC #4
VDC #3
VDC #2
VDC #1
ADC DIGITAL ADC DIGITAL LEVEL OUT LEVEL OUT etc.
LED 7
LED 7
VDC #7
LED 6
LED 6
VDC #6
LED 5
LED 5
VDC #5
LED 4
LED 4
VDC #4
LED 3
LED 3
VDC #3
LED 2
LED 2
VDC #2
LED 1
LED 1
VDC #1
0VDC
0VDC
etc.
etc. etc.
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
Ref
Ref
Sig
Sig
VDC #7
VDC #6
VDC #5
VDC #4
VDC #3
VDC #2
VDC #1
0VDC ADC
7
6
5
4
3
2
1
etc.
7
6
5
4
3
2
1
ADC
LED and Digital LED and Meters Digital Meters am Theon diagram the leftLED on shows theAND left old-school shows old-school LEDMETERS metering. LED metering. At the input, At the audio input, hasaudio already has been already rectified beentorectified DC andtohad DCballistics and hadapplied, ballisticsper ap DIGITAL voltage 1. Its DC level voltage is fanned level is identically out identically toLED the Signal toAtthe reference Signal pins on a stack pins of aop-amps stack ofconfigured op-amps as comparators, as10comparators, who divider alreadyon built-in. The LM3915 by National configured Semiconductor has LEDwhose output The diagram on theout leftfanned shows old-school metering. the input, audioreference withto 3dB spacing between each, and pin. was forWhen many years considered has alreadypresents been rectifieddifferent to DC and had ballistics applied, perhapsof likeathe or e voltage low. The divider voltage divider presents fractions different of fractions a fixed reference fixed outputs, reference voltage voltage each Ref to pin. each When Ref the Signal the voltage Signal is voltage lower than is lo the industry standard. VU meter in Diagram 1. Its DC voltage level is fanned out identically to the exceeds DC #1, the VDC first #1, comparator the first comparator will change will itschange output from its output low to from high, low turning to high, onturning LED 1.on If itLED goes 1.higher If it goes than higher VDC than #2, both VDCLED #2, bot 1a Signal reference pins on a stack of op-amps configured as comparators, The built-in. diagram on the right looks inside an analog-to-digital converter, or ADC. Ithas 10 LED prietary orth. Proprietary chips exist chips with the exist comparators with the comparators and voltage and divider voltage already divider already The built-in. LM3915 The by LM3915 National by Semiconductor National Semiconductor ha whose output can only be one of two states: high or low. The voltage divider also has built-in comparators and a voltage divider, but instead of controlling each, was for and many was years for many considered years considered the industry the standard. industry standard. presents different fractions of a fixed reference voltage to each Ref pin. LEDs the device merely counts how many outputs are turned on, and delivers When the signal voltage is lower than VDC #1, all LEDs are off. When it exceeds
the result as a binary number. Software algorithms can then be written to tell
#1, the firstright comparator change its from low to high, turning am Theon diagram the VDC right on looks the inside looks anwillanalog-to-digital inside anoutput analog-to-digital converter, converter, or ADC. Itoralso ADC. built-in also voltage and divider, aofvoltage butdivide inst a display what has to doItand whenhas tocomparators dobuilt-in it. An 8-bitcomparators ADCand woulda contain a depth on LED 1. If it goes higher than VDC #2, both LED 1 and LED 2 will be turned comparators, while 12 bits provides 4095, which is why more then bits equals unts merely how counts manyhow outputs manyare outputs turnedare on,turned and delivers on, andthe delivers result the as 255 aresult binary asnumber. a binary Software number. algorithms Software algorithms can be canwritten then be to wr tel on, and so forth. Proprietary chips exist with the comparators and voltage more detail. dobitit.ADC An 8would bit ADC contain would a depth containofa255 depth comparators, of 255 comparators, while 12 bits while provides 12 bits4,095, provides which 4,095, is why which more is why bits more equalsbits more equals detail. more d
RE-BUILT FROM THE TOP-DOWN
Today with digital audio, it is very important to avoid 100% bit saturation of the digital converter. The kind of scale we’re most used to refers to that point as 0dBFS, or 0dB Full Scale, and proceeds downward in negative decibels. The point at which any analogue ins or outs on a digital device agree with 0dBu could be –24, –20, –18, –12dBFS, or any other figure that takes the fancy of the manufacturer who’s gear you are using. Again, there is no single universal standard. On stand-alone devices a dBFS meter is likely to be constructed using LEDs, as opposed to a moving needle, but on more complex equipment will probably occupy part of a graphical display. Some meters now display both VU and PPM simultaneously, with VU expressed as a solid bar, and PPM as a moving dot above it. Such schemes AT 44
were proposed as far back as 1981 by Michael Dorrough of Dorrough Electronics, who has since gone on to invent a new type of ballistic which he calls the ‘Loudness Meter’, or LUFS (Loudness Units Full Scale). Why not create a new meter? If anything is clear from all of the above it is that anybody who ever created a meter scheme was really just interested in getting some visual assistance to see what was happening to the signal, or the equipment. There’s nothing intrinsically empirical about any of the PPM schemes; people just decided to do things a certain way. If it helps an audio engineer perform their job well, there is nothing to stop you, me, or Mrs Tech Bench meddling with timing and alignment, or applying new wacky algorithms to create any kind of meter we want. Who knows — if we come up with a good one, it might catch on.
AT 45
REGULARS
PC Audio Buying a new audio interface can also mean losing an old friend — here’s how to minimise your stress levels. Column: Martin Walker
The first audio interfaces were plug and play ‘soundcards’ that you installed inside your PC, firstly in ISA format, then PCI, and most recently PCIe. Early models were primarily intended to accompany games playing, and didn’t really offer audio quality good enough for commercial recordings. I can still remember the first soundcards I reviewed that tipped the balance for me over to ‘professional’ audio quality; the 20-bit Echo Gina, Darla and Layla models, released in 1997 for Windows 95/98. As the available options mushroomed, I/O began to expand into rack-mounting cases, and nowadays the format choices can seem rather more complicated. I still see a lot of people asking for advice on upgrading PC audio interfaces, and they seem to fall into two main categories. The first comprises musicians reluctantly abandoning their current interface because they’ve either upgraded to Windows 10 only to find it doesn’t have W10 drivers, or upgraded their PC only to find it doesn’t have the right ports to connect it and there isn’t a suitable adaptor card. I suspect most of us have been in a similar situation at least once or twice by now. A few dedicated people go to great lengths contorting existing legacy drivers so they work on newer versions of Windows. Even manually hacking drivers to the point of having no idea whether they are running successfully on the current operating system. Inevitably we all have to abandon ship sooner or later. This is the nature of the modern computer beast. NARROWING DOWN THE OPTIONS
If you have to replace a favourite PC audio interface then your research should be fairly standard: what do you want to record (determining the combination of mic, instrument, and line inputs you’ll need); what digital extras will help you integrate the proposed new interface with the rest of your gear (perhaps coaxial or optical S/PDIF, ADAT and AES/EBU); and how you’re going to allocate its various analogue outputs (depending on whether you’re producing in stereo, surround sound, or need to patch in hardware rack gear). Along with other options such as MIDI I/O, headphone monitoring and even integral DSP AT 46
effects, this should narrow down the options considerably. However, a few caveats on format. Don’t ignore the USB format, as it’s still perfectly capable of running quite a few channels with respectable latency, and while Firewire seems to be well on the way out despite its popularity a few years back, don’t be sucked into the mindset that Thunderbolt is the only modern choice. Yes, there are some wonderful interfaces out there, but few actually perform any better because of their Thunderbolt interface. MORE PLEASE
The second category of musicians wanting to upgrade their interface are driven by concerns of audio quality or features, either wanting a complete replacement that sounds better, adds more features, or fancy expanding/improving their existing one to achieve better recording/ playback quality or more simultaneous channels. Once again, don’t be sucked into the mindset that the latest interfaces will always have significantly better-sounding converters. While there have been huge leaps in converter technology over the last couple of decades, most interfaces released during the last four or five years will sound great (even budget ones!). Audio improvements are likely to be incremental rather than radical. You’ll find a similar situation with onboard mic preamps. While budget offerings have improved over the years, quality mic preamps have been readily available at bargain prices for many years now. Yes, if you allocate lots more money for your new interface, it may well sound slightly better, but in many cases spending that amount of money on adding acoustic treatment to your recording environment will probably improve the sounds of your recordings even more. EXPANDING SIDEWAYS
If you’re still happy with your existing interface and its driver support, yet need more or better audio features, you can instead try the bolt-on approach. For instance, I’m still using a basic PCIe interface I bought way back in 2001, but have subsequently greatly increased its audio playback quality by plugging its S/PDIF output into an
upmarket standalone D/A converter (in my case the Lavry DA10). A few years after that I added a high quality stereo mic preamp with built-in A/D converters (the late Audient Mico) that plugs into my soundcard’s S/PDIF input. As and when I finally need to replace my old soundcard, I can still carry on using my standalone converters and preamps. Similarly, if you find yourself running out of analogue inputs, many musicians already own interfaces with ADAT I/O that will let you plug in eight more mic preamp channels. At the budget end of the market, examples include the very affordable Behringer ADA8200, while others such as the Presonus DigiMax DP88 could noticeably improve recording quality, as will Audient’s ASP800 and 880 models, the former even offering versatile ‘retro’ sound-shaping options, such as MOSFET harmonic saturation and a transformer input, to their otherwise clean sound. For many musicians, such expansion options are very enticing, although sometimes it can make more financial sense to replace the entire interface with a new one that already incorporates all the channels/features you are likely to need for the foreseeable future. DRIVER QUALITY
Finally, while many musicians still seem to think that the mic preamps and converters are the only keys to sonic excellence in an audio interface, for some, low latency performance may prove a more important deciding factor, especially if you’re recording bands and need ‘real-time’ monitoring for their headphone feeds. Many interface manufacturers buy off-the-shelf converter chips that come with pre-written drivers. They have to depend on the chip designers to update them to either cope with new operating systems (this is why so many interfaces fall by the wayside when Windows versions change), or unexpected bugs. Other interface manufacturers take the longer road of designing chips in house and writing the drivers themselves (RME is probably the most famous example), so they are able to update and refine their driver code as part of the regular development and maintenance cycle. If low latency performance is particularly important to you, read the AT reviews to see which ones really cut the mustard!
