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Awards 2020

AWARDS 2020 REWARDING QUALITY AND INNOVATION

Resolution Awards 2020 Recognising outstanding quality and innovation in professional audio

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oronavirus hasn’t stopped the music. If anything, the speed with which we received nearly 2,000 votes shows interest in audio equipment is close to 0dBFS! Designers have created innovative products and, somewhat unexpectedly, many pro-audio manufacturers prospered. In November, Focusrite announced revenue grew more than 53% to £130.1m in 2020. Phil Dudderidge, founder and executive chairman, said: “2020 will be remembered as the year of the COVID-19 pandemic. The Group has benefited in many respects by the growth in demand for our music and recording products, no doubt because so many people, professional musicians and amateurs alike, are having to work at home or having more time to enjoy their passion for music creation.” Meanwhile, British online retailer Gear4music said it served 482,000 customers in the six months to September 2020, up 47% on 2019. The same was true on the other side of the Atlantic. “As soon as the shutdown occurred in March, our sales went through the roof,” said Dusty Wakeman of microphone-maker Mojave Audio. North American online retail also blossomed. “We saw our overall business grow more than 200% during the worst lockdown months, mostly from ecommerce. We’ve emerged significantly stronger since,” said Bobby Montemurro, vice president, sales and marketing, Alto Music (a top 10 US gear retailer). Resolution Awards recognise quality and innovation in professional audio; nomination is our accolade. The winning products here have been judged to be outstanding in these respects by the best informed readership in the industry. Nigel Jopson, editor nigel@resolutionmag.com

© 2020 S2 Publications Ltd. All rights reserved / I


Genelec RAW

Easy on the eye. Easy on the environment. Introducing RAW, an eco-friendly reimagining of our most iconic studio, AV and home audio models. Featuring a distinctive, recycled-aluminium MDE enclosure design, RAW loudspeakers require no painting and less intensive finishing than standard models. The result is a unique design aesthetic that allows the raw beauty of the aluminium to shine through. And because it’s Genelec, you know it will sound as good as it looks – in any setting. We will be donating a percentage of every RAW speaker sold to the Audio Engineering Society’s fundraising initiative – helping this much-loved organisation continue its valuable work throughout the current COVID-19 crisis.

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/ Resolution Awards 2020

ANALOGU E MIXER /SU MMING DE VICE

Winner: Solid State Logic ORIGIN As the first large-format analogue mixing console from SSL in over a decade, the ORIGIN generated considerable excitement when previewed at the 2019 AES show. A purely analogue, inline, dual channel design, with 16 busses, E Series ‘Black Knob’ EQ, 2 listen mic compressors and the classic Bus Compressor, it certainly painted a wide smile on the face of anyone in search of analogue power for band and ensemble recording. The lack of automation features kept the list price to a sweet spot of £33,500 — but with mic preamps switchable from super linear and clean to something more coloured, and unity-gain fader-bypass switches, the console is ideal for DAW-based workflows. The PureDrive mic pre, a new design based around modern FETs and transistors, delivers flavour: “Punching in the Drive button really takes the input stage into the realms of doing what analogue consoles do best — colouring with saturation,” wrote Russell Cottier when he reviewed the ORIGIN in Resolution V19.3. “It sounds just as good as the SSL Variable Harmonic Drive [Duality console], if not

better. Punching a rock vocal, slamming snare or a heavy kick though this stage can really add something special; 2nd order harmonics are introduced at lower gain levels, then as the gain is cranked the 3rd order harmonics are introduced. A light touch across a full 32 channels of a mix can really add a significant weight and thickness.” With 88 inputs at mix-down the ORIGIN represents a console of significant capacity, but with a physical footprint less than 2m wide there’s no requirement for a control room with a huge footprint, power provision and cooling to install the console. “Only time will tell if we have achieved the right balance for today’s hybrid production workflow,” said SSL director of new products, Niall Feldman, “but our instincts tell us that ORIGIN has all the right ingredients to take its place in the legendary legacy of SSL consoles that have helped reshape music production”. www.solidstatelogic.com

Nominees AMS Neve

API

BURL

A ‘small format console with a large format sound’, based on the 80-series console range and priced at £17,950 / $24,950. The 8424’s dual-input channel strip allows for seamless switching between recording and mixing inputs without additional patching. Simple input connectivity is via 24 line-level inputs, dual 1073® preamps, and dual Instrument DI channels. The 8424 offers an analogue mixing platform with 24 DAW returns across 24 channel faders or, for larger sessions, a 48-Mix mode that allows a total of 48 mono inputs with individual level and pan controls to be mixed through the 8424’s Marinair® transformer-coupled stereo mix bus – additional features include Stereo Insert, 2-band shelving EQ and Neve’s proprietary Stereo Width control.

The 2448 has quickly become sought after as a mid-sized console that packs API’s large format punch. The 2448 is available in 24, 32 or 48 channel sizes, with eight busses. This console features an in-line configuration, offering the bonus of two signal paths on each channel. The 2448 also offers Final Touch automation as an option. The API 2448 boasts both 1550A parametric EQ and 560 Graphic EQ giving you the choice of two of API’s flagship EQs. The four stereo returns section boast 500 Series slots to accommodate your favourite 500 series EQ or compressor. The stereo buss section includes API’s 529 stereo buss compressor, delivering the unmistakable API mix buss sound thanks to its 2520 and 2510 op-amps and choice between old/new compression types.

The Burl B32 Vancouver is a 1U rack-mounted, Class A, 32-channel analogue summing bus, with a switchable nickel transformer option. With the Burl B32 Vancouver, you can select eight mono inputs or 24 stereo inputs “to give your digital recordings the sound and separation that analogue consoles provide”. Burl has made its name by concentrating on the audio clarity that comes from its careful component selection. Signal paths are kept to a minimum on the Burl Vancouver, with no capacitors in-line and Class A circuitry throughout for the most transparent sound. Now fitted with BURL’s ‘NextGen Analog’ circuitry, the Vancouver features BOPA8 op amps, which the company say delivers tighter bass and higher definition throughout the audio spectrum. The B32 Vancouver allow selection of the BX5 nickel core transformer, to add greater warmth and a fuller sound to your recordings, and a +6dB Gain button to drive your mixes that little bit harder.

www.ams-neve.com

www.apiaudio.com

www.burlaudio.com

8424

2448

B32 Vancouver

Winter 2020 / III


D AW/ S O F T WA R E S U I T E

Winner: iZotope RX8 The two-time Engineering Emmy Award-winning RX Advanced now allows audio professionals greater audio fidelity, whether repairing streamed dialogue with new features like Spectral Recovery or correcting pitch modulation with Wow & Flutter. The Batch Processor and Loudness Control have also been revamped, saving users precious time when repairing or delivering audio files. RX 8 doubles the previous 16-tab limit, allowing users to view and edit up to 32 files within RX Audio Editor. Horizontal scrolling is now built-in to the Spectrogram Display, the scroll gesture on a trackpad or mouse moves audio across the X-axis for fast edits. New RX 8 Standard features allow production pros to instantly adjust guitar recordings with Guitar De-noise, to rebalance their mix and create or isolate stems with the improved Music Rebalance, and to prep music

for streaming with the updated Loudness Control. Both the standalone audio editor and the included plug-ins allow users to surgically fix audio problems and music performances with RX 8. De-Hum now features independent

frequency reduction bands, and a redesigned, intuitive interface. Loudness Control instantly loads preset loudness targets and conforms production audio to broadcast requirements, monitoring levels using the built-in numeric and history plot readouts for integrated, short-term, an momentary loudness. Spectral Recovery (Advanced Only) restores frequencies above 4kHz, turning bandwidth-limited audio into clear recordings. Wow & Flutter (Advanced Only) corrects pitch variations and fluctuations associated with tape, vinyl and optical transfers. Wow has been designed to fix longer, sustained pitch drift, and Flutter to correct pitch variances that occur at faster rates. RX8 is available in three versions: Elements ($129), Standard ($299), and Advanced ($1199). www.izotope.com

Nominees CEDAR

Merging

PreSonus

The top-tier audio restoration platform has been updated, introducing a new impulse noise eliminator, Retouch 8 and improved performance. The new INR is a real-time impulse noise reduction module that was developed as a more powerful alternative to CEDAR’s Declickle. Retouch 8 is the next version of CEDAR’s spectral editing suite and adds two new capabilities: Matching and Repair. Matching makes use of machine learning to identify unwanted sounds. The user marks one of the offending sounds and the software searches for all other instances, a threshold control allows users to bias decision-making. Having identified all of the matches, they can be eliminated using the appropriate Retouch tool.

Merging Technologies has worked closely with Dolby to achieve full communication between Pyramix and the Dolby Atmos Renderer, and also to bridge the Dolby Atmos 7.1.2 maximum bed size and traditionally larger bus sizes available to Pyramix users. By detecting when a user is mapping the bus-based channel to an object, Pyramix sends the correct metadata to the Dolby Atmos Renderer. Merging is well-known for its powerful MassCore engine, providing near zero latency in a multicore CPU. Pyramix 25th has made the long-awaited shift to Native multithreading for performance improvements. New features include Flux Spat Revolution and OSC support, cloud-based licenses and playback engine performance improvements.

