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TABLE OF CONTENTS
Picture Perfect: Soundfirm’s Dolby Atmos Room & 15 Years of Post
Inside EAW’s ANYA
Mix Masters: Tom Elmhirst on Beck’s Morning Phase
JoeCo MADI Blackbox & Direct.Out Andiamo
Korg Kross Synthesizer
PC Audio Last Word: Gotye & Franc Tetaz
Nektar Panorama Keyboard Controller
Prismsound Titan Audio Interface
GENERAL NEWS PRESONUS LAUNCHES THE V2 When something is given a “major redesign” it’s only normal to assume the original was some kind of lemon. However, lots of other possibilities abound — a main component supplier might have gone bust for example. Heaven forbid, the company might have simply decided they can do it better. Crazy talk? Actually, Presonus is constantly listening to customer feedback and regularly upgrades products. Presonus is now shipping the new Monitor Station V2, a completely revised design of its desktop studio-monitor control centre. Monitor Station V2 offers management of multiple audio sources and up to three sets of monitor speakers. Features include four stereo inputs, being two pairs of balanced 6.5mm TRS and one pair of unbalanced RCA Aux inputs that are gain control-managed with a source-select switch. A 3.5mm TRS unbalanced input is summed with the RCA Aux inputs. New to V2 is a S/PDIF stereo input supporting 44.1, 48, 88.2, and 96k digital audio — a button switches between the Aux and S/PDIF inputs. If multiple monitoring isn’t really your thing, note that the V2 has four individual headphone outputs, each with a choice of two signal sources or the Aux. That might make the V2 temptingly dual purpose. National Audio Systems: 1800 441 440 or email@example.com
A REMINDER FROM GIBSON Is this a clever idea or taking things too far? Gibson has come up with the Memory Cable, a guitar cable with a built-in digital recorder for capturing those moments of inspiration when re-plugging into another device isn’t practical or possible. The Gibson Memory Cable records in 44.1k, 16-bit wave files, so its application isn’t restricted to guitars — anything that uses an unbalanced TS cable can be recorded. With the supplied 4GB MicroSD card you’ll get around 13 hours of recording and it has an automatic mode that kicks the recorder into life only when you’re playing. The MicroSD system
allows you to transfer your inspired genius to a DAW. Power is supplied by a single AA battery that will last eight hours. A secondary LR44 battery looks after date stamps and such for a year. The Gibson Memory Cable appears to have a dedicated in and out, which means you’ll have to use it with the recorder hanging off your instrument — it can’t be reversed and have the bulky bit at your amplifier end. If either, or both, batteries run out of juice the Memory Cable still works as a guitar cable. Australian Musical Imports: (03) 8696 4600 or www.gibsonami.com
TASCAM CLOCKS IN Among the glitz and glamour of new DAW systems, audio interfaces, studio microphones and all the other fun stuff, some of the workhorse devices don’t get a lot of attention. Tascam has recently announced three new clock generators, the CG-2000, CG-1800 and CG-1000 models, each was specifically designed for a particular market segment and, of course, different budgets. Equipped with high-precision internal OCXO (Oven Controlled Xtal Oscillator) clocks, each unit promises users a much more accurate clock system compared to rubidium or GPS 10MHz clocks, plus improved jitter management
and glitch-free relocking. Top of the menu is the CG2000, a video sync/master clock generator designed for broadcast and post-production facilities. Two power supplies protect the system from any power surge interruptions. The CG-1800 is aimed at post production and supports NTSC; PAL; S/PDIF for audio and HD Tri-level for video, word and AES 3/11. The CG-1000 is a master clock generator for mere recording studios, sound engineers and professional musicians in mind. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
ANOTHER GEM FROM FOCUSRITE Let’s hope Focusrite doesn’t cause too much headscratching by telling users the new Saffire Pro 26 connects “seamlessly” to any Thunderbolt port — via a Firewire to Thunderbolt adapter that isn’t included in the box. Back to the shop you go… Otherwise the feature of working happily with both protocols is good, future-proofing the Saffire Pro 26 against the vagaries of computer manufacturers chopping or changing preferred connections (they wouldn’t do that, would they?). It’s a Firewire 800 connection (yes, you do get a cable) so either way — and not necessarily using an Apple machine — you’re getting
a heap of bandwidth. The Saffire Pro 26 offers an 18-in and 8-out matrix via ADAT, S/PDIF and analogue connectors of which only four are fitted with Focusrite preamps. Saffire PRO 26 ships with a decent software bundle. Included is Ableton Live Lite, Focusrite’s professional Midnight and Scarlett plug-in suites, Novation BassStation virtual synthesiser and 1GB of Loopmasters sample content. You also get Saffire Mix Control which gives you onscreen, comprehensive signal routing for your DAW. Electric Factory (ELFA): 03 9474 1000 or www.elfa.com.au