AT 47
REGULARS
Apple Notes Apple Sticks Its Tongue Out at WWDC Column: Brad Watts
It was that time of year again. World Wide Developers Conference in San Jose, California ran from the 4th to the 6th of June and, as always, focused on software developments for, well, developers — those folk who create software for the macOS and iOS ecosystems, which are steadily converging. That convergence is promising to make cross-platform apps a reality, with Apple showcasing its own ports of iOS apps to macOS. Unfortunately, there wasn’t much mention of apps going the other way; the likelihood of porting your favourite DAW to iOS still a distant proposition. Disappointingly, there was no news regarding anything audio or Logic Pro X. However, there’s a lot to like about the soon to be released iOS 12, which is compatible with iPhones as far back as the iPhone 5S — a five-year old device. That in itself is an impressive feat for any manufacturer. While you’d expect a current operating system to run like a proverbial dog on such an old device, iOS 12 is imbued with smarts to let OS run extremely quickly on older devices. iPads from the iPad Mini 2 — another five-year old device — and up will also gain performance enhancements. Basically, if your iDevice can run iOS 11, the upgrade to iOS 12 should result in better performance. An interesting iOS 12 Siri feature that could prove to be extremely useful is Siri’s new shortcuts. Say you want to request particular actions on your phone whilst driving, you can select from a multitude of Siri shortcuts and assign those a ‘trigger’ phrase. Other less inspiring iOS 12 additions include a new regime for viewing and dealing with notifications, and good ol’ Animojis now recognise tongue and winking movements… great, that’s all we needed. You can create your own personalised emoji based upon how you look, building up what Apple call a ‘Memoji’, much like you would on an Xbox 360. A novelty to say the least.
surround format via your top-shelf Apple TV device. Apple has also promised that movies bought on iTunes will be upgraded to Dolby Atmos and 4K versions as those formats become available. While Atmos has already been possible with units such as the Fire TV, the Xbox One, and the Chromecast Ultra for a good while now, adding Apple to the fold means post-production mixers will increasingly have to familiarise themselves with Atmos as consumer appetite grows. So what is happening in the Apple and audio world? Not much to be frank. While Apple has been rampaging along with its iDevices, the desktop arena has slowed to a complete standstill as we wait to see the new Mac Pro… in 2019. Twothousand-and-nineteen! Six entire years since the last official Mac Pro. It’s no wonder professional users are revamping 2012 Mac Pros, wrestling Frankenstein Hackintoshes to the mat, or simply jumping ship to Windows 10. It’s a sad state of affairs over at One Infinite Loop.
SURROUNDED BY ATMOS
BEAT PEACEMAKER
Still, the closest news we hear regarding audio at WWDC is the inclusion of Dolby Atmos on the Apple TV 4K and within iTunes. If you’ve shelled out a pile of cash for a Dolby Atmos capable receiver you can now access Dolby’s sophisticated
With so very little to report, I’ll point y’all to a quirky little instrument plug-in I recently came across, courtesy of Beatmaker. This outfit creates virtual instruments of a vintage electronic persuasion, with the free offering of RT-7070.
AT 48
RT-7070 is a pretty decent recreation of one of Roland’s seminal drum machines — the TR-707. While the TR-707 isn’t as revered as the 808 or 909 Rhythm Composers, it still found its way onto countless electronic productions from its release in 1984 and throughout the 1990s. One of the unit’s saving graces was it offered both MIDI and DIN sync, making it an exceptional translation device for many, even if they didn’t use the sounds. The rimshot output could also trigger devices accepting a voltage pulse. Personally I’m a fan of the drum sounds, and once you hear them you’ll recognise it’s pushy snare. Beatmaker’s rendition offers 20 different drum kits based on the original 12-bit samples and is utterly authentic in its dry form. It’s available as a VST/VST3 and AU instrument plug-in for both macOS and Windows. Or if you just want the samples you can download those, too. Aim your browser to beatmaker.xyz and grab it before the price goes up.
ZLX
is now streaming on Bluetooth
Introducing two new members of the ZLX series: ZLX-12BT & ZLX-15BT. The superior components, coverage and clarity of the world's best-selling portable PA speakers, combined with the latest Bluetooth technology to deliver the lowest-noise, highest-fidelity audio streaming experience in their class. Wireless convenience with Electro-Voice sound quality.
3+2 YEAR EXTENDED WARRANTY with product registration
ZLX-12BT ZLX-15BT Ph: 1300 026 724 | Fax: (02) 8850-2230
| Email: boschcomms@au.bosch.com | www.boschcommunications.com.auAT 49
REVIEW
Sennheiser Evolution G4
Wireless Microphone & IEM Systems It’s been 10 years since the G3 launch. The theme of G4 is ‘more’. Review: Christopher Holder
How big is the wireless mic market? Have a guess. I found a report from Grand View Research (released in October last year) that valued the global wireless market in 2016 at US$2b, with the handheld segment accounting for about 65%. The market is set to hit US$3.5b by 2025, according to the report. So we can all agree: it’s a significant market. Sennheiser and Shure are like the Coles and Woolies of the wireless market. Yes, there are other competent players and there are niche competitors but Sennheiser and Shure rule the roost. 10 YEARS SINCE G3
NEED TO KNOW
10 years ago, Sennheiser launched the third generation of its Evolution wireless. It hit a real sweetspot — the price was right and the systems were rich in features. The market has been anticipating a G4 for some time.
PRICE See box next page CONTACT Sennheiser (02) 9910 6700 www.sennheiser.com
AT 50
PROS Wider frequency bands Easy to build high-channel count systems Backwards compatible with previous series
It’s arrived. The G4 offering has been somewhat simplified, or at least easier to get your brain around the alternatives. There’s the 100 Series, the 500 Series and an IEM series. The 300 series is aimed more specifically at commercial AV purposes. The ENG kits are called 100P and 500P. The headline improvements include: • Easy linking and setup of up to 12 100 Series channels. • Up to 72MHz bandwidth for up to 32 compatible channels with 500 Series (100 Series remains at 42MHz). • More power; all transmitters except the 100 Series adds a ‘Hi’ 50mW RF output setting, equivalent to the power of the 6000 series. There are a few other key points that will squeeze a sigh of relief from rusted on users. There’s now a dedicated Esc button on the rack units. For infrequent users of G3, exiting menus
CONS Same old battery approach
THE FREQUENCIES In Australia, the available bands for the 100 series are: AS (520-558), G (566 - 608), B (626-668) and the 1G8 band (1785-1800). Spectrum allocation gets a fair bit wider for the 300-500 series, but the widest available band is 72MHz: AS (520-558), GW (558 - 626), GBW (606-678). Naturally, the maximum number of selectable frequencies has also been expanded to 3520, up from 1680 on the G3 series.
SUMMARY It’s an incremental evolution of the Evolution series, maintaining its reputation as a reliable, rock solid performer at the right price. Evolution customers want to build systems with higher channel counts and now it’s easier than ever.
NOTES FROM A G3 USER Mark Davie, Editor: For some, RF is second nature. Keeping hundreds of channels on the air, with the antennas and jurisdiction juggling that comes with it? No problems. For others of us rolling into theatres and small stages around the country, it can be a dark art — like acoustics, or IT networking. Even with a healthy grip on the science, every now and then a channel can just disappear into a mess of static for seemingly unknown reasons. That’s why setting up and shifting channels has to be simple for a wireless system to succeed. Even the base 100 series Sennheiser G3 units handled that task with relative ease. Either elect a simple
system of banks and channels, or individually tune in your frequencies. Those cover all your bases, and IR sync is the cherry on top. Still, even with this simple procedure, setting up multiple units has always felt a bit like ‘manual labour’. There had to be a quicker way than churning through the same steps for each unit. With G4, Sennheiser has cut that time significantly. There was always a data port on the back of G3 100 receivers, but on G4 they actually do something. Link up as many as 12 units by daisy chaining their data sockets with the provided RJ10 cables and you’re ready to go. Now you simply use one
receiver as your master unit. Dive into the menu, scan the channels, and when you’ve found a bank with enough available frequencies you can simply populate the rest of the units with those frequencies by scrolling through the bank and confirming which receiver you want to absorb that channel. It saves having to dive into the menu of each particular unit, and any time saved during setup is gold. Of course, if you want to plot your frequencies manually, you can squeeze 20 channels out of a 100 series system. Up to you.