Studio One 5 introduces a fully integrated live performance environment – the Show Page combines playback of backing tracks with patch management for instrument players in a single window. Composers and arrangers will appreciate V5’s new dedicated Score View for the Note Editor. Based on PreSonus’ Notion music composition and notation software, the new Score View is available on its own or as a companion side-by-side view with the Piano and Drum Views, allowing users to enter, view, and edit notes in standard music notation. “Improvements to an already great DAW, intuitive score editor, Show page, useful Listen Bus” was the verdict when we reviewed in Resolution V19.4.

www.cedaraudio.com

www.merging.com

www.presonus.com

Cambridge v13

IV / Winter 2020

Pyramix 25th

Studio One 5


G A L A X Y 6 4 S Y N E R GY C O R E

64 CHANNELS OF ANALOG AUDIO. DANTE, HDX AND THUNDERBOLT CONNECTIVITY. ONE DEVICE. Galaxy 64 Synergy Core is a high-end recording system that connects your entire studio within a 2U rack space while providing high-fidelity audio and onboard plugin processing for your most demanding projects. With Dante, HDX and Thunderbolt connectivity, the world’s first 64-channel AD/DA converter offers unbeatable flexibility when moving sound around your high-profile production facility, live stage, broadcast, or other enterprise applications. One giant leap in the simplification of the most complex operations. Light years ahead in terms of audio performance and workflow possibilities.

antelopeaudio.com

GO BEYOND W H A T ’ S K N O W N.


D I G I TA L M I X E R / C O N T R O L L E R

Winner: Solid State Logic System T V3.0 In July 2020 SSL announced the latest version of its System T broadcast platform, which is designed to expand AoIP integration for direct console routing control for ST 2110-30 and AES67 streams, as well as providing an embedded operating system upgrade for remote production and increased IP expansion and connectivity. This version of System T, v3.0, provides new functionality across the whole console range, including the S500, S500m and TCR. Users who have already committed to the System T platform will be pleased to see enhancements like NGA and immersive audio, and DAW and dynamic automation. System T control surfaces and TCRs run embedded OS Windows to “provide a mission critical platform with an advanced feature set”. For broadcasters, the advantages of reduced development cycles and advanced feature deployment are significant. System T V3.0 migrates to Windows 10 Embedded, from Windows 7 Embedded. Transitioning to Windows 10 allows continued updates, including Microsoft’s latest security additions, plus additional future console features layered above

the embedded OS. “With this new level of management and control, the System T console becomes your routing control system for AoIP of any variety,” said Tom Knowles, SSL broadcast product manager. From V3.0, System T fully supports running TeamViewer directly on-board the consoles for remote access, enabling remote support, configuration and at-home control as if you were sat directly in front of the console. Working from anywhere in the world with a standard internet connection, you can directly access the console and all software control features via any computer or touchscreen device. The presets feature for channel and bus paths has been completely overhauled. Multiple processing blocks can now be stored and recalled at the same time. Filters allow the user to selectively choose which processing block is stored, ensuring only the processing they have intentionally saved is recalled whenever that preset is loaded. www.solidstatelogic.com

Nominees Avid

Lawo

Softube

An eight-fader EUCON control surface designed to work in conjunction with Avid’s Pro Tools Control app for iPad, to provide hands-on control of software. Featuring eight motorised, touch-sensitive faders and knobs as well as touchscreenintegrated keys, soft keys enable the user to perform complex tasks at the press of a single button to speed up your process. EUCON also makes it possible to switch between several workstations at the touch of a button. Up to four S1 units can be chained together to provide a 32-fader surface. The £1040 (ex VAT) S1 can also be connected to the Pro Tools Dock to add dedicated transport controls and more. You can cascade up to four S1 Controllers and the Pro Tools Dock to expand your Set-Up at any time.

Ruby’s Power Core mixing engine provides nearly 400 channels of AES67 and Madi signal capacity in a 1U-high rack space. Expansion slots offer the option to add extra mic, line, studio, AES3, Madi and Dante signal capacity as required. The SMPTE 2022 reliability standard enables IP networks to cope with the unexpected, with protocols to prioritise delivery of real-time digital audio over AoIP. SMPTE 2022-7, also known as Seamless Protection Switching, enables simultaneous dual transmission of identical audio streams via independent network paths, with instant, undetectable switching to backup.

Softube’s €599 10-fader controller comes with two software-based component-modelled analogue console emulations, Tube and Discrete, which offer filtering, saturation and more. The ten faders can be configured to control volume levels, high and low cut filter values, drive and drive character (with emulated desk saturation) and more. The interface allows for switching/ paging the ten faders, catering for mixes with hundreds of channels. There are three Send modes for faders, and the enhanced integration with Studio One, Cubase, Cakewalk and Reaper means you can directly control DAW faders, Pans, Mutes (rather than those in the plug-in) and DAW Sends. Seamlessly expands the hands-on Console 1 experience, bonus Spacialization, Drive and Filter capabilities.

www.avid.com

www.lawo.com

www.softube.com

S1

VI / Winter 2020

Ruby Radio Mixer

Console 1 Fader


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Resolution AWARDS 2020

NOMINATED


D Y N A M I C S ( H A R D WA R E )

Winner: Rupert Neve 5254 A dual-channel diode-bridge compressor inspired by the classic 2254 compressor, developed by Rupert Neve in the ‘70s. Based on the circuit found in the acclaimed Shelford Channel (reviewed Resolution V18.3), the Dual Diode-Bridge Compressor greatly improves on old vintage units without sacrificing the tone. “A diode bridge is a very unique sound characteristic compared to other types of compressors. It is possible to create a cleaner sounding diode bridge than we have here, but we were after a particular flavour of harmonic content”, says Dennis Alichwer, engineering manager, Rupert Neve Designs The unified Timing control on the £2,999 (ex-VAT) 5254 consists of six selectable settings, chosen for different applications – a departure from the slow, fixed attack of the vintage units. Fast and MF settings are designed to help clamp down on more transient signals like drums, plucked string instruments and fast vocals. Med and MS have slightly slower attacks and releases which allow more transients through, while

having a slightly longer recovery. Lastly, Slow and Auto are both significantly slower and feature complex nonlinear release times, with smoother, less audible level control. Fast mode increases the speed of both the attack and release for each setting, effectively doubling the number of time constants from 6 to 12. This allows for a dramatic increase in attack as compared to the 5254 compressor’s vintage ancestor, the 2254. In addition to the Timing control, the diode bridge compressor can be further manipulated by the Ratio, Threshold, Gain, S/C HPF, S/C Insert and Link controls. The RATIO control has six selectable positions on the rotary switch and allows the user to set the slope of the compressor curve with ratios of 1.5:1 through 8:1. The S/C HPF is 12dB/octave with continuously variable frequency control from 20-250Hz.  www.rupertneve.com

Nominees Bettermaker

McDSP

Warm Audio

Bettermaker’s new Bus Compressor (£1650 ex VAT) treads a now familiar path for the Polish hardware manufacturer – digitally controlled, recallable, high quality analogue hardware. This 1U box includes front panel control of all functions, either directly or indirectly. A number of extras are provided compared to the SSL design from which this device takes inspiration. A Mix control enables parallel compression, and a HPF Sidechain is continuously variable from 20-370Hz; you can even monitor the Sidechain signal or the Insert Sidechain. A big advantage is the option to change from Peak to RMS, the latter being much smoother and more ‘invisible sounding’ than the traditional Peak mode, especially when compressing heavily. “It sounds fabulous, and glues the mix beautifully instead of pumping unnaturally” said George Shilling in his Resolution V19.3 review. “Further warmth and goo is provided with the VCA THD knob which lets you drive the THATS VCA into harmonic distortion.”

The APB-8 (MSRP $4,499) is an eight-channel version of the amazing APB-16, as reviewed in Resolution V18.4, with identical functionality and capability. Seamless ‘analogue processing as plug-in’, simple hook-up, great sounding processors, lovely saturation — or pristine clean. Combining the flexibility of software with the fidelity of premium analogue processing (in-the-box control, hardware process) — each channel can be controlled by an AAX, AU, or VST3 plug-in. APB-8 and APB-16 can be combined on the same Thunderbolt bus, in any combination, up to five units, for up to eighty channels of analogue processing. Processing options include compressors, mastering limiters, transient enhancement devices, multi-channel and multi-band applications. The APB-8 (and APB-16) now support Logic Pro X 10.4.x and later versions, and Cubase 10.x and later versions.