STRUM UP A STORM Between the rack extension name and marketing blurb you might get a little confused. Propellerhead’s new A-List Acoustic Guitarist plug-in is, in fact, an acoustic guitar instrument and references to “rhythm guitar” are still talking about the same thing. A-List Acoustic Guitarist promises Reason users a fast, easy way to create professional rhythm guitar tracks by including over 600 rhythms spread across more than 50 styles that can be played and combined in real time. Single MIDI notes trigger chords and progressions, while adding additional notes allows users to explore alternate voicings and patterns. With immediate hands-on
control over chords, strumming patterns, tonal parameters and more, non-guitar playing musicians can dial in a perfect performance thanks to the underlying samples or spend time experimenting with endless variations that probably no selfrespecting guitarist would ask their fingers to attempt anyway. 13 chord types cover all musical complexities. We can understand replacing those pesky, loud solo guitarists, but hey — when did an acoustic player ever harm anyone? Electric Factory: (03) 9474 1000 or www.elfa.com.au
VMR SLATED FOR RELEASE SOON Slate Digital is never shy to talk up its products. In this case it’s the forthcoming Virtual Mix Rack which, as you can see, is a kind of 500 series plug-in rack and the introductory processors — two classic EQs and a pair of likewise classic compressors — promise to be “indistinguishable from the originals”. Slate has taken over two years to develop and model these effects in the pursuit of perfection and will happily encourage any software-vs-hardware comparisons, confident it has delivered the goods. Like similar products it seems the VMR itself will be
a freebie and come with a single processor to whet your appetite (that’s how it will be launched anyway). From there you buy additional modules and Slate says that thanks to its designs and algorithms you’ll be able to run “tons” of processors with no latency and minimal stress for your CPU. Cost is going to be low too with that first bundle discounted by a hundred bucks to US$149 and subsequent releases affordably priced. Final programming and tidying up should see the VMR released soon. Time enough for some virtual touch-up paint on the FG-401 as well?
ONE’S VERY OWN MICROPHONE It wasn’t hard to see this coming. For some time now Waves has been releasing plug-ins based on equipment used at Abbey Road Studios, both in the past and present — and some from the dustiest corners indeed. Now you can buy the Abbey Road Collection which includes the recent Reel ADT effect, the J37 Tape Emulation, R56 Passive EQ, a chunk of legendary console with the REDD and finally the King’s Microphones which is exactly that — emulations of three microphones tailored and used exclusively for speeches recorded by
three reigning monarchs: King George V, George VI and Lizzy, who we’ll point out is technically a Queen. Makes you think, if Chuck ever gets in the Big Chair what microphone will he get? Probably a USB podcaster. But back on subject, the Abbey Road Collection is available in Native or Soundgrid formats and, as always with Waves, represents a significant saving over buying the plug-ins individually. Sound & Music: (03) 9555 8081 or www.sound-music.com
NEVE 1073 FOR UAD UNISON If there was ever a magic word in this business it would probably be ‘Neve’, and the Grand Master Wizard undoubtedly Rupert himself. One particularly potent mix Rupert brewed up was the 1073 Channel Amplifier in the 1970s and it still represents, perhaps more than anything else the company has since done, that much sought-after Neve sound. AMS Neve still manufactures the 1073 as a hardware module and as a 500 series device. Now you have another option from UAD, which has been busy releasing Unison plug-ins, which digitally turn the stock Apollo preamps into desirable models of classic gear. So, of course, the Neve 1073 was always going to be high on UAD’s must-Unison list. There are many, many attempts at cloning the 1073 out in the world wide webbernet (the bundle includes UAD’s previous 1073 and 1073SE Legacy plug-ins), but UAD are claiming this plug-in as the only authentic, end-to-end circuit emulation of the genuine thing and with Neve being properly involved by providing the license it’s as fair a call as you’ll get. Aside from providing presets by a host of big-name engineers, plus the usual benefits of working with plug-ins, UAD has resisted tweaking the original design with any digital extras. Versions are available for UAD-2 hardware and Apollo interfaces using that Unison technology. Sound & Music: (03) 9555 8081 or firstname.lastname@example.org