IEM: GETTING IN YOUR EAR Sennheiser has simplified the evolution IEM proposition. There’s one series priced at $1499. The bodypack is well built with a clear OLED display and volume knob that feels right and is easy to manipulate. Out of the box it’s designed to accept a stereo feed. Switching to mono
could be a stumper. The power supply has also been squished into a low-profile design that only occupies one socket space on a power board! It’s the little things. Before you get too excited, it’s still the bigger wall wart for the antenna distribution accessories. The 100 series handheld transmitter also now comes with a programmable mute switch. Yay? You’re wondering. Well, never fear, you can disable it completely. On the looks side, well, the new white receiver chassis and space grey IEM units look hot, and all the transmitters now have OLED screens. As you’d expect, there’s a variety of configuration kits (instrument, lav, headset) along with the choice of your favourite Sennheiser capsule. And like all Evolution products, they use Sennheiser’s HDX compander and are interchangeable, if you only want to upgrade one component. HIGH PERFORMER
I was sent a channel of 500 Series handheld (with a 965 head), a channel of IEM and a 100 Series headset kit. I’ve largely been a Shure wireless guy. The experience I’ve had with Sennheiser wireless has mostly been with its cheaper 2.4GHz D1 models. Meanwhile I’ve mostly had experience with Shure’s SLX and BLX series. The 500 Series handheld system feels like a step up. You firstly notice the heft of the transmitter — the mic is a solid aluminium construction and you know you’ve got it in your hand. Apparently the handheld is slightly trimmer than G3 — it’s still reassuringly solid. The sound of the e965 condenser capsule is superb — big and detailed. There’s also no handling noise to speak of. The wireless performance was faultless. This is Generation 4 after all, it’s not Sennheiser’s first wireless rodeo, and it shows.
required a look at the Quick Guide but that’s no hanging offence. The IEM kit ships with IE4 headphones which work well, albeit without any insert alternatives. More likely than not, most people will have their own ear pieces. The system is compatible with Sennheiser’s
WSM (wireless system manager) and will happily work with up to 16 systems (more than enough). I suspect that there are plenty of groups (bands, theatres, churches etc) that are still sitting on the IEM fence, not knowing whether to jump or not. The G4 IEM is an excellent option.
MORE IS MORE
CONSTANT EVOLUTION
Everyone wants more wireless. When Evolution wireless started life, the expectation was that the customer may want a channel or two. If you needed eight or more then you were entering the pro realm and should pay accordingly. Now, eight channels is nothing. Theatre, touring bands, churches will routinely have that many channels and more. Wrangling lots of wireless usually requires some organisational chops and a little RF know-how. For G4 to take care of your frequency management is a real asset. In fact, G4’s overall approach to the simplification of working with multiple channels is most welcome. But most people will tell you, the biggest hassle with working with lower cost wireless is the battery situation. It’s a pain in the neck, and if you’re using disposable alkalines then it’s terrible for the environment and expensive.
It’s interesting, the Sennheiser literature says, ‘Evolution wireless users have come to expect an ever-evolving feature set with each new generation, and evolution wireless G4 is no exception’. I think this statement perfectly encapsulates G4. It’s an evolution. Don’t expect a revolution. Saying that, you can expect rock solid performance, great sound, and if you’re going for the 500 series, you’ve got a genuine pro system that can accommodate 32 channels. All at the right price. Evolution has been the go-to wireless for so many for so long, and G4 ensures it will stay that way.
A TAKE ON LITHIUM
I was really hoping for a big step up in the battery situation. An innovative long-life rechargeable lithium-ion set up (or similar) that would shake up this end of the market. My recent experience with Shure’s GLX-D Advanced system’s batteries with integral charge bay was a real eye opener. The downside is GLX-D is 2.4GHz. In Sennheiser world, you can purchase Ni-Mh battery packs and charge bays for up to eight hours of battery life, but not the 13+ hours you get with the lithium alternative. If Sennheiser did something similar to the GLX-D Advanced with G4 it would have ripped the lid off the wireless universe.
PRICE 100 Series: Kits start from $849 (instrument set) Handheld kits start from $999 (835 cardioid capsule or bodypack/lapel kit) 500 Series: Kits start from $1349 (instrument set) Handheld kits start from $1399 (935 premium cardioid capsule) Film kit with bodypack transmitter, portable receiver, MKE2 lapel mic and plug-on transmitter $1999 IEM Kit: $1499
AT 51
REVIEW
UNIVERSAL AUDIO OX Load Box & Emulator Dummy load boxes are just that… dumb. UA’s Ox is much more.
NEED TO KNOW
Review: Preshan John
The struggle is real. You’re trying to lay down electric guitar tracks through a vintage tube amp and quad — in your living room. The guitarist insists the ‘true tone’ of the amp exists beyond the halfway Volume knob mark, but it feels like the beastly 100W head explodes with the slightest twist above two. Mattresses go up and curtains are draped to counteract the tsunami of SPL but with little effect on your neighbour’s nerves. You resort to an amp simulator plug-in. The UA Ox is designed to solve this age-old
problem for guitarists. The Ox can be used in two ways. First, you can hook up a cabinet direct so it acts as an in-between load box to allow your amp to enter its favourable range while maintaining full control over the level sent to the cabinet, without sacrificing tone. Second, you can use the several cabinet emulations which already exist inside Ox to complete your tone. This means you get genuine tube amp tone and feel but with the ability to tap that tone as a line signal into whatever you want — be it a DAW in the studio or a mixing console on the stage.
PRICE Expect to pay $1999
CONS Reactive load boxes aren’t cheap
CONTACT CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
AT 52
PROS Fabulous tone from cabs and effects App integration is flexible and easy to use Enough output options to satisfy most
MILKING TONE
Physical connections include Line Out and Headphone outputs, both of which are postcabinet emulation, then there’s a Speaker output without cab emulation to go directly to a physical guitar cabinet. In addition, there’s an S/PDIF output via RCA or Toslink and a footswitch port. A number of chunky switched knobs adorn the front panel. Out of the box, the six positions of the Rig knob provide quick access to different preset cabinet and mic configurations. The
SUMMARY Sometimes you want to crank your amp to 11. Ox lets you do that by strapping a quality reactive load across its output. With UAD’s software onboard, you can also strap quality simulations and effects across your presets, control speaker breakup, and virtually place a rug under your amp. Ox ain’t no dummy load.
Speaker Volume knob sets the level going to a cabinet and the Room knob controls the room ambience volume. With a Marshall JCM 2000 Dual Super Lead tube head connected to the Ox via the 16Ω input, I plugged in my Strat and had a strum with the Rig set to Position 1. Listening through a JBL eight-inch studio monitor connected to the Ox’s Line Output, it felt like playing through an amplifier — because I was. The best part is I could crank the Marshall’s output to a glorious tube-y crunch and dial down the Ox output to enjoy every drip of tone without blowing the windows out. Now, not all load boxes are created equal, which is why some (Ox included) can cost the equivalent of a boutique amplifier. Ox has a fully analogue reactive load onboard, which means it has a continuously variable impedance and response, to counter the varying attenuation. In short, it sounds good no matter what head you plug in, or what level you run it at. WIRELESS COMMAND
To get the most out of Ox’s inherent abilities you’ll need to make it talk to its own app via a computer or iOS device. After downloading and launching the Mac app, follow the onscreen prompts to first connect to Ox’s built-in Wi-Fi, then connect it to your usual Wi-Fi. Once you are told the software has been registered, the Ox app GUI appears. I love that the interaction between Ox and device is completely wireless, especially when you have the unit sitting with your guitar amps at the other end of your studio. The interface is basic and intuitive. First, there’s the rather extensive list of cabinets ranging from a 1x10 to several 4x12 varieties. 50W/100W output level options are provided and the Speaker Breakup knob lets you “control the harmonics and sonic complexity that occurs only at certain frequencies and volume levels on various speakers”. No complaints here whatsoever — playing through the cab emulations is a thoroughly lifelike experience and UAD’s modelling and DSP prowess shines in each model. Level/Mute/Solo controls for Mic 1, Mic 2 and the Room Mic let you find the perfect balance capturing the cabinet’s tone. Mic emulations include an SM57, MD421, R121,
The best part is I could crank the Marshall’s output to a glorious tube-y crunch and dial down the Ox output to enjoy every drip of tone without blowing the windows out.
160, U67, C414 and Direct. A four-band graphic EQ with HPF and LPF is provided for each mic, or you can choose a parametric EQ if you prefer. I was impressed with the natural character of the Room Mic channel and the Damp switch (which puts a rug under the amp). ALL-IN-ONE
On the Master channel is a tasteful chain of an EQ, 1176 SE Compressor, Delay and Plate Reverb inserts. These effects are what I think makes Ox more than just a load box. It’s a self-sufficient, stage-ready guitar rig. Many guitarists will find there’s more than enough configurability within Ox to get you through a show without needing tons of pedals or extra gear. Of course, the app allows you to create and save your own presets. Click Assign at the top of the app to designate any of these presets to the six positions of the Rig knob on the unit for easy access to your favourites. As a recording solution Ox offers loads of tonal options and outputs to work in any studio setting. Its app integration and presetbuilding ability makes it comfortably handle the demands of guitarists playing live. Whether your home is the studio, the stage, or both, Ox is a well implemented fix for amp lovers who never get to turn their amps up. AT 53
REVIEW
ELEKTRON ANALOG 4 MKII Desktop Analogue Synth Elektron combines the best bits of analogue and digital in its desktop beast, and gets a bit freaky under all those layers. Review: Mark Davie
NEED TO KNOW
I’ve been toying with an Arduino board lately. Not building anything, mind you, just picking it up occasionally and turning it over in my hands, pondering what analogue interfaces I can digitally wrangle with the tweeny-priced processor. These days, digitally controlling analogue is a common combination that makes a lot of sense; you get precise, repeatable control without introducing conversion into your signal flow. It’s everywhere, from audio interface and console makers implementing digitally-controlled head amp gain, to boutique guitar pedal manufacturers like Chase Bliss Audio layering digital control over analogue circuits to draw all manner of sound combinations out of a tiny die cast box.