Since 2011 this Austin, TX company have built up a range of studio outboard based on classic designs. The Bus-Comp is an all-analogue, 2-channel, stereo VCA bus compressor based on classic SSL circuitry. An additional HPF grooms the sidechain, rooting out lows from the detector circuit. Frequencies are 30, 60, 105, 125 and 185Hz and it works a treat, preventing the low end wreaking havoc with stronger settings. Reviewed in Resolution V19.5, George Shilling said: “As with any SSL or clone it is easy to make things pump or grab in a slightly undesirable manner. However, having the lower 1.5:1 Ratio and the provision of the HPF both mitigate that in many situations.” USA-made CineMag transformers may be inserted into the signal path, driven by discrete operational amplifiers. “There is added richness to the juicy upper mids where the ear is most sensitive” Shilling enthused, “a sense of scooping out some wallowy muddiness lower down the spectrum, and a tightening up of the bass end.” Great value at £649 (inc).

www.bettermaker.eu

www.mcdsp.com

www.warmaudio.com

Bus Compressor

VIII / Winter 2020

APB-8 Analog Processing Box

Bus-Comp


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©2020 All Rights Reserved, PreSonus Audio Electronics. PreSonus is a registered trademark of PreSonus Audio Electronics, Inc. All other trademarks are property of their respective owners.

1995 - 2020 CELEBRATING 25 YEARS


E Q ( H A R D WA R E )

Winner: Bettermaker Mastering Equalizer A high-end hardware EQ combining Pultec-style equalizers with three band parametric EQ and lowpass/high-pass filters, with front panel or plug-in (AAX/ VST/AU) control. All functions are available from the front panel; the plug-in enables control from your DAW and, of course, setting storage; there is on-board storage also provided within the hardware. You can use the EQ in stereo, or as two channels: left and right, or mid plus sides. The HPF is selectable at 12dB or 24dB per octave, and there is an optional Resonance mode with a variable percentage. The Low Pass filter ranges from 2kHz-38kHz and is fixed at 12dB per octave. There is also a Pultec-style shelving attenuation band with roll-off at 5, 10 or 20kHz. The other Pultec-style bands are also present: the low band has conventional settings at 20, 30, 60 or 100Hz where you can boost and cut simultaneously for that famous curve,

and the DAW plug-in graphically displays boosts and cuts as they’re made. The High Boost band has the usual frequency choices

plus a few extra: 3, 4, 5, 6, 8, 10, 12, 16, 20 and 28kHz, along with bandwidth and boost knobs. Four parametric bands each have +/-15dB boost or cut, a huge Q range and wide overlapping frequency ranges: EQ1: 23Hz– 400Hz (bell/low shelf), EQ2: 80Hz–2kHz (bell), EQ3: 400Hz–10kHz (bell), EQ4: 1kHz–23kHz (bell/high shelf). In our review Resolution V18.1 review, George Shilling declared the sound “silky-smooth and extremely clean and musical”. Shilling continued: “The analogue circuitry is excellent, and the digital control works very smoothly. Particularly for mastering, the ability to save settings in a DAW is a bonus. Having such a comprehensive range of EQ styles in one unit is very handy and could well save you chaining units.” www.bettermaker.eu

Nominees Drawmer

Manultec

Radial

The 1974 is a true ‘parametric’ equaliser, with the four bands having fully variable frequency controls and offer cut and boost of +/-12dB’s. However, unlike EQ’s with a no bandwidth adjustment, or a simple switch, the two mid bands have completely variable filter bandwidth controls enabling the user to focus in on very narrow sections of the audio spectrum or apply a broad natural sounding filter, or, of course, anything in between the two. The £844 (inc) 1u EQ, 4-band parametric inspired by 1970s-era gear. Dual-channel potentiometers provide control over Low, Low-Mid, High-Mid and High EQ bands. The Low-Mid and High-Mid bands allow adjustment of frequency, bandwidth and boost/cut, while the Low and High bands allow adjustment of frequency, boost/cut and slope. Boost and cut range for all bands is ±12 dB. A Peak setting for the Low band adds a narrow bell shape to the 12dB per octave low-cut filter at the knee frequency just before it rolls off, providing extra low-end weight without sacrificing clarity in the lowest frequencies.

Some producer members of our nominatingpanel have esoteric taste! “The minimal, yet beautiful design offers a timeless appearance and makes this equalizer an absolute joy to use” say the boutique, Cologne-based manufacturer. The Orca is a €3805 (inc) solid state mastering EQ with all-passive tone sculpting circuit. Hand-crafted in Manultec´s custom shop in Germany, hand-wired with Vovox conductors. Low boost (shelf) is stepped, +14db at 80 Hz-100Hz, low cut (shelf) –14db at 150 Hz / 330 Hz; Focus Bandwidth control offers high boost (bell), and high boost is stepped and switchable 3-12KHz. The editor paid attention when we mentioned hand-wound inductors for treble boost. Discrete opamp make-up gain powers the output transformers. In an unusual configuration ‘to handle high levels without overloading’, the 1/2 dB steps Gain control is placed before the output amplifier.

Inductor coils again (like the Orca), but in a compact and cost-effective (MSRP $800.00 inc) package. The Q3 features four controls: top-end boost, mid-range cut, low frequency bass boost, and gain makeup to offset the signal loss that occurs when using passive equalization. Each of the three EQ bands employs a 12-position switch with a different preset curve at each position. These are carefully crafted to introduce subtle accents. Each EQ setting is augmented with a miniature toggle switch that enables you to attenuate the curve for more subtle effects. When the SHIFT switch is set to the right, the effect of each curve is gentler. When moved to the left, it is bolder. Perfect for use with the Radial Workhorse and API Lunchboxstyle racks.

www.drawmer.com

www.manultec.com

www.radialeng.com

1974

X / Winter 2020

Orca

Q3


_AudioFuse Studio The new studio reference An 18-in/20-out desktop interface like no other that includes our Creative Suite of monster plugs. Pristine preamp clarity, ultra-low noise and a staggering 119dB dynamic range. Class-leading connectivity, including USB-C, MIDI, S/PDIF and Bluetooth. Advanced monitoring, low latency performance and a tactile control panel that puts creativity first. Sometimes only the best will do.

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I N T E R FA C E ( A - D / D - A )

Winner: Antelope Audio Orion Studio Synergy Core The Orion Studio Synergy Core is a well-specified 24-bit, 192kHz Thunderbolt 3 and USB audio interface that features 12 discrete preamps with six DSP processors, plus two FPGA FX processors. More than 50 of Antelope’s Synergy Core FX, modelled on hardware studio units past and present, can be used in real-time.   “Guitars sounded clear and defined, the pres managed to capture the transients of my drum kit. There was plenty of gain for ribbon mic overheads and my favourite chain of 1073 into a Blackface 1176 for vocals was easily achievable in the virtual AFX rack” said Russell Cottier in his review of the Orion Studio Synergy Core in Resolution V19.3. Cottier found the reamp

outputs on the front panel particularly useful, enabling him to chain from the reamp output, through a Marshall Jackhammer and then back into the Orion Studio. The rear of the unit offers eight combo-sockets, two inserts and a pair of monitor outputs on TRS, with 16 line outputs on two DB25 connectors, in addition to an array of digital I/O with ADAT Toslink ports and S/PDIF connectors. The dynamic range of the mic preamps is -121dB while the A–D and D–A converters offer

124dB or 128dB respectively. If those extra dB are required, you can bypass the preamps using a ‘direct’ option. The four front–panel pres have Hi-Z input switching, while mic level, line level and phantom power options are supported on all 12 analogue inputs. When we reviewed the £2,215 (inc VAT) unit, Antelope bundled the Auto-Tune Synergy plug-in (normally €249) — which has the facility to operate in ‘near real-time’ via the Synergy Core’s DSP. While noting that the ‘Direct mode’ offers CV/Gate connectivity, Cottier concluded his review by saying he saw “a promising future for the 3rd Generation Synergy Core”. www. antelopeaudio.com

Nominees Arturia

Crane Song

PreSonus

The £799 (inc VAT) desktop unit features four ‘Discrete Pro’ preamps fed by front-mounted mic/line combi XLRs, with channels 5-8 served by rear TRS connectors, channels 5-6 include the option of RCA sockets with a built-in phono pre-amplifier. There’s a total of 18 ins/20 outs, Bluetooth functions, two pairs of analogue speaker outputs and stereo headphone output, ADAT and S/PDIF inputs as well as sampling rates up to 192kHz 24-bit. Unusually comprehensive connectivity includes Bluetooth streaming to monitors or a channel in your DAW, USB-C, ADAT, MIDI and Word Clock connectors. The included Creative Suite covers software effects and virtual instruments.

The HEDD Quantum AD/DA Converter is a 24-bit, 192kHz A/D-D/A converter featuring master word clocking, with clock outputs and a 22 Element VU digital meter. Legendary Crane Song designer Dave Hill has made it his mission over the years to continuously improve the (already amazing) clock throughout his converter products; when he re-designed his Avocet IIA he managed to achieve the subpicosecond jitter clock called ‘Quantum’, which is now available in the £3,799 (inc VAT) HEDD. The benefits are pristine imaging, depth of sound and detailed transient response. The classic HEDD DSP emulation of Triode, Pentode and Tape Emulation is also provided. Alongside the AES/EBU connections, there are now Optical I/O via Toslink connection, the HEDD will output on all three digital outputs and the analogue output at the same time. Usefully, the A/D and D/A section of the interface can be run at different sample rates — at the same time. An analogue Crane Song HEDD Dither is available and is selectable between 16 or 20-bit.

Controls from PreSonus’ FaderPort V2, together with a USB-C interface featuring two XMAX mic preamps, 24-bit 192kHz A/D-D/A converters, headphone and line outs. An eye-catching feature is the price: just under the magic $300 point in the US, and typically around £260 (inc VAT) in the UK. By default, controls are configured for PreSonus’ own excellent DAW, Studio One; for Pro Tools (HUI) the Next button is held while powering up, and then the Touch button pressed. For Logic (MCU) the Next and Mute buttons are pressed. For Windows users, Universal Control App allows management of ASIO driver settings, and enables ‘loop back’. “Amazingly good value” was our review conclusion (Resolution V19.5), with controls instantly configured for all major DAWs, and two preamps offering 80dB of clean mic/ instrument gain.

www.arturia.com

www.cranesong.com

www.PreSonus.com

Audiofuse Studio

XII / Winter 2020

HEDD Quantum

ioSTATION24c


/ Resolution Awards 2020

MICROPHONE

Winner: Schoeps CMC 1 Production sound pros must have been out in force at our ballot box, as the CMC 1 forged ahead in our Awards voting. The CMC 1 is the German mic-manufacturer’s new Colette microphone amplifier, about one-third the size and 40% lighter than the well-known CMC 6 microphone amp. Importantly, it is compatible with all components of the Schoeps Colette system and can be used in place of the CMC 6 ‘without compromise.’ “Now why, you may ask, am I excited to be given, not a whole new Schoeps microphone to review, but just a redesigned head amplifier?” asked Simon Clark our BAFTA-winning reviewer (Resolution V19.6), production sound recordist, and head of production sound at the UK’s National Film & Television School. Movie and TV sound recordists love modular microphones — if you’ve every visited a movie set, the reason is easy to see — as soon as the director calls for a wide shot, the sound

person’s job becomes a tricky task of covert mic concealment. “The only component which

needs to be on the set is the capsule itself,” Simon explains. “Head amplifiers and High Pass filters can be at the operator end of the boom pole connected by a so-called ‘active cable’.” As with everything audio, the devil is in the detail: this is where Schoeps continue to excel, having pursued a high-quality program of ‘uncompromising miniaturization’ over the last years, whilst making sure their users’ investment in capsules is validated. The CMC 1 is not only smaller and easier to conceal, with superb build quality, it also has a lower current drain (2mA from the Phantom supply important for battery operation). CMC 1 also boast an extra 4dB maximum SPL handling over the CMC 6, and (for modern movie sets) features Schoeps’ excellent Radio Frequency Interference shielding. www.schoeps.de

Nominees AEA

DPA

Josephson Engineering

Legendary ribbon mic designer Wes Dooley has done it again: a stage-proof directional ribbon mic capable of handling a quoted maximum SPL of 135dB. Our reviewer (Resolution V18) Jon Thornton: “Go as close as you dare with the high pass switched out and that massive proximity effect is on tap for stadium like snare sounds. Or switch the filter back in for something a little more natural and acoustic. The same goes for electric guitar. It pretty much sounds great straight out of the box, and playing with distance and the filter setting is mostly all that’s needed. It ‘just works’ on vocals too.” The £995 (ex VAT) mic gives great results on a wide range of sources, including ribbon sound for vocals in a live environment.

With the DPA 4560 CORE Binaural Headset Microphones you are, in effect, miking-up the outside ‘earhole’ surrounded by your pinnae. The system is an ingenious combination of a pair of 4060 CORE miniature omnidirectional microphones, mounted on two earhooks ‘borrowed’ from DPA’s 4266 Flex headset. The matched 4060s are handpicked on sensitivity within ±1.5dB. Two foam windscreens options (large/small) are supplied to offer a comfortable fit and some damping of wind noise. DPA are not the first to have come up with the idea of a binaural rig on the human head, but this is by far the highest quality system commercially available. “Superb, low-noise, realistic binaural capture which will surely become a benchmark.”

An FET largediaphragm condenser microphone with a fixed cardioid polar pattern. The £2,125 (ex VAT) Josephson C705 uses a single diaphragm cardioid capsule derived from the C715 — which reviewer Jon Thornton called “a modern classic” in his Resolution V16.7 review. The capsule uses a 5-micron gold-metallised diaphragm and boasts acoustic ports to create its cardioid polar pattern, which provides improved directionality at low frequencies. It also features the same discrete cascode input circuit and the symmetrical balanced transformerless output circuit of the 716, which is used for buffering the capsule as well as converting the impedance. As a result of its low impedance, the C705 works with a great range of preamps and input stages.

www.aearibbonmics.com

www.dpamicrophones.com

www.josephson.com

KU5A

4560 Binaural

C705

Winter 2020 / XIII


MONITORING

Winner: Genelec 4430 The Finnish studio monitor company launched the 4430 as a product to “give the AV world exceptional sound quality over IP”. It’s a two-way active installation loudspeaker. Our nominating panel noted the useful features of delivering power and sound to multiple loudspeakers over a single network cable, and controlling them remotely using software. It seems voters in our 2020 Awards agreed, with the 4430 just passing the PSI Audio A14-M v3 monitor by a narrow margin (3.2%). To be clear: no AC power required! This opens up a world of potential applications in broadcasting, movie making, and even in music recording environments outside of conventional studios. When our editor heard the sound quality and peak handling of these little Genelecs, he asked where the flux capacitor was hidden! The 4430 accepts both Dante and AES67 IP audio streams, and derives power via both PoE and PoE+ Power-over-Ethernet formats, with

the 4430’s proprietary internal power supply helping to deliver an impressive 104dB of short-term SPL, via two integral 50W Class D amplification stages feeding the woofer and tweeter. As well as receiving both power and audio-over-IP, the 4430’s single rear panel CAT connector also allows access to Genelec’s Smart IP Manager — a downloadable software

tool running on Windows 10 that allows installers to configure an almost unlimited number of rooms, zones, loudspeakers and audio channels, and includes device discovery, a versatile room equalisation tool set, system organization and status monitoring. The 4430 supports up to eight audio channels in a stream with sample rates of 32–96kHz and 16/24-bit resolution. Streams can be managed by both Dante Controller and Dante Domain Manager software, as well as legacy balanced line analogue audio. The 4430 delivers a frequency response of 45Hz–23kHz (-6 dB) via a 5-inch woofer and 3/4-inch metal dome tweeter, and the 4430’s compact, lightweight enclosure utilises Genelec’s trademark Minimum Diffraction Enclosure and Directivity Control Waveguide technologies to ensure neutral, uncoloured sound. www.genelec.com

Nominees Adam Audio

PSI Audio

Quested

Despite their affordability, the T-series range of monitors are still based around Adam’s wellregarded Accelerated Ribbon Tweeter. The £518 (per pair inc VAT) T8V utilises a Class–D bi-amp pack providing 20W RMS to the tweeter and 70W RMS to the woofer. Adam state a pair of T8Vs can produce peak SPLs of 118dB at 1m. Balanced XLR and unbalanced phono inputs are provided, with a slide switch to select between them. A pair of three-way slide switches offering -2dB, 0dB and +2dB adjustment for both the LF and HF. When we heard these at NAMM, we felt they gave a good account of themselves given their position in the market.

The new £1867.2 (per pair exc VAT) v3 model of this classic and well-respected monitor replaces the previous externally manufactured tweeter with a specially developed driver, which is manufactured entirely in PSI Audio’s own Swiss factory. While A14-M was already popular with smaller studios and mobile applications, the new version surpasses it, especially when it comes to reproducing high frequencies. Designed to work in harmony with the full range of A series monitors, the A14-M also make ideal rear speakers in a surround sound setup or can be developed into a more powerful system by adding the A225-M sub — thus protecting your investment.