PRICE $2199 CONTACT Innovative Music: (03) 9540 0658 or info@innovativemusic.com.au
AT 54
Sounds that would barely be achievable in the analogue domain alone, even with a console’sworth of knobs and switches. Elektron has taken a similar approach with the Analog 4, digitally controlling a four-part multitimbral analogue synth underbelly. There’s a fine line between enabling layers of control and introducing too much complexity, a line the Analog 4 flirts with in every instance. It’s an instrument that requires investment, not a desktop ornament you can treat as a hobby. If you’re not already familiar with Elektron’s instruments, you will be diving through menus, you’ll have to decode its nomenclature (like Scale referring to sequence length parameters, not musical scales, and Kits having little to
PROS Four analogue synths in one Moog-inspired low-pass filter Very friendly with other analogue gear Loads of control, sounds & memory
CONS Elektron-noobs will need a manual dive
do with drums), and you’ll have to internalise layers of function buttons and how they apply to different states so you don’t accidentally erase a track preset when you mean to clear 16 beats of a 64 beat sequence. When you do, you’ll be so deep, it’ll be an extension of your creativity. Shortchange the process and you’ll be hobbled by menu browsing and manual trawling. DIVIDING LINES
I reviewed Elektron’s second iteration of this device, the Analog 4 MkII, which has a more refined shape and space grey colour scheme and enhanced internal electronics that are supposed to result in better low end than its predecessor. A major addition are the separate mono outputs
SUMMARY There aren’t many great analogue synths for under two grand. Analog 4 MkII has four of them onboard. Combining the beauty of analogue subtractive synthesis with precise digital control belies the fact this beast can be as random and weird as you desire.
for each of the four onboard synths, alongside the main left and right outputs. Only having a main stereo out was a major knock on its predecessor. The new version also adds two expression/CV inputs to go along with the four CV/gate outputs, better integrating the Mk II into existing analogue synth environments. The Analog 4’s entire front panel has been overhauled. It’s laid out similar to the Mk I, but all the buttons are larger and backlit, instead of having separate LEDs. There’s also more buttons on the surface to unearth things like Global Settings, Tap Tempo, Octave Shift, and the Track, Kit and Sound menus. The division of analogue and digital is relatively straightforward. Anything you’d want to sound analogue is: oscillators, amplitude modulation, filters (ladder and multimode), overdrive, and the amplifier section. While everything controlling those circuits is digital: envelopes, LFOs and sync. The only sections you could argue have ‘missed out’ on the analogue treatment are the noise generator (which benefits from a sliding mix of noise types, rather than a selection between two or three), and the effects; ‘wide shift’ chorus, ‘saturator’ delay, and ‘supervoid’ reverb. CHAIN GANG WORKS THE TRACKS
There are a few layers to the Analog Four MkII’s way of working. On the sound side of things; you have Sounds, Tracks and Kits. Sounds are presets or user-defined sounds, like a ‘bass sound’ or ‘lead sound’, that can be assigned to a Track. There are four Tracks corresponding to the four onboard synths, which can each play a separate Sound. Kits are a collection of Track Sounds, along with FX and CV parameters. On the arrangement side, you have Patterns, which are individual sequences up to 64 steps; Chains, which are multiple Patterns linked in a row; and Songs, formed by a combination of Patterns and Chains over the four tracks. By separating the sounds from the arrangements, Elektron allows you to not only audition the effect of different Sounds on particular Patterns, but you can also switch out Kits to see what an entirely new set of Sounds will do for your arrangement. You must remember to save any changes as a new Kit, otherwise it will alter your original Kit and the sound of its corresponding Song. At the top of the tree are Projects. There are 128 of those, which can contain 16 Songs, 128 Patterns, 128 Kits, and a pool of 128 Sounds
selectable from a Sound Library of 4096 Sounds. Multiply those numbers out and you’ll only have to start thinking about clearing memory when you’ve made 2000 songs. PLUS SIDES OF SUBTRACTIVE SYNTHESIS
Well, let’s dive into those sounds. All four synth Tracks have the same architecture. Each subtractive synth is a dual-oscillator design, with sub oscillators on each (though the level isn’t adjustable). All the main waveforms are present; saw, triangle, pulse or transistor pulse (which adds a bit of saw angle to one half of the pulse waveform, which introduces more even harmonics — in raw form it sounds a little Chip Tune-like). While the oscillators are DCOs, and therefore have no inherent drift, you can globally apply Oscillator Drift in the Sound menu if you want a perceived ‘analogue-ness’. You can adjust and modulate the pulse width of each oscillator for harmonic variation and movement. The sub oscillators come in four varieties: either a square wave (50% pulse) at one or two octaves below the root note, a 25% pulse two octaves below the root note, or a 33% pulse a perfect fifth (seven semitones) below the root note. With careful planning and a lot of patience, it’s possible to build chords with the sub oscillator’s 5th setting and tuning functions. For example, a Gmin7 (G, Bb, D, F) can be achieved by tuning the oscillators up +7 and +10 semitones (for D and F), and letting the sub oscillators handle the G and Bb. You can’t quite get to augmented and diminished chords, or sixths, with a single Track, but it does let you maximise your voice count. From there your sound goes into two onboard filters, applied in series. The first in the chain is a four-pole (up to 24dB/octave) lowpass ladder filter. It’s a fully analogue, transistor-based design, just like the one Bob Moog designed all those years ago. You have control over the cutoff frequency and resonance, but the cutoff frequency only shows MIDI values of 0-127 (with .01 divisions), not the actual frequency. It also can’t be turned off, so for the flattest frequency response, wind it right up, and set the resonance to around 25 (0 is a scooped filter setting). The two-pole multi-mode filter has two lowpass and hi-pass slopes, a 6dB/octave band pass filter, a band-stop filter to notch out frequencies in the sound, and a peak filter for boosting certain frequencies. You can either use the multi-mode for cleaner versions of the ladder filter’s lowpass or stack
them together for some crazy 36dB/octave slopes. There’s also a built-in overdrive, which can travel in positive and negative directions. In the positive realm, it’s a clipping distortion that can add brightness to your sound, turned back past zero, it gets muddier and sounds more like a soft distortion or fuzz. It makes sense putting the control in the filter section, because it’s heavily responsive to how the notes react to the filter cutoff and resonance. MAIL IT HOME
There are three envelopes to control all these parameters; the amplitude envelope, and two assignable envelopes. The first of the assignable envelopes can modulate two user-selectable destinations, but is also hardwired to the cutoff parameters of the two filters. The second isn’t hardwired to anything, but also gives you two userselectable destinations for a total of four. Similarly, there are two LFOs, each with two user-selectable destinations. There’s a host of waveforms available, from simple sine to exponential saw, ramping and random waveforms. You can also multiply the speed parameter up to 2000 times by a factor of the current tempo, 120bpm, or by syncing it to the second oscillator. You can also set its fade style, starting phase, and trigger mode. The FX are laid out in a serial arrangement. The chorus can be sent into the delay and reverb, and the delay can be sent to the reverb. The effects are well-chosen, with useful features like an overdrive on the delay to saturate the tails, a feedback on the chorus, which at its extreme can add a cockpit warning warble to your track. All of them also have high and low pass filters, and the reverb additionally has an adjustable high-shelf for either taming or enhancing the top end. The FX section also has two assignable LFOs. ANALOGUE MOUSE TRAP
At a basic level, there has to be a reason to get out of the box. For many, that will simply be because they want to engage with an instrument, for others, it’s the sound of analogue. For the second lot, while the digital control does result in impeccable tuning, there’s enough analogue girth here to get you away from your mouse. By comparison, I fired up Ableton’s bog standard Analog synth, (which has a similar subtractive architecture) and played the same sequence through both. The base sound of the Analog 4 Mk II’s saw and pulse waveforms AT 55
already carried a richer tonality, partly because it’s impossible to take the filters out of the chain, so you’re always running through multiple analogue stages. Once you get into the filter section, it’s yards apart. The resonance on the low pass filter produced a much more throaty and vocal sound, less harsh than the bird tweet ring you get at midto high frequencies on Ableton’s synth. It’s worth the price of admission alone. While the inability to turn filters and oscillators on and off is all part of the Mk II’s ‘analogue’ charm, it can make it difficult to decode their contribution to a sound when building a patch. With bass lines, it was quite easy to build growling patches with some heavy low pass filtering, a bit of overdrive, and the chorus added simple depth, without clouding the sound. There’s a huge range of sounds available from tight synced basses, to huge saws, subtly evolving pads, computer glitch effects, hardcore analogue percussion, and searing leads. Every synth has its idiosyncrasies, and while you’d assume a purely digital synth would have enhanced flexibility, the Analog 4 greatly outstrips Ableton’s synth in that regard. As far as subtractive synthesis goes, there’s really not much you can’t do. On top of the standard oscillators and functions, you can also choose from other inputs, including external signals, feed the ladder filter output back into oscillator one’s input, or route the output from the preceding track’s multi-mode filter into the input of oscillator two, setting up a chain of events over all of your four tracks. TRIG HAPPY
Once you’ve dialled in your sound, you’ll need to start building a Pattern (which is really a sequence). A Pattern is made up of up to 64 ‘Trigs’. There are two modes of inputting Trigs; Grid and Live Recording mode. Grid Recording is a classic sequencer mode, pressing a combination of a Trig and a keyboard button to manually inlay them. Live Recording involves holding Record while you press Play, the Pattern will start playing (you can set up metronome pre-roll) then you simply play in your Pattern of Trigs using the keyboard. Both are simple enough to use, however, if you’ve laid down a Pattern in Live Recording mode and hit a bung note, you can’t simply switch to Grid Recording mode, deselect it and select the right one. This will reset the note length to its default value. You’re better off playing the entire Pattern again, or editing any issues in the Note menu, which gives you control over Note Value, Velocity, Length and other things like Micro Timing offset to adjust groove. You can also adjust quantization globally or on a per-Track basis. At the front end of every Track, there’s also a fully-featured arpeggiator, which can be controlled via preset configurations or programmed by step. Once your Pattern is there in raw form, the fun really begins when you start messing around with parameter locks. These allow you to set individual parameter settings for each Trig. You can set your cutoff resonance to go crazy on one note, then completely wind it out on the next. Every Pattern AT 56