The V2104 is the latest (and smallest) model in Quested’s V-Series range of active, powered, monitors. The ported, direct radiator design utilises a 4”/100mm paper cone bass driver and a 28mm /1 1/8th” soft-dome high frequency unit. Power is derived from a 2-channel class D module, chosen primarily for its sonic performance and also for its compact dimensions, keeping the cabinet size to an absolute minimum. The diminutive size of the V2104 makes it the perfect choice for a portable monitor with the special edition MiniRedz version including a shoulder carry case. The £1,700 (per pair ex VAT) V2104 can be paired with the SB8, SB10 or QSB112 subwoofers to extend its frequency range, or create several configurations from Stereo to 5.1 surround and more.

www.adam-audio.com

www.psiaudio.swiss

www.quested.com

T8V

XIV / Winter 2020

A14-M v3

V2104


/ Resolution Awards 2020

PLUG -IN

Winner: PSP InfiniStrip A modular ‘infinitely’ configurable channel strip plug-in (AAX, AU, VST, VST3, RTAS), which includes a wealth of zero-latency processing modules, providing a full processing workflow from recording to mixing. InfiniStrip features 22 processors, split across eight defined category slots: Preamps, Filters, Compressors, Equalisers, Limiters; one slot to incorporate Gate/ Expander/Ducker; the main output slot Control incorporating a fader, width balance and metering; a slot called Special includes a De-esser and De-Hummer module, and two insert slots that can be used for any module. The 9-slot rack uses a drag and drop process for custom configuration, with three alternative view modes. “After a week of mixing on various projects using InfiniStrip, I noticed a few stand-out things,” wrote Grammy-winning engineer Alan Branch in Resolution V19.4. “I found CPU usage

low even on my ageing Mac Pro, and I liked the unusual 12-bit ADC Drive control in the ADC ‘90s pre-amp. This low bit-depth nonlinear emulation reminded me of the bite you got from the sound when sampling on an old MPC, the perfect amount of crunch for some electronic drums. The filters were noticeable smooth to adjust, whilst the EQ and compressors were as I expected, varied in

style and full of character and rich saturation. A powerful feature of InifiStrip is the ability to switch modules whist the settings remain the same, so it’s easy to A-B compare the difference between the FET and Opto compressor or the RetroQ and PreQursor EQ, excellent for experimentation and a huge help in finding the sound you’re after.” PSP have already added a sample-accurate Brick Wall Limiter and a Saturator module since our review. In an era in which plug-in awesomeness frequently challenges CPUs, the Infinistrip’s economical ways will be welcomed, especially by laptop users. The zero-latency processing is a persuasive feature for production pros without FPGA-style interface processing. A deserving winner, and a bargain at $199. www.pspaudioware.net

Nominees Leapwing

Liquidsonics

sonible

Three faders (Sub, Thump, Punch) with adjustable crossover frequencies (32-56Hz, 62-104Hz, 110-196Hz) add subharmonic content. The analysis-synthesis algorithm generates clean, phase-aligned subharmonics for a subtle and musical addition to your bass that fits the original sound, without being hyped or artificial. We handed the £179 RootOne to our editor to review for Resolution V19.4. This bass-challenged tape era engineer certainly appreciated the RootOne’s subharmonic abilities. “The Harmonics channel conveys the impression of burgeoning bass on small speakers, particularly with kick drums and the like. Anything more than 50 on the Drive control really brings-on the ‘my boom-box speakers are blowing up’ effect of cone distortion,” he wrote.

A reverb dedicated to the intense acoustic demands of surround room simulation. Available in two editions, both supporting channel formats from stereo up to 7.1.6 for use with the latest Atmos bed workflows. “The reverbs are just stunning,” said reviewer Jon Thornton. “There’s a lushness to larger spaces that never really sounds synthetic, and early reflection patterns manage to avoid sounding coloured or phasey even when emulating very small spaces or ambiences. …the ability to apply scaling to nearly every parameter in every plane is eye-opening in terms of sound design and creating immersive sound.” A Professional version at $399 offers extended multi-channel editing capability.

A €129 plug-in which delivers customtailored reverb by adjusting its processing to the individual characteristics of the input material, and creates a navigable range of styles from scratch. A menu enables selection from Drums, Snare, Guitars, Keys, Vocal, Speech – then a Learn button tailors spectral and temporal settings to the recording. “It has the knack of adding just the desired amount of room — and very tuneable it is as well,” said reviewer Nigel Jopson in Resolution V19.4. He made the point that EMTs and Lexicons were designed in an era when most recordings were made in great acoustic environments, whereas modern recordings might require a different type of more realistic room. “The smart:reverb is a really modern-sounding ambience generator, with the facility to create tight and up-front rooms to bring individual instruments to the fore in a mix.”

www.leapwingaudio.com

www.liquidsonics.com

www.sonible.com

RootOne

Cinematic Rooms

smart:reverb

Winter 2020 / XV


PREAMP

Winner: Neumann V 402 The V 402 is Neumann’s first-ever stand-alone microphone preamp — although the company has created several generations of top-quality preamp modules for its mixing consoles, such as the V 476 B of the ‘80s. “All Neumann microphones, classics as well as current models, have highly unique sound characteristics”, explains Ralf Oehl, the Berlin company’s new CEO. “The V 402 provides the ideal signal path for all Neumann microphones because it is free from any colouration whatsoever. No artificial additives could make the sound of a Neumann U47 or U87 more desirable than it already is. Perfection cannot be perfected!” The V 402 is a dual channel microphone preamplifier carefully designed to maintain the sonic integrity of the original signal. Its unique transformerless circuitry amplifies the microphone signal without unwanted coloration

or sonic artefacts, such as noise and distortion. While this is also often claimed for simple preamps such as those in audio interfaces, Neumann say the V 402 is built to much higher standards. “It took an elaborate development process with extensive series of measurements and critical listening tests to create a preamplifier worthy of the Neumann name,” comments portfolio manager Sebastian Schmitz. “And we’ve

included the monitoring side, too. The V 402 establishes a fully transparent signal chain extending from the microphone all the way to the engineer’s ear. Finding the best microphone position has never been easier.” The V 402 is equipped with a ‘studio grade’ headphone amplifier ensuring monitoring quality at the recording stage. Independent volume controls for each channel enable a latency-free monitoring mix without affecting the recorded signal. It features a switchable high-pass filter and a -20dB pad which allows the V 402 to be used with high-level sources, up to a claimed 28dBu without distortion. Mic inputs maximum gain is +60dB, 10Hz to >100kHz (-3 dB@ 40dB Gain). The €2,749 (MSRP) V 402 comes in a 2U 19” rack enclosure. www.neumann.com

Nominees API

Grace Design

Millennia

In the early ‘70s Saul Walker, founder of API, designed his first microphone preamp and named it the 312. Nearly 50 years later, API have introduced a recreation of this highly sought-after classic. Used as the front end of the first API consoles, the original circuitry was housed in card cages under the front ‘knee space’ of early mixers, with only the gain pot on the control surface. The new $755 (£663) preamp module is designed to fit into API’s various rack configurations, and offers one of API’s most famous mic preamps in the popular 500 Series format. Utilising API’s 2520 op-amps and proprietary transformers, the 312 delivers the analogue warmth and unmistakable sound for which API is known.

The mk2 streamlines the company’s 2-channel microphone preamp concept by paring away functionality in favour of optimised, audiophilestyle preamp performance. While its predecessor offered converters and an M-S matrix, the m201mk2 focuses on refining the fully balanced signal path, resulting in additional dynamic range. A new power supply design provides “the lowest noise performance we have ever achieved”. Noise (referred to input) at 60dB gain, with a 150Ω source 22-22kHz is quoted as 127dB. Default mode offers 18-64dB of gain in 2dB steps, or 28-74dB in ribbon mode (which shifts the gain range up 10dB while deactivating 48V phantom power, bypassing the decoupling capacitors and optimizing the input impedance). The rear panel features two XLR outputs per channel, a handy feature enabling a separate feeds during recording. MSRP $1,895.

The Millennia HV-316 is a 16-channel remotecontrol Ethernet mic preamp designed for ‘adverse environments and mission-critical applications’ such as live touring rigs, scoring stages, location recording, and multi-room studio complexes. HV-316 uses the same high-speed transformerless design as the well-respected HV-3 series (channels matched to 0.08 dB) and can seamlessly operate on a single Ethernet network — up to 384 redundant audio channels — letting you remotely control Gain, Polarity, Pad, Mute, channel Linking, HPF, and selectable ribbon or phantom paths on every channel. You also have unlimited scene recall scripting and virtual scribble-strips. The HV316 includes a high-quality AKM A-D converter chip delivering 32-bit/192kHz resolution and >130dB of dynamic range thanks to its multi-paralleled architecture. An FETbased output stage (analogue output option) drives 300m cable runs with ease. Both battery 12VDC and 80-264VAC worldwide powering are possible. £3,999 (SSP inc VAT).