allows you to manipulate 128 Parameter Locks. You can change that Parameter Lock on every Trig in your Pattern and it still only counts as one of your available 128. You can input parameter locks on the fly in Live Recording mode or manually fine-tune each Trig in Grid Recording mode. The editing functionality doesn’t stop there, you can also set up Conditional Locks, to only play Trigs when they meet certain conditions. This allows you to inject randomness or unpredictability into your Songs and Patterns. They can be conditional on things like what’s occurring on neighbouring Tracks or a general probability percentage that it will trigger. Once you have your Patterns all dialled in, you can further manipulate them live using the 10 Macro performance controls. Each Macro can control up to five parameters from any of the six tracks (including the FX and CV tracks). You can rename the Macros to reflect their function, and use the 10 main knobs to adjust them in realtime when in Performance Mode. IT’S A STEAL
There are a total of four voices, which you can either set to be monophonic — one voice for each track — or polyphonic, through a variety of dynamic methods. You can choose to either reset, rotate or reassign the consecutive voices as notes are triggered. Alternatively, you can set the polyphony to Unison. All of these modes will use voice stealing, so as soon as you run out of voices, the next note will steal the least recently used voice. It’s not the most usable polyphony, given the unpredictability of the results, especially when trying to implement a unison mode in a busy track. Because it’s trying to share four voices between four tracks, there’s inevitably going to be a lot of overlap between notes, and hence, voice stealing going on. If you want to use a unison mode, then you’re probably better off just having one track active. Of course, unpredictability isn’t always bad. You can also force the selected voices to use the sounds of
the four tracks instead of letting it dynamically switch to the played sound for some weird results. In a way, this is the main sacrifice of the Analog Four Mk II’s multi-track approach. Polyphony is more of a limited extra feature, you’re best to think of this device as a four-track, monophonic synth. IN FOR THE DEEP DIVE
It’s a tough job reviewing something as fullyfeatured as the Analog 4 Mk II, it just does so much. There’s just not much it can’t do. As well as all the sonic tweaks, the Mk II is layered with thoughtful features like being able to globally tune the synth, extensively programmable MIDI configuration that can even output old school DIN 24 and 48 if you have a TB303 lying around, as well as four fully customisable CV/gate outs and two CV inputs you can calibrate to interface with your external gear. The Elektron does sound great, and you can appreciate it even more with the individual Track outputs on the Mk II. The design has improved leaps and bounds, while still maintaining a familiar layout. All the additional buttons only make it easier to navigate the interface, rather than cluttering up the panel. The only letdown at the moment is the delay of Elektron’s integration with its Overbridge DAW plug-in. This will allow users to fully control the synth via USB, integrating it perfectly into a digital setup. While it’s a shame the integration hasn’t surfaced yet, there’s really no need to have a digital device connected to get the most out of Analog 4. If you’re more of a keyboard masher, and don’t have the patience for proper sequencing and synth manipulation, then I’d at least wait for it to get connected to Overbridge before taking a good hard look at the sounds it can deliver. Nevertheless, if you’re after an analogue desktop synth that does more than one trick, can get wild if you dig deep, plays extremely well with existing synths, and has enough control you can actually perform with it, this is the synth for you.
Best-in-class SPL performance, sophisticated DSP processing, unrivaled lowfrequency sound reproduction—the new DZR delivers on all fronts realising a truly professional sound with a level of power that belies its compact, lightweight design.
FIND A DEALER au.yamaha.com
EVENTS AND PROMOTIONS yamahabackstage.com.au
LIKE US ON FACEBOOK facebook.com/yamahabackstagepass
FOLLOW US ON INSTAGRAM instagram.com/yamahabackstage
WATCH VIDEOS
youtube.com/yamahaaustralia AT 57
REVIEW
SPL CRIMSON 3
Interface & Monitor Contoller SPL’s Crimson analogue-heavy interface is decidedly unfashionable, but contains precisely what the modern recordist needs.
NEED TO KNOW
Review: Preshan John
Three versions in and I can’t help but feel the SPL Crimson still looks like it belongs in a different era. It’s decidedly unfashionable. While a cockpit-like dashboard of lights and knobs was once the norm, modern interface design has thrown away much of the hands on controls and tucked it all away in the box. The Crimson stands alone in that regard. It’s big, has tons of buttons and knobs, and there’s no PC-control malarkey here. Let me introduce you to the most analogue-y digital desktop interface you’re likely to encounter.
Founded in Germany in 1983, SPL (Sound Performance Lab) is renown for high-end analogue audio processing gear — names like Transient Designer, Vitalizer, Passeq, and Phonitor may ring a bell. So when SPL makes an interface, you can expect its circuitry to be superior to a run-of-the-mill, Chinese op-amploaded plastic box. True to form, SPL has avoided IC-based preamp design, preferring to trust its analogue prowess and design a pair of discrete microphone preamplifiers. Both preamps have 60dB of gain
PRICE Expect to pay $1299
CONS Limited I/O expansion options No power switch
CONTACT Link Audio: (03) 8373 4817 or info@linkaudio.com.au
AT 58
PROS Solid & spacious metal build High quality preamps & analogue monitoring Flexible routing
on tap and the frequency response up to 200kHz means it’s canine and dolphin-friendly. INS & OUTS
The SPL Crimson 3 is a 16-input interface, but not all of those are recordable. One of Crimson’s touchstones is that it also functions as an analogue monitor controller. It means some of those 16 inputs never see your DAW, but are plumbed directly into the monitoring path. Those include three stereo input pairs on 1/4-inch TRS, RCA and 3.5mm mini-jack connections.
SUMMARY The SPL Crimson 3 is a no-fuss, high-quality recording interface with a generous side serve of flexible monitoring options. If your studio doesn’t have sky-high I/O requirements, Crimson 3 might just be the perfect desktop hub for all your gear. It also comes in white.
Then there’s the stereo digital S/PDIF I/O, the input of which is recordable. That’s eight inputs down, leaving eight, according to SPL’s count. However, the remaining eight analogue inputs can’t be used simultaneously. There are two instrument inputs on the front panel, a pair of mic inputs on the rear, and four 1/4-inch line inputs, but you can only record four at any one time. On the output side, two pairs of speakers can be connected — one via XLR, the other via 1/4-inch TRS (with trim pots for calibration) — and you can switch between them with the A to B button. Crimson 3 is a USB 2.0 interface that can run in class compliant mode for plug ’n’ play operation on a Mac or when using it with an iOS device. It’s also the best route for low latency audio. Alternatively, you can download SPL’s bit accurate driver for Mac or PC, which increases latency but also ups the possible sample rate to 192kHz. While SPL records its lowest measured roundtrip latency as 6.49ms (96k, 32 sample buffer), we had varying real world results. Practically speaking, when recording at 48k with a 32-sample buffer, the latency was noticeable in Studio One, but barely there when recording in Bitwig. You could happily record vocals while recording through plug-ins in that DAW. Of course, the Crimson is built on a foundation of impeccable analogue monitoring, so you can simply choose to direct monitor your inputs signals, or blend in any amount of your DAW return signal with the Monitor Mix knob. BACK ON TRACK
I really like the old-school analogue feeling of recording with the Crimson 3. The knobs don’t have any play and are coated in a grippy rubberised finish. The buttons are big and have a reassuring press action, though most click audibly in your cans. There’s acres of space on the top panel, clear legending, and every function — setting gains, routing inputs, monitoring sources — requires pushing real buttons and spinning real knobs. I found the whole approach really refreshing — not to mention, fast — given most of these functions in today’s interfaces are controlled remotely via software. One quirk to the legending is that the back panel has two sets of names; one right way up, and the other ‘upside-down’. Super handy when
you’re peering over the top of the interface to plug in a cable. Once you’ve plugged in an input source, you have to enable the input number pair with a button press. By default, inputs are panned hard left and right when monitoring (i.e. Mic 1 is left, Mic 2 is right, etc.). Activating the Mono button places everything down the middle. With so many switches, it’s a bit of letdown that there’s no power switch. You have to turn it off at the wall, or unplug the power pack to power down. The preamps sounded classy with a clarity and firmness to them that’s a bit SSL-ish. A relatively gentle 80Hz high-pass filter and 48V phantom power are available for each mic input. The gain range of 60dB means the pot can often end up in its upper limits, but the good news is there’s virtually no noise even when it’s hard up at maximum. Visual monitoring comes by way of three LEDs for each analogue input — green for signal, yellow for -6dB (equivalent to +9dBu) and red when you’re clipping (+15dBu). It’s two more than you might typically get but you’ll still want to rely on external metering for transient-heavy sources. MIX OF MONITORING
There’s plenty on Crimson 3 to ensure your artist receives an optimum, no latency headphone mix. There’s a gaggle of dip switches on the Crimson’s underside. Flipping them around will change the architecture of the unit, and setting the fifth switch to On engages Artist mode. In the normal mode, any routing is mirrored in your headphone and main outputs. In Artist Mode, it will automatically route DAW 1|2 to your main pair, and allow you to create a separate mix for your artist that is available in Phones 2 and Speakers B. You can send them DAW 1|2 or DAW 3|4 outputs and use the Monitor Mix knob to dial in the right balance between that and the input signal. Crimson 3 also has a built-in talkback microphone, a nice touch for any desktop interface. Talkback gain is set by way of a tiny pot in between the instrument inputs and it’s automatically routed into Phones 2 and Speakers B when Artist Mode is engaged. Pressing Talkback dims the artist mix. You’ll find a similar pot, labelled Crossfeed, jutting out to its right. It’s a feature peculiar to SPL’s Phonitor products. When the Phonitor
Matrix button is engaged, Crossfeed controls how much of your headphones’ left channel is fed into the right, and vice versa. This technique results in a less exaggerated stereo image that more accurately mimics the sound of speakers in a room. It’s not a fancy DSP-based spatial simulator — like Waves Nx or the discontinued Focusrite VRM — but it’s a nice option to have if you mix primarily on headphones. Because the monitoring section of the interface is entirely analogue, much of the I/O is operational and routable without needing to plug in a computer. It means Crimson can act as a mini analogue mixer. In a pinch, you could take it on stage, hook up a couple of mics, instruments and line sources, then send it off to the respective outputs using just the onboard routing buttons. Super handy. THAT I/O THOUGH
If you could put icing on the Crimson cake, it would be more digital I/O expansion options like ADAT or MADI. Digital I/O is becoming somewhat of a staple these days — an increasing number of desktop interfaces have an ADAT input — and the Crimson’s only option of the twochannel S/PDIF protocol feels limited. I respect SPL for its decision to keep the Crimson 3 miles away from any kind of software reliance. Tracking and mixing with this interface will take you back to the simple joys of routing without requiring a mouse. Plus, doing away with software updates and DSP will mean it’s more reliable. It is larger than the average interface, but if you have the real estate and count an analogue monitor controller as a must-have, then this is a great option. Especially for dedicated mix engineers, or professional scenarios that only require a handful of inputs: like a VO booth, or vocal producer.
AT 59
REVIEW
PRESONUS STUDIO ONE V4 DAW Software Studio One is growing up and catering to a younger crowd. Review: Graeme Hague
NEED TO KNOW
Way back in 2009 Presonus’ Studio One DAW software was introduced to the industry as a ‘bloat free’ application. It offered plenty of features, it just promised to deliver them without any of the baggage its competitors had accumulated by maintaining compatibility with previous versions. A crime you can hardly blame them for. After all, one studio engineer’s absolutely must-have, Grammy Award-winning workflow function is often another musician’s,
PRICE Expect to pay Artist: $139 Artist Education: $79 Professional: $499 Pro Education: $299
AT 60
CONTACT Link Audio: (03) 8373 4817 or info@linkaudio.com.au
“Why the hell would I ever use that?” defunct menu item. It’s tough to trim functions that seem largely obsolete without prompting someone’s outrage, just as it’s difficult to decide which improvements are truly worth introducing. Developers can’t react to every Feature Request thread on the forums. Almost 10 years later, Studio One hasn’t bloated, but it’s put on a bit of weight — or maybe muscle is more accurate. The software
PROS Ripple editing Impact XT gets loads more pads Sample One XT can sample from anywhere
CONS Traditional recordists had their day in the sun with V3
has inevitably grown and expanded significantly to cater to users’ demands, and the new release has added more. This latest Version 4 has an interesting focus, and that’s caused some grumpiness around the chat rooms, but I reckon it’s not warranted. The 2015 release of Version 3 was chock full of cool goodies to delight users who prefer to point microphones at things, play instruments — particularly keyboards — in real time, and tweak
SUMMARY While some users may be disappointed to find the focus of this upgrade is on music production, Version 4 brings Studio One ever closer to a DAW that eschews the need for third-party software.
Designed for Excellence in Every Detail. Introducing Apollo X. UA’s new flagship interfaces, featuring elite-class A/D and D/A conversion, new HEXA Core UAD plug-in processing, Unison preamps, +24 dBu operation, Surround Sound,* and more. All in a sleek Thunderbolt 3 package for Mac and PC.
w w w.u a u d i o.c o m * Surround
Sound features coming Q4 2018.
Distributed Australia wide by CMI Music & Audio, find your nearest authorised dealer at cmi.com.au AT 61
SAMPLE ONE XT
IMPACT XT
By clicking on the Presonus logo at the top, the GUIs will cycle through a variety of colour schemes.
With eight banks of 16 pads, Impact XT now offers a total of ... ah, a gazillion pads.
mixdown plug-ins (gotta love the Fat Channel). With Version 4, now it’s the music producer’s turn; with improvements to functions that support slicing and dicing clips, sampling, and creating EDM, dance and beat music genres. While it might miss the target for some, bear with me because upgrading to Studio One 4 has plenty of appeal for everyone. STUDIO ONE GETS RIPPLED
Okay, among the usual myriad of fine-tuning tweaks, what big-ticket items are on offer? First, a quick mention of something that makes yours truly very happy — the introduction of Ripple editing. As someone who narrates and produces audiobooks — resulting in a zillion small vocal files to tell the tale of a bored housewife and immoral tradesmen — ripple editing is a huge improvement to post-production workflow. A small but important addition is a MIDI drum editor view. Many musicians, myself included, program everything in the standard Piano Roll mode, but plenty of people prefer the dedicated drum editor view for percussion. Presonus has put together a comprehensive drum editor GUI with tools and functions that will let you turn your most basic kick-snare boredom into Mike Portnoy mayhem. Similarly, a Patterns function is kind of like a ‘take’ feature for your MIDI programming. Really, it’s a step sequencer that allows the keyboard-inept, like me, to build MIDI programming one note at a time. The new angle to Patterns is variations on each clip. You can create multiple versions of each one and swap between them all without losing earlier ideas. AT 62
Writing notes on each channel — actual textbased reminders and details — is another low-key but very handy new feature. Sure, some of you may yawn and say, “DAW X has had channel notes for ages”. Still, I like that Presonus has plainly assessed this feature, listened to its user-base, and considered it worth adding, even if it is a bit old hat. Visually, the DAW has a lot more colour schemes from which to choose including some eyeball-searing ‘light’ schemes that are a far cry from its normal utilitarian, gun-metal grey appearance. I’ve always been a fan of aesthetically pleasing GUIs, finding them inspiring — call me shallow — and there’s now a large palette of colour designs to match the wallpaper in your studio. Also, the layout of the mixer has been changed with some buttons shuffled around and icons redesigned ... to be honest it’s hard to get enthusiastic there. THE IMPACT OF XT UPGRADES
The serious excitement in Version 4 is the overhaul applied to two of Studio One’s native VST instruments, Impact and Sample One, now respectively called Impact XT and Sample One XT. Impact is a sample player previously offering 16 assignable trigger pads with editing and soundcontouring effects on each. The XT upgrade increases the number of triggers available by multiplying those 16 pads by eight banks. Plus, the parameters for each pad have been enhanced and include multiple layers for each. It’s become a VST producers and DJs can use without ever running out of resources. Musicians who design their own synths and instrument presets can endlessly tweak and twiddle.
Sample One XT is a sample processor and likewise a supercharged version of its predecessor. A new Record function can source a signal from anywhere in the DAW — the output, a subgroup, or even a track or VST instrument direct. Any kind of sound-grab can be recorded in real time and tortured into what you really need. The appeal of these two upgrades is how Presonus has made Impact XT and Sample One XT accessible and intuitive to users who might normally shy away from these kinds of instruments. I’m not suggesting that anyone who prefers a matched pair of microphones in front of an acoustic guitar is suddenly going to give Daft Punk a run for their money, but Sample One XT in particular has become a tool that can solve problems and provide creative results. Don’t write them off as beat-boxing, EDM instruments, instead consider them as clever sound processors with intriguing potential. DOING IT ALL
Studio One 4 certainly has a focus on less traditional signal paths and more modern musicmaking practises. However, in the overall scheme of things this latest upgrade is really bringing the electronic music production tools up to par with those functions addressed in Studio One 3, and is a further step towards the holy grail of a DAW capable of anything you need. In the process Version 4 makes some specialised, perhaps slightly daunting, features more accessible to everyone. At first glance it’s not everyone’s cup of DAW tea, but I suggest you have a closer look. Don’t worry, you won’t suddenly break out the ‘Backpack Kid’ dance in the vocal booth if you don’t want to.
THIS IS STATE-OF-THE-ART WIRELESS COMMUNICATION
TS
PE N
5
NG DI
PA T E N
BOLERO WIRELESS INTERCOM • • • • • • • • • • •
Up to 10 beltpacks per antenna 100 antenna, 100 beltpack system capacity Best-in-class voice clarity “Touch&Go” beltpack registration 6-channel beltpack plus dedicated REPLY button Built-in microphone and speaker for Walkie-Talkie mode Smartphone integration via Bluetooth Ergonomic, robust beltpack design Sunlight-readable display with Gorilla Glass™ Decentralized AES67 IP networked antennas Seamless integration into RIEDEL‘S ARTIST intercom matrix
www.riedel.net AT 63
REVIEW
STEINBERG UR-RT4 Audio Interface
The Neve name may be the ultimate clickbait for gearheads. Will Yamaha’s collaboration with Rupert Neve Designs keep us engaged beyond the headline?
NEED TO KNOW
Review: Andrew Bencina
PRICE UR-RT4: $999.99 UR-RT2: $599.99 CONTACT Yamaha Music Australia: 1800 805 413 or au.yamaha.com
AT 64
PROS Two highly usable preamp shades Built like a tank Keeps things simple Software solutions no matter your platform
CONS Hardware minimalism at the cost of direct access Monitor mix software a little outdated Onboard DSP underpowered Absence of digital I/O reduces flexibility
SUMMARY It’s easy to be cynical about big name collaborations but it’s the combination of four Yamaha D-PREs with tailored Rupert Neve Designs transformers that makes the Steinberg UR-RT4 worthy of consideration. Not as expandable as the competition, the UR-RT4 is for those who make their digital decisions using analogue logic.