www.apiaudio.com

www.gracedesign.com

www.mil-media.com

312 Mic pre

XVI / Winter 2020

m201 mk2

HV-316


/ Resolution Awards 2020

PROCESSOR

Winner: AMS RMX16 500 Reverb Nearly 40 years since making its debut on the professional recording scene, the iconic AMS RMX16 Digital Reverberation System has been re-launched in hardware format as part of AMS-Neve’s rack-mounted 500 Series. In keeping with its predecessor, which AMS-Neve say “was the world’s first microprocessor-controlled, full-bandwidth digital reverberator”, the new RMX16 500 Digital Reverb Module is designed to be musical rather than simply implementing mathematical algorithms. Great care has also been taken to replicate the complex sonic characteristics of the original unit’s analogue and converter circuitry, which played a big part in the overall sound. The AMS RMX16 was designed by ear, with each program tuned and re-tuned to provide as wide a sweet-spot of settings as possible by means of ‘carpet graph’ parameter tables, interactively linked control by control. These design principles are key to its longevity and have been carried forward to the

new RMX16 500 to ensure that it replicates the original’s performance. Extra features on the 500 series version include a Mix control for a percentage value of wet to dry. And there are now memory registers for saving 100 presets within. The other difference is that this new version uses a much more powerful processor, and far better

converters. “This results in incredibly quiet operation” said George Shilling in his Resolution V19.1 review, “the original used to hiss and whine in a manner that would not now be tolerated by younger engineers!” “The Room settings were good for short, thickening settings to give instruments a small amount of interesting space, and the Halls were sometimes awesome for huge and epic reverb. But the most interesting setting was Nonlin2, a richly grainy pseudogated reverb effect that shouted ‘1980s’. It still sounds great — thick, and always interesting. Similarly, the two Reverse programs have a great fade-in effect if you want something characterful.” The RMX-16 500 is £995 (ex-VAT). “This reissue seems to sound smoother and more hi-fi than the original,” and is “great to have as hardware,” Shilling concluded. www.ams-neve.com

Nominees CEDAR Audio

Eventide

Lawo

A Dante-enabled hardware unit that offers eight simultaneous channels of noise-reduction, allowing you to reduce the sound of air conditioners, traffic, rain, fan noise, and more in real time. It can also improve unflattering acoustic environments, poor mic positioning, and excessive reverberation. This is the first CEDAR with eight channels of Dante over Ethernet ports, providing industry-standard networking with minimal latency. The £4,900 (ex VAT) unit also gives eight channels of AES I/O over DB25 connections. A browser-based remote control is available. “I tried, traffic, crowd noise, hums, buzzes and even staccato transients like birdsong — all of which were sonically eaten by the black magic inside the box,” said Simon Clark in his Resolution V19.3 review. He summarised: “Extremely simple interface; lack of colouration despite powerful real-time noise reduction; Dante integration; 12v DC and mains option; insignificant latency.”

Blackhole reverb sound is not of this Earth. The Blackhole is based on one of the more extreme algorithms in the Eventide cosmos, found in rackmounted units, as a plug-in (review, Resolution V17.6), and also on the Space and H9 pedals. Blackhole features modulation built into the reverb structure itself: this modulation can be used to smooth out the rough edges of the most extreme settings and offers unique tone-shaping ability. ‘Infinite mode’ continuously layers new sound on top of a suspended reverb while ‘Freeze mode’ holds the effect in stasis, allowing musicians to play over the reverb tail. The Freeze Footswitch allows instant access to this feature. Blackhole can load as many as 127 presets via MIDI and they are also accessible in the preset list on the Eventide Device Manager (EDM).

Lawo have a new software package for its versatile Power Core AoIP mixing engine and I/O node: Power Core MAX. ‘MAX’ — short for Multiple Access — makes it possible for a single Power Core device to be the mixing engine for multiple on-air mixing consoles. “Power Core MAX lets you unlock all those resources to power two, three or even four small mixing surfaces. That’s a capability that anyone who wants to maximise their equipment budget will immediately appreciate,” says Lawo Radio Marketing Specialist Clark Novak. Power Core is an AoIP workhorse. Dual-redundant front-panel AES67 ports can accommodate up to 128 streams with a total of 256 audio channels. Four front-panel MADI ports can handle 64 MADI channels each, for a total of 256 channels, or 128 MADI channels in dual-redundant mode. Thanks to its abundant I/O capacity, Power Core can be used as a gateway between legacy audio formats and standards-based IP networks.

www.cedaraudio.com

www.eventideaudio.com

www.lawo.com

DNS 8D

Blackhole pedal

Power Core MAX

Winter 2020 / XVII


RECORDER

Winner: Zoom H8 The H8 an ambitious but cost effective (£506 inc VAT) handheld recorder. It comes with a stereo X-Y pair of microphones which can be exchanged with modular mics from a range of newly-developed capsules and input modules. These include a four-capsule Ambisonic array, which the H8 can decode into a variety of stereo and multi-channel formats, and the ‘Expander Capsule’, which provides four additional mic XLR inputs. There’s also the MSH-6 mic for mid-side recording, the SGH-6 ‘hyperdirectional’ shotgun, and an unusual mid-side/shotgun. The SSH-6 mid-side stereo shotgun capsule includes a super-directional microphone for picking up sound in the centre, as well as a bi-directional side mic for picking up sounds from the left and right. This allows you to record a fully monocompatible stereo image — ideal for video projects. For example, you can capture dialogue with the centre mic, and then mix in the desired amount of environmental sound from the side mic directly on your Zoom recorder, or in postproduction. The H8’s recording sources connect to the octagonal centre section, and comprise two combi inputs for mic/ line/instrument signals, plus four XLR mic inputs. The lower section is dominated by a large colour touchscreen, while

the bottom edges house the SD-card slot, a USB port, headphone and line output mini-jacks. The touchscreen allows access to recording controls and software features, and Zoom have developed three different ‘apps’ for it, each with a different use in mind. The Podcast app lets you load up to 13 sounds into the H8, and trigger them using a touchscreen soundboard; the Music app focuses on recording and monitoring, enabling overdubs with a full mixer, with control over compression, EQ and effects; the Field app is tailored for sound designers and location recordists, and features high-resolution metering to help ensure recordings never clip. Other features include remote control via an iOS app, a suite of guitar amp and effects modelling processors, and the ability to work as a multi-channel computer audio interface over USB. The Zoom H8 comes with Cubase LE and Steinberg’s WaveLab Cast software. WaveLab Cast helps finalise projects with a range of comprehensive editing and mixing tools. WaveLab Cast connects directly with podcast directories, such as Spreaker or Podbean. You can upload episodes quickly, plus directly create and publish RSS feeds. www.zoomcorp.com

Nominees iZotope

Lectrosonics

Sound Devices

Portable, one-touch recording — the £359 Spire Studio is a portable recorder featuring built-in WiFi. Aimed at musicians, it enables easy record, mix and editing on the move. Spire sets up compression and gain for recording with a one-button soundcheck, once a track is laid down, it automatically arms the next one. The integrated omni mic is supplemented by two combi-XLRs with 48V phantom. Spire includes a variety of iZotope effects ranging from reverb and delays to amp models. It provides zero-latency monitoring and the ability to sync tracks between hardware and the (free to download) Spire software. Crucially, it’s easy to export audio to Pro Tools, and Logic, or upload to storage platforms like iCloud, Dropbox, and Google Drive.

A compact, dual-channel digital wireless receiver offering double the channels of the UCR411a in the same sized package — and on-board recording. Compatible with mono transmitters from the D-Squared line (DBu, DHu, DPR) and capable of decoding signals from Lectrosonics’ stereo transmitters like the DCHT and M2R. The unit is backwardscompatible with Digital Hybrid wireless transmitters. Two TA3M audio output jacks can be independently configured as mic or linelevel analogue outputs or as two-channel AES3 digital outputs. The on-board SDHC recorder is capable of recording .WAV (BWF) files while the transmitter is sending wireless signal, making it a truly useful backup option. The DCR822 is available worldwide from Lectrosonics authorised UHF dealers, MSRP: $4,740

A portable multitrack recorder, mixer and USB interface with internal LTC timecode generator with output, designed for field recordists and filmmakers, and offering 32-bit float recording to removable SD, SDHC, or SDXC media cards. Up to five audio tracks can be recorded at 44.1-192kHz to broadcast quality .WAV file. The operator has the ability to adjust ratio, release and threshold parameters of the internal limiters when using in 16 or 24-bit mode. Other software features include a 10 seconds pre-roll buffer and auto-copy to USB drive function. The £830 (ex VAT) MixPre is powered via USB connection, L-mount lithium-ion batteries, 8x AA alkaline batteries via the included sled, or included AC power adapter.

www.izotope.com

www.lectrosonics.com

www.sounddevices.com

Spire Studio

XVIII / Winter 2020

DCR 822

MixPre-6 II


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REVEALED

/ Cameron Craig: mixing Melua / Tommaso Colliva: Zoom music video / Remote production special

/ SSL Origin mixer: a 4k core / CEDAR DNS 8D: Dante de-noiser / AMS Neve 1073® OPX

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/ Future: VR concert production / Neve 1073®: a legend turns 50 / Pop Music Production – Phil Harding

/ PreSonus Studio One 5: scores big / CEDAR Cambridge v13: powerful new tools / sonible smart:reverb

/ Segun Akinola: Doctor Who composer / Robert Sanderson: forensic audio / Martin Garrix STMPD recording studios

/ Tall Pine Records: API Legacy in Poland / Dean Street Studios: Atmos with PMC / Resolution Award nominations

/ PreSonus ioStation 24c: desktop control / Liquidsonics Cinematic Rooms: lush space / iZotope Neoverb