No doubt, the first thing you noticed about Steinberg’s UR-RT desktop interfaces were those tailored Rupert Neve Designs transformers, integrated into the signal path as switchable inserts following the Yamaha-designed D-PRE preamp outputs. Beyond the RND-adornments, the two UR-RT boxes are fairly straightforward: the four in/ two out UR-RT2, and six in/four out UR-RT4, each channel 24-bit/192k audio and MIDI to and from Windows, macOS and iOS devices, via USB 2.0. Notwithstanding the differences in channel count, size and price ($599 and $999 respectively), the UR-RT sisters share the same hardware, driver and bundled software (Cubase AI, Cubasis and Basic FX Suite of VST3 plug-ins). For this review I was working with the more amply endowed of the two. UR WHAT UR
The UR-RT4 is built like a brick house. It’s reassuringly hefty, plain-looking yet refined, and bedecked with silky aluminium knobs. While few of the connectors are obviously panel-mounted, the boards are secured to the chassis in key locations (particularly at the USB connection) and there is no ‘play’ during connection and disconnection. While it flirts with the look of a rack-able interface — it’s basically 1RU high if you minus the rubber feet, and about two thirds of a rack unit in width — it’s officially a desktop unit. It’s powered by a 12V DC wall wart with a generous 2.5m cable, though the absence of a locking connection or cable restraint is mildly troubling. On the front you’ll find four combo connectors (two Mic/DI, two Mic/Line), each with LED peak indicators — additional signal indicators would have been a welcome addition — and a +48V phantom LED per pair. Regrettably, phantom power is switchable in channel pairs only. The switches are located at the rear of the unit on the recessed panel, affording some protection against accidental activation. Next to the inputs are four illuminated switches for the insertion of the RND transformers (more about the trannies later) and the input gain controls for each channel. The pots appear to provide analogue control of gain, with no stepping or associated gain indication within the RT4’s software interface, they also don’t feature a numerical legend. Space is tight, and for my fat little fingers there was just enough room to adjust one without disturbing those alongside. The RT4 is blessed with two discrete headphone outputs each with their own level control — and there was plenty of level on tap. The first mirrors the Master/1-2 outputs while the second is software-switchable between outputs 1-2 and 3-4. Alongside these is a level control for the Master output. The rear accommodates a smattering of TRS I/O pairs: Line Inputs 5 & 6, the Master stereo bus (post Output pot), as well as Line Outputs 1 & 2 (pre Output pot) and 3 & 4. MIDI DIN I/O, USB and a switched power input (it’s not as common as it should be) round out the connections. While there’s certainly a great deal crammed elegantly into limited space the absence of any
digital audio connections (ADAT, AES/EBU or S/ PDIF, for example) may be notable for some users. I’d hazard the absence of per-channel phantom switching, and lack of access to physical control of the available polarity and HPF toggle functions might be more contentious for some. I must stress however, that I don’t see these choices as negatives. They are choices. Choices that put the RT4’s likely user in perspective. WORK WITH ME
Installation to a Windows 10 system using the packaged CD was straightforward; as we’ve come to expect. I also elected to download and install Cubase AI, using the bundled license. At over 6.5GB I wonder if the CD couldn’t be replaced with a thumb drive and also include a recent version of the DAW for both Windows and macOS to make things even simpler. Cubase AI is not prescribed, but if you don’t have any existing DAW affiliations the provided version is relatively powerful and incorporates direct access to the RT4’s settings and monitor mixing capabilities. You can even save custom configurations within each project. For the remainder of this discussion I’ll instead focus on the dspMixFx application, required by all non-Cubase users to configure the driver and DSP settings. Both the UR-RT4 and RT2 incorporate an onboard SSP2 DSP chip to handle monitor mixing duties. While they can run standalone, with the last configured Scene (including the capability to run as a four-channel preamp with inputs routed directly to outputs), you need the software interface to access many controls. Sample rate and Buffer settings are standard but you’ll also find settings for the cutoff frequency of Input 1-4’s HPF, channel assignments for Headphone bus 2, level calibration for line inputs 5 & 6 (-10dBV or +4dBu) and Loopback (the routing of driver output pairs back into the equivalent driver inputs; handy for recording screencasts, presentations, software instrument outputs, or any number of other possible uses). dspMixFx allows users to configure up to two low-latency monitor mixes (under Windows 10 the roundtrip latency was 0.94ms when running at 48k) blending input sources with software outputs. The mixer incorporates EQ/Compressor channel strips and four Yamaha guitar amp emulators which can be monitor-only or applied to recordings. An effects send routed to the REV-X reverb is also included. All of these useful and highly usable onboard effects are also bundled as VST3 plug-ins. Just be sure to check they’re supported by your DAW. If you’re monitoring via your DAW, the roundtrip latency performance doesn’t equal market leaders’ but it’s definitely within a good range for most applications. Steinberg do however deserve credit for accurately reporting this latency within the driver (add the Input and Output latency figures to determine roundtrip latency). While latency performance was mid-market, this additional buffer may explain the far more positive results in my audio stability testing; which will be at least as important for those considering live applications.
The real issue with dspMixFx and the RT4 is the lack of DSP power. While the smaller RT2 may be well served by the onboard DSP, it’s not possible to insert a mono instance of the channel strip onto all six input channels of the RT4. What’s more, the amp sim (only one per device) and channel strips cannot be chained, the use of even a single amp simulator cuts the number of available channel strip inserts to two, and while volume and pan controls are unique to the two discrete monitor ‘Mixes’, per-channel reverb send settings are only active in one of the mixes with the return applied to both. Notwithstanding, for many simple applications all this would likely be a non-issue. What the dspMixFx mixer does, including the saving of effect settings and mix scenes, it actually does very well; it’s just not as full-featured a mixing experience as those currently offered by RME or MOTU devices, of similar channel counts. MORE THAN MEETS THE EYE?
I know, I know. I’ve buried the lead. What you all really want to know about is those Rupert Neve-designed transformers. Perhaps most importantly, the UR-RT4 sounds pretty damn good. I very quickly jammed out a little five-track ditty (drum machine, P-bass, synth, high-strung acoustic guitar and electric guitar) using Radial JS and Lehle splitters — along with calibration tones — to record simultaneously to channels with the transformers enabled and disabled. For my test ’control’ chains, I used a Radial JDV mkIII DI and UA 2108 preamp. For starters, the D-PRE channels performed very nicely indeed. In fact, the difference between the recordings made through these channels and the controls were at times negligible. When ABing against the ‘transformed’ recordings, the differences weren’t drastic, but they were notable. In very simple terms they had slightly less sparkle but more punch and low end bloom; and for some sources this augmentation was definitely worthwhile. Interestingly, when carrying out earlier listening tests for the DACs and headphone channels I’d noted a similar, though not as pronounced, voicing when making comparisons with other reference devices. While it’s not trying to be everything for everybody, the UR-RT4 does have a sound and it’s more musical than scientific. MORE OR LESS
It might seem trite to point out that the Steinberg UR-RT4 is a UR-RT2 with more; but it’s the extras that let you know where ‘UR’ priorities rest. The I/O count expands while the DSP and digital interface options remain static. Steinberg deliver more of less to keep things simple for users who care about capturing a variety of tones with a minimum of fuss. The bundled cross-platform software options make it an ideal platform for recording musicians or perhaps an upscale solution for podcasters and vloggers. Sometimes a Super Supreme or Long Island Ice Tea is just too much, the UR-RT series identifies a few key ingredients and lets you season to taste.
AT 65
REVIEW
JBL 7 SERIES
Studio Monitors Past and future collide with JBL’s 7 series, combining super-wide, pillowy wave guides with all manner of digital connectivity.
NEED TO KNOW
Review: Brad Watts
PRICE Expect to pay 705P: $1799 each 708P: $3499 each
AT 66
CONTACT CMI Music & Audio: (03) 9315 2244 or sales@cmi.com.au
PROS High SPL I/O out the wazoo Low distortion
CONS Waveguide theory may not be your thing Connectivity comes at a price
SUMMARY JBL’s 7 series monitors may have a historic past, but they’re rooted right here in the present. With all manner of digital connectivity and DSP enhancement, these puppies can be trained to play nice in any environment as part of any system.