/ Source-Live Low Latency: remote heaven / Emika: solo piano to electronic symphony / Making Waves: The Art of Cinematic Sound

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03/11/2020 11:59


Technology

Genelec GLM 4 Managing monitors — THOMAS LUND explains the science of listening

G

LM 4 is the fourth generation of the Genelec Loudspeaker Manager software, designed to calibrate and control professional monitoring systems consisting of a wide range of Genelec Smart Active Monitors (SAM) and subwoofers. Despite tightly controlled directivity and point source design, two company hallmarks, it is still important to minimise the unwanted acoustic influences of a listening room, following placement of the monitors. GLM calibration ensures reference listening conditions to be met, and therefore good translation to other rooms, including clubs, cinemas, headphones etc. Based on continuous research and data gathering from thousands of studios around the world, each generation of the software has seen improvements and new features added, version 4 being no exception. The evolutional concept also reflects Genelec’s goal of providing a long life for its designs, facilitating improvements of monitors and subwoofers that have been ten years in the field or even more, for instance 8200 and 7200 series models. GLM 4 comes as another free upgrade, and the graphical user interface has had a complete makeover by renown Finnish designer, Harri Koskinen, who is also the mastermind behind the physical appearance of most Genelec monitors. His Nordic roots shine through in the UI too, XX / Winter 2020

Active

Smart Active

8010 8020 8030 8040 8050

8320 8330 8340 8350

Subs

Subs & Woofer Systems

7040 7050

7350 7360 7370

8430 1032C S360

7380 7382

Previous

8331 8341 8351 8361

1234 1235 1236 1237 1238

W371

8130 8240 8250 8260

7260 7270 7271

/ List of current Genelec monitors and types — new and previous — compatible with GLM 4


revolving around an element representing unlimited, natural growth: the hexagon. The seamless growth potential is an elegant feature of GLM 4, because no downsides or hurdles are introduced as more channels are added to an existing setup, regardless if it happens one step at a time. There are no sudden penalties when going from single stereo to multiple stereo, when adding one or more subwoofers; or even when building huge, immersive monitoring systems. GLM 4 has also seen monitor control functionality boosted with new touch screen compatibility, solo and mutes have been enhanced; and immersive systems consisting of more than 80 devices are supported. This takes the application into new territory as an indispensible monitor control companion for DAWs, where processing and physical output resources need not be tied-up on monitoring, adding to an all-round enhanced user experience. GLM 4 runs on Windows and Mac including macOS 10.15 Catalina. It enables calibration on the local computer as well as cloud-based calibration, with an acoustical help service and guidance based on comprehensive, actual room data.

Are we receivers of sensory information from the environment, or are we actively collecting it? which time influences what we are even capable of hearing. Some ways are comparable to the capabilities needed to understand a new language, others relate to how quickly fatigue develops, but the primary reason is our modest

perceptual bandwidth. We simply cannot take in sensory stimuli comprehensively nearly as quickly as we tend to believe. In order for accurate listening not to become too time consuming, variables consequently

Slow listening

Are we receivers of sensory information from the environment, or are we actively collecting it? Numerous recent trials have concluded the latter to be the case. Human perception is primarily based on previous experience and sensory verification, known as ‘active sensing’. Considering hearing, the brain is a highly active participant, not only in the decoding of minute temporal information, but also as the main element of a sense relying heavily on internal feedback and tuning. Hearing therefore doesn’t only have nerve impulses going from the inner ear to the brain, it also has a substantial number of efferent nerve fibres with traffic going in the opposite direction. Depending on behavioural goals, training, experience and expectations, efferent nerve fibres constantly send information back from the brain to the middle and inner ears, adjusting the reception system itself. In principle the same as vision with its saccadic eye movements: 99% of our visual field primarily detects movement, and only a tiny fraction, the fovea, registers perspective, colour and spatial details. The brain therefore regularly needs to move the eyes for a closer inspection of visual details. In contrast, the brain’s tuning of the inner ears over a range of 80dB (!) is hidden, but there is also an overt component to acute listening: head and body movement. In either case, what we hear and how we move largely depends on what has personally been experienced before. Furthermore, accurate listening takes time, as various auditory events and qualities need scrutinizing. We recently introduced the term ‘slow listening’, as a reminder of the many ways in Winter 2020 / XXI


Genelec’s DNA is to strive for neutral reproduction as the pinnacle of monitoring have to be kept at a minimum. Listeners should either use a room and equipment they know intimately, or they should have plenty of time to get to know an acoustic environment before any subjective tests or judgements are performed. Based on a limited perceptual bandwidth and eight hours of dedicated listening per day, getting to know a room and equipment in any detail takes at least a week, but assuming years would be safer. Great mastering engineers therefore make sure they have time on their side in many ways, so the content becomes the only variable. Bob Ludwig, for instance, even tries to avoid listening at different sound levels because of the time it takes to mentally re-calibrate. Instead, clients are invited to use headphones in case they wish to listen louder.

Headphone compatibility

Using headphones for listening is an excellent method to check a recording for artefacts, and for instance to be able to perform any work at XXII / Winter 2020

/ 3-way loudspeakers exhibit “Christmas tree” behaviour, as each driver becomes directive with frequency

all under noisy conditions. However, the main purpose of monitoring is to evaluate audio in a neutral way, and to ensure good translation to other reproduction conditions. Standard headphones don’t fulfil those requirements because they exclude the influential external ear from our auditory system, thereby breaking the link to natural listening that we have acquired over a lifetime. Traditional headphones cannot externalise sound, and they also inhibit the other essential facilitator of critical listening detailed earlier: Active sensing. Even when passing around the same set of

headphones, what you hear is therefore different from what the next person hears, so a meaningful and detailed discussion about sound quality can really only happen between people listening and moving in the same room. The most coherent way of optimising translation between headphones, if that is expected to be the primary method of playback, instead would be to rely on GLM 4 calibrated ultra-nearfield monitors, ideally based on a Genelec Ones stereo system, e.g. at 50cm listening distance [I’m waiting to review the Genelec ‘mix chair’ with 50cm head-height monitor mounts — Ed].


/ Technology

Reference monitoring

It has always been in Genelec’s DNA to strive for neutral reproduction as the pinnacle of monitoring, rather than aiming just at pleasing a listener. From the earliest, pioneering active designs of the ‘80s right through to today, the Genelec monitoring line has therefore shown a constant evolution towards neutrality in more and more dimensions, like frequency response, directivity, time, phase, immersion and envelopment. Taking directivity, for instance, it is unreasonable to impose the same, arbitrary directivity on all sources of a mix, thereby exciting the room in biased and unpredictable ways, unless the listener is situated in an anechoic room or in a completely diffused one. However, this is what most loudspeakers do, so a 3-way model generally exhibits ‘Christmas tree behaviour in both planes, as each driver becomes more directive with frequency, until the next cross-over point is reached. Taking another evolutional step, drivers also need to be at the same location in order not to colour off-axis sound at the cross-over points, or to make listener head movements produce ambiguous results. Genelec The Ones therefore combine those two important characteristics: superbly controlled directivity and uncompromised point source radiation. GLM 4 is the binding element that intimately knows each monitor in a system as well as the room, and balances all the dimensions mentioned from stereo to immersive, but without forcing a particular sound on the user.

Personalised monitoring

With its professional auto-calibration function, GLM 4 quickly establishes neutral monitoring conditions for a given system, room and listening position, or even for an unlimited number of listening positions. However, a primary requirement, such as a flat frequency response of perceived direct sound, includes

both objective and subjective factors. Adjustment for monitor and listener placement can reduce uncertainty about the apparent frequency response by 15-20dB, while compensation for listening level reduces

GLM 4 establishes neutral monitoring for a given system, room and listening position

uncertainty about the perceived frequency response by 10-15dB. Leaving both unchecked, on the other hand, leads to an uncertainty of the perceived frequency response by more than 30dB, even from an excellent monitor, pre-calibrated for a flat frequency response in an anechoic room. GLM 4 easily takes both ambiguities out of monitoring, considering your specific setting; and then the final spectral balance can be freely tweaked and stored to fit a certain application, or your preferred listening level. For instance, if working with dialogue, rolling off a few dB above 6-10kHz can improve speech intelligibility of the content; or, if you are typically listening below 80dB SPL, boosting the lowest octaves enables you to perceive very low frequency components or artefacts, that would otherwise be below the threshold of hearing. In case you are new to pro audio, and on the verge of appreciating the benefits of calibrated, neutral audio monitoring, learning to work against a flat frequency response is recommended. However, if you have long-time experience using un-calibrated monitors, the transition can be made easier by applying system-wide broad GLM 4 tweaks after auto-calibration, such as a full frequency range tilt. With a broad tilt, you don’t get distracted by a spectral balance that feels ‘off’, while still being able to take advantage of a monitoring frequency response without misleading bumps or dips. In conclusion, reliable, coherent monitoring is an essential foundation in professional audio, regardless of whether you are working in music, broadcast, film or gaming, because listening is formative. Therefore, you are not just creating the best possible program or music track, you are also investing time in solidifying or eroding a personal sensory baseline, whenever a room and monitoring system is being used. GLM 4 is the guarantee that your time is being productively spent on both counts.