JBL, or more succinctly, Mr James B. Lansing, has been trotting out sound reinforcement drivers for the motion picture industry since the 1920s; the musical instrument field with Fender’s use of the D130F speaker for its instrument amplifiers since the late 1950s; and for the professional studio sphere since 1973 with the 4300 series studio monitors. The company’s been around several blocks multiple times, and these days, JBL has pretty much become a household name. But JBL didn’t start out as JBL. The company’s founder, James B. Lansing, kicked it all off in 1927 with the Lansing Manufacturing Company. In 1941, the Lansing Manufacturing Company was acquired by Altec Service Company, hence Altec Lansing. It wasn’t until James Lansing left Altec Lansing to form a company using his initials that JBL was born. Long story short; JBL has been in the monitoring game far longer than most. The company knows its schtick. Not being a manufacturer to rest upon its laurels, virtually each year or so there’s another instalment to its studio monitoring lineage. Most recently, and sitting on my monitor bridge is a pair of the JBL 7 series — the 708P to be precise — the daddy of the outfit. The series offers four flavours. Either the 705i and 708i with five-inch or eightinch drivers in an unpowered format with no DSP, or the 705P and 708P five- and eight-inch biamped models, sporting digital I/O and user adjustable DSP. Before we get too deeply involved with the DSP and amplification we’ll have a look at the drivers and cabinet design from the vantage point of the 708P. PHANTOM SPECS
Cabinet-wise the 708P is a largish box measuring 438 x 248 x 292mm. They’re surprising light, weighing in at 11.5kg. Unique to the 708P cabinet are recessed handles either side of the cabinet, which seem a little like overkill for such a light monitor. Handy for pulling them out of the boxes though. There’s no indication of the cabinet material other than “birch plywood reinforced for mounting”, which you could interpret as ply reinforcement for the top and bottom four-screw mounting points in what is otherwise a box made of MDF material. What the listener sees and feels is arguably more important, which is a smooth black surface with a moulded plastic waveguide and metal grille covering the woofer. The specs indicate a four-hole mounting point system on the rear of the 708P, however I couldn’t see anything rearward for mounting duties. There seems to be a few discrepancies with the published website images and the monitors in the flesh, as the website shows phillips-head screws while the actual monitor uses hex-head fasteners throughout. Maybe this explains the phantom rear-mount points. WIDE & WAVEY
The design incorporates a lower front port placed below the low-frequency driver and upper waveguide around the high-end driver. Both the port and waveguide are part of the entire front baffle — an extruded single piece of plastic. JBL have
introduced yet another high-frequency waveguide in the 7 Series — the ‘Image Control Waveguide’, which of course has a patent application pending. It’s a dramatic departure from waveguides designed by the company in the past such as the LSR series and even my old 4200 series monitors. This waveguide is quite large, taking up the entire width of the 708P, reminding me of ancient AltecLansing horn designs. It’s virtually identical to the waveguide used on the M2 mastering series. JBL says the waveguide provides an ‘acoustically seamless transition between the woofer and highfrequency transducer’ and it certainly seems to live up to this claim. The waveguide also aids spreading of the on- and off-axis frequency response, enabling a wider listening area. As for the drivers, JBL has employed an entirely next-gen design for both the low and high frequency drivers. The high-frequency drivers low-mass annular diaphragm offers a frequency response beyond 36kHz. Annular diaphragms allow for extremely high output and very low distortion. It’s certainly evident with the 708P high-end 2409H driver, and driven by a discreet 250W class-D amplifier it pushes out some high pressure levels. Driven again by a 250W Class D amp is the newly designed 728G low-frequency driver which uses differential drive technology to provide low-frequency performance down to 35Hz. JBL claims the 708P can deliver two to three times the output of other monitors, with 114dB SPL (peak) at one metre. Trust me, they’re loud. They’re also remarkably clean and lacking in distortion, although I’ve still not found any published distortion specifications. Overall frequency response is quoted as -1.5dB from 45Hz through to 23kHz (overall response apparently 35Hz – 36kHz), with the DSPbased crossover network set at 1.7kHz.
When it comes to I/O the 708P is about as featured as a monitor gets. The rear of the 708P houses a typical XLR analogue input, but there’s also an XLR AES3 digital input and an XLR pass-through output. Sample rates can be anything up to 192kHz, including obscure clock-speeds, as the DSP incorporates sample-rate conversion. A Harman HiQnet port enables feature updates and interfacing with control systems such as HiQnet AudioArchitect. HiQnet is compliant with Dante protocol so you can integrate the 708P into networked systems.
HIGH DIGITAL HIQ
When it comes to I/O the 708P is about as featured as a monitor gets. The rear of the 708P houses a typical XLR analogue input, but there’s also an XLR AES3 digital input and an XLR passthrough output. Sample rates can be anything up to 192kHz, including obscure clock-speeds, as the DSP incorporates sample-rate conversion. A Harman HiQnet port alongside the AES XLR ports enables feature updates as they become available. As well as interfacing with control systems such as HiQnet AudioArchitect, HiQnet is compliant with Dante protocol so you can integrate the 708P into networked systems. But what’s to update? Well there’s actually quite a lot. Being DSP-based there’s a tonne of EQ and delay settings. The 708P features onboard room equalisation for compensation of room acoustics. In addition to 12 bands of room EQ, there are eight bands of user EQ so users can customise response curves to compensate for placements behind cinema screens and the like. Once dialled in, there are six memory locations for storing said EQ setups. There’s also user-enabled delay for virtual alignment of speakers relative to the listening position and what JBL term ‘AV Synchronization’ delay to compensate for latency in video displays.
Adjustment of all DSP features is via a simple scroll wheel, a couple of menu buttons and a small backlit LCD on the rear of the 708P. While this arrangement gets the job done it does feel a little under-engineered. The buttons feel as if they’ll fail if you use them too much. Cutting a long story short, and considering the 708P has pretty much any connectivity feature you could require, the question remains about how the 708P sounds. In a stereo setting they’re a very analytical monitor, and no doubt due to JBL’s nextgeneration drivers are both loud and startlingly accurate. The low-end is remarkably tight and succinct, while the top-end is staggeringly detailed. Personally, I find JBL monitors to be a little fatiguing, primarily due to the company’s allegiance to titanium or other metal dome tweeters; but these are far less so. I spent quite some time in front of the 708Ps without feeling the need to tone it down for a while. Far from flattering, they’re an accurate, loud, and capable monitor. AT 67
REGULARS
LAST WORD with
John Meyer, Part II
Recently, Meyer Sound’s top tier descended on Sydney to celebrate its new alliance with Audio Brands. We sat down with the Meyers to talk about 12 year-olds on radio, the Grateful Dead’s Wall of Sound and chasing audio holograms. Here is the second instalment.
AT 68
We haven’t been able to project sound forward, like a sound hologram. It’s starting to happen in video using lasers, but we haven’t got the equivalent with audio. Everyone’s hoping we’ll figure out 3D sound… we’re hoping, too. It’s like the Greeks hoping to fly. I asked my mechanical engineer friends, “Could you build a hang glider out of materials that existed during Greek times or do you need high-tech materials to make a hang glider?” They never figured out how to get a kite that wasn’t tethered to fly. They pulled people up with a string, so did the Tibetans, but no-one ever got a glider to work. Hang gliders seem fairly simple now. Why couldn’t they do it? It’s how I feel about 3D sound; we know what we want to do — project sound so it sounds like someone’s next to you — it’s just that no experiments have worked so far. We’ve been working on that with UCLA for 20 years now and can get it to work a tiny little bit. We could be a thousand years away from this, or someone might discover it anytime. To me, it’s like stumbling across Leonardo Da Vinci’s helicopter which is not clear enough in his picture to see how to build it. Was it even possible? Or, is it a fantasy in his head or a vision of the future? When we do a Constellation system the best musicians and conductors agree it sounds like their concert hall. We know that technology, but we have to physically get rid of room sound, because we don’t have any good way of doing that electronically yet. We did an experiment for the movie industry in a lunch room next to a movie theatre. We tried to see if we could get rid of the low frequency sound that’s making it through the wall from the lunch room. We had a copy of the movie track, so we didn’t have to pick it up with microphones. We started with one speaker and got a zone cancelled, then put another one over there to cancel that zone, but then the first one wasn’t working. After days the room was filled with speakers, and we were just starting to get one note. It’s cheaper to build a thicker wall. We’re trying to make sure we’re not leaving anything out. Our quest has been to try and transfer as much information as we possibly can. Everyone asks if it’s overkill. Overkill is thinking, ‘Maybe we don’t have to do that much work.’ In the ’70s Kodak announced a million pixel CCD would be more resolution than we’d ever need! It was on the cover of a magazine. Really? We’re done? Now 4k is barely here and they’re already talking about 8k television. People talk about overkill in the audio industry all the time… it’s ‘good enough’. People see how far they can compress the stuff
until you don’t listen to it any more. At a certain point, where you won’t pay for it, they’ll back off a little step. It’s a race to the bottom, rather than FM, which was a race to the top. We spent five years developing Leopard. We already had a line array so we decided to see if we could make it really hi-fi. It came from John [D.S.] Adams saying he was down at Disney Hall and didn’t like the sound of the speakers. We needed to build something for classical people, because PA came from the ’20s, when they were just trying to announce and communicate; it wasn’t about a hi-fi experience. The industry has grown up from that perspective so trying to talk about hi-fi and PAs is like trying to talk about a hi-fi walkie-talkie. Everyone thinks you’re nuts. We started on the Leopard project to see if we could build a hi-fi speaker that would be easy to use and have him start to integrate it into his music and playback. In music they talk about how one note would have a different feel than this note. They live in a completely different world than we do from an engineering point of view. We’re trying to measure everything and build things to bridge the gap. You can’t make Adams use anything, so it was super-exciting when he did. Then we did the Blue Horn, another project where we were trying to make a speaker absolutely better. We tried a gas field where you charge up all the air so you have a million little needles and you run electricity through it to create ozone. Then you modulate the ozone, so it’s massless. But that has the same property as a regular loudspeaker so you haven’t gained anything. The problem isn’t mass — which everybody thought it was — it’s that low frequencies just don’t want to come out as quick as the high frequencies. For whatever reason, it just takes longer for them to form. With computers it opens up the possibility that we could reconstruct it. We built the Blue Horn just to see if we could got rid of all the phase anomalies. It was a five-year project to do all of that, get the computer powerful enough to do trillions of calculations and have dynamic range at the end that’s over 120dB, so it’s quiet and doesn’t have any artefacts. We’re like the gold miners from the 1800s who found a way of following porous quartz down to find the gold. They’d tunnel then stop, tunnel then stop, and slowly work their way. Today we need an atomic spectrometer to find porous quartz, it’s microscopic, but they figured out ways of doing that via exploration. That’s what we feel like, it’s the nature of exploration. We have enough tools to figure things out, but the tools get better as we play around.
AT 69
AT 70