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Craft

The detection of fakes is going to get more and more difficult also working on a program named SemaFor, whose point is to identify semantic inadequacies in deepfakes. While other areas of professional audio are in a state of flux, one field is experiencing rapid growth with the production of more and more source material every day — whether ‘deepfake’, surveillance, reportage or law enforcement. This is audio forensics, the use of advanced audio analysis and filtering to try to extract voices and meaning from badly-captured and noisy recordings, as well as methods of validating and presenting the resulting evidence to courts and other bodies. Strangely, there’s currently no formal qualification in this field. If your aim is to become ‘CSI: Audio’, what can you do? The University of Colorado in Denver is trying to rectify this missing training: sign up for a Master of Science in Recording Arts and you can focus on Media Forensics in the MSRA-MF degree programme offered by their National Center for Media Forensics (NCMF). The intention is to take students from a wide range of disciplines and prepare them for careers in the fields of audio and video forensics, as well as other areas of hi-tech crime fighting. We caught up with one of the NCMF team, Cole Whitecotton, to discover what it takes to be and audio sleuth.

Cole Whitecotton Deepfakes and media forensics — NIGEL JOPSON discovers CSI: Audio

A

report from Deeptrace, the Netherlands-based cyber security group, identified 7,964 deepfake online videos at the start of 2019. After nine months, the figure nearly doubled to 14,678, and has been growing since. In March 2019 it emerged criminals had used artificial intelligence-based software to impersonate a chief executive’s voice, ordering a fraudulent transfer of €220,000. The CEO of a UK-based energy firm thought he was speaking on the phone with the boss of his firm’s German parent company, who asked him to send the funds to a XXIV / Winter 2020

Hungarian supplier (according to the company’s insurance firm, Euler Hermes Group SA). In April 2020, Extinction Rebellion (XR) activists released a deepfake video of the Belgian Prime Minister Shophie Wilmès making a speech linking Covid-19 to the climate crisis. “There is a broad attack surface here — not just military and political but also insurance, law enforcement and commerce,” says Matt Turek, programme manager for MediFor, a media forensics research program led by the Defense Advanced Research Projects Agency (DARPA), part of the US defence department. DARPA is

How did you become involved with the NCMF University of Denver, Colorado? I’m an alumnus of the programme, having graduated from it with a Master’s degree three years ago. At the time, the Center was doing work for DARPA — the US Defense Advanced Research Projects Agency — and I became involved right at the beginning of the project. DARPA is the organisation responsible for the development of emerging technologies for use by the US military and, when I completed my Masters, the Center opened up a position that combined working on the project together with classroom support and online teaching. Is this part of the initiative to identify ‘deep fake’ audio? We were one team out of many working on detecting deep fakes. Our job was to generate fake audio and video materials that some of the other teams then tested to try to detect them. One of these was using straight-up machine learning AI, and others were working with hybrid versions that were a mix of learned and trained algorithms along with traditional


It’s an arms race and I don’t think it’s ever going to go away methods like green screen, rotoscoping, and so on. We had meetings at least twice a year and met a lot of people ranging from big name universities, to the FBI, to companies like Honeywell, that are involved in AI. The project is moving to the next level now, to what they’re calling Semafor, or semantic forensics. I think that there’s going to be even more research into deep audio fakes during this stage. Is finding fake audio harder than detecting a fake image? Right now, not so much, but I think it will be. Companies like Lyrebird have made a lot of progress towards imitation and generating new, fake voices. Adobe also had something that they released as a beta version a while back [Adobe VoCo, Resolution V16.1] but after a couple of months they decided to pull it, because it seemed dangerous. All of these companies have been working to remove the

robotic sound that for decades has been the tell-tale of a fake voice and, just like there’s thispersondoesnotexist.com I think that there’s going to be thisvoicedoesnotexist.com or something similar soon. The detection of fakes is going to get more and more difficult.

Moving away from the DARPA project, can you tell us about NCMF’s courses? There are two areas of teaching audio forensics at NCMF; there’s the Masters programme and there are the training courses for law enforcement. In our new classroom, we have

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racks of equipment at the back of the room, and there’s a station back there that allows you to use the various systems and see them working. In my day, we had a sort of offshoot — a closet really — where we had all of the audio equipment, so we didn’t get to spend a lot of time with it. But now that it’s in the room it’s easy to fire it up to show it to the whole class. Unfortunately, we can have anywhere from eight to ten students in a class, so not everyone has the chance to use the systems but, after the demonstration, the students and trainees can start to experiment and learn what they can do. How do you demonstrate audio forensic equipment to a class of students? For example, our former Associate Director Jeff Smith has a good example of CEDAR processing that he loves to demonstrate during our three-day course on forensic audio analysis and enhancement that’s geared toward people working in forensic labs. He talks with loud music in the foreground to simulate a bar, and then uses reference audio from a CD to remove the music in real-time using the system’s Time Align and Cross Channel Adaptive Filter modules. A lot of times that we do this, no-one even hears that there is somebody talking in the original… and then we remove the music. It’s a great example of ‘here is something that sounds like nothing’ and then finding the meaning in it. The biggest reaction for people who don’t really have a lot of audio background is: “I didn’t know that was possible… I see stuff like this on TV and movies but I always assumed it was Hollywood magic!” We run into that a lot.

XXVI / Winter 2020

/ Jeff M. Smith Associate Director NCMF (and Chair of the AES Technical Committee on Audio Forensics)

Huge amounts of forensic data are being generated You’ve mentioned the CEDAR Cambridge Forensic System. Can you tell us more about your use of that? The aspect I like most about CEDAR is the real-time stuff. It’s almost like the difference between a node-based system and a non-linear editor. I love how you can connect this process to that process, and how the UI shows you a visual representation of what audio is coming in, what’s going out, and what each process is doing. Another feature that Jeff really likes is how you can create markers in modules like the spectrum analyser and then transfer them to other processes like Debuzz, or the EQs, to create the right filters. We also love to use CEDAR for background images. It looks so cool that anytime we do any classroom pictures we have its UI up in the background. Because you can have a bunch of different windows open for a single project, it’s also how CSI looks. But that brings up another point. I’m sure that a lot of people get into this field because of shows like CSI, and their expectations can be way too high because of that!

Is operating audio equipment part of the course? We’re not here to teach our students specifically how to use CEDAR. We have many different audio and video systems and this is important, because each law enforcement agency has different tools from the next. So when the students come to us they can see those they know right next to others that they don’t. Every tool does something slightly different, and everybody has its own way of working. The Masters programme is designed to make sure the students have a broad foundation that they can then build on. Enhancement is one part of this, audio authentication is another, understanding how audio is generated is another, and so on. Then we add the things you need to know if you’re going to become a forensic expert — things like report writing, testimony and how to present what you did. So the focus is more on understanding the theory and workflow? Anybody can push buttons, but you have to understand what the buttons do, when you would want them to do it, and why. That’s all part of the course, preparing the attendees for working in different fields such as an FBI lab, or in local law enforcement, or in private companies like Target or Walmart. The first semester concentrates on legal stuff - the federal rules of evidence and what it means to


/ Craft

be an expert witness. Then we have semesters that concentrate on audio, or on video, or on MatLab or Python scripting, and so on. We’ve also had people who have come here to help them develop their own forensic tools, but they don’t necessarily have the computer science and coding skills necessary. If they did, they would also need a certain amount of luck… having the right idea at the right time. I can’t remember the number of times when people have said to me: “oh, that idea was obvious, I could have done that” — but they didn’t. So we ask them: “The person who did that probably had no more information than you, so why didn’t you think of that idea first?” It’s something that’s really worth thinking about. NCMF occasionally carries out fieldwork, and we noticed that you put out a press release about a counter-terrorism case that the Center worked on a while back. Can you tell us anything about that? Jeff and Catalin (NCMF Director, Catalin Grigoras) are rock stars of the US forensic world, really well known and well regarded, so people come to us to ask for help and we sometimes work pro-bono with local police and other agencies. But I can’t tell you anything more than that. What I can tell you is this

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— huge amounts of forensic data are being generated. The amount keeps growing and growing and growing, and that’s the biggest problem: dealing with the immense amount of digital data by cataloguing and categorising, as

well as making sure that you’re not missing relevant information and not handing over irrelevant information. It’s just going to get worse as time goes on. It’s an arms race and I don’t think it’s ever going to go away.

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Resolution Awards 2020  

Resolution Awards 2020: Recognising quality and innovation in professional audio, and the products highlighted here have been judged to be o...

Resolution Awards 2020  

Resolution Awards 2020: Recognising quality and innovation in professional audio, and the products highlighted here have been judged to be o...