NOT A SHRED OF NOISE This latest Stagebug from Radial Engineering might interest guitar players and studio owners alike. The SB-15 Tailbone is designed to sit at the beginning of the guitar-to-amp signal chain and drive multiple pedals to get rid of the noise that often plagues high impedance circuits — no, not any <Stairway To Heaven> riffs. The Tailbone has a standard hi-Z instrument input feeding the signal through Radial’s Dragster load correction circuit to replicate the tone and feel as if connected directly to the amplifier. It’s then buffered via a 100% discrete, class-A unity gain amplifier. Unlike most similar devices that employ ICs to buffer the signal, the discrete design reduces the need
FIREBIRD TAKES OFF Acoustic Technologies has released a new compact line array system called Firebird. The components are the CLA600 Composite Line Array and the CLA LF6000 Sub Bass Enclosure. The CLA600 is a two-way cabinet with a Bass/Mid design of four 8-inch drivers and a dual-element Isophasic HF Aperture. Total power handling is 2960W — Acoustic Technologies could have said 3K watts and no one would quibble, but the accurate maths is appreciated. The dual 18-inch transducer CLA LF6000 Sub Bass System can handle 6,000W program with a frequency response from 32Hz to 200Hz. A Firebird system comprising two CLA600 and one CLA LF6000 per side can easily cater for audiences in excess of 1000 people. The cabinets are finished in black AcoustiCoate polymer with powder-coated grilles. Acoustic Technologies is offering turn-key Firebird CLA600 + CLA LF6000 systems complete with Blackbird 5k Amplifiers and BSP 2.6 Processors all mounted in racks and with cabling to suit.
for phase-cancelling negative feedback producing a more natural tone. Once buffered, the SB-15 Tailbone lowers the impedance and susceptibility to hum and buzz caused by radio interference and electromagnetic fields. The SB-15 Tailbone is able to drive multiple pedals distances of 15m without noise. It also provides an alternative power solution for Radial’s Tonebone pedals, which need 15V and don’t sit well with global 9V power supply bricks. The Tailbone can take two standard 9V feeds and output the 15V required. Amber Technology: 1800 251 367 or www.ambertech.com.au
WHAT YOU MIX IS WHAT YOU GET Dutch company Alcons has been pushing hard this year for its share of the line array market. In January we saw the release of the RR12 “building block” cabinet. Now it’s the turn of the LR24, a 3-way, larger-format line-source sound system to be used as vertical array either in stacked or flown configurations. The HF section employs the RBN1402rsr pro-ribbon driver with 14-inch voicecoil. The combination of the high sensitivity and “the unusually high” HF peak power handling of 3000W promises intelligibility and throw with ample dynamic headroom in reserve under any conditions. There’s a bit of secret squirrel stuff happening — the purposedesigned Alcons RBN1402rsr driver features no less than four patents. The MF-section uses four highefficiency Neodymium 6.5-inch midrange transducers in a symmetrical loaded configuration. The LF section consists of two long-excursion, reflex-loaded 12-inch woofers with carbon-reinforced cones. Alcon’s catchcry is “What you mix is what you get.” Loud & Clear Sales: (02) 9439 9723 or www.loudandclearsales.com.au
PRESONUS AI SERIES REMOTE CONTROL PreSonus’ new SL Room Control is systemcontrol and performance-monitoring software for StudioLive AI Active Integration loudspeakers. SL Room Control gives you the equivalent control of a rackmount loudspeaker management system and more, and it works wirelessly or over a wired (Ethernet) local area network. Tools include the usual suspects to compensate for room anomalies, create delay systems, eliminate feedback, and more. The software also lets you remotely select any of four DSP contours that customise the loudspeaker for normal operation, playing low-bitrate MP3s, floor-monitor operation, or extended low-frequency boost for the StudioLive 18sAI subwoofer. This lets you optimise each speaker for its application without needing access to the back panel. You can
also remote control the 100Hz high-pass filter on each full-range model and the polarity reverse on the 18sAI subwoofer. You can edit, save, recall, and share speaker setup parameters, as well as set custom labels and comments for each speaker. Monitoring key performance indicators in real time for each StudioLive AI speaker on the network will let you spot thermal problems, driver over-excursion and keep an eye on signal levels. SL Room Control for Mac OS X and Windows is free to registered StudioLive AI-series loudspeaker owners and can be downloaded from the customer’s online My PreSonus account. SL Room Control for iPad is a free download from the Apple App Store. National Audio Systems: 1800 441 440 or email@example.com
A 4K, 3D-capable Christie projector provides a colour-grading-quality image onto a Harkness, perforated, zero-gain screen.
Smart AV Tango in the centre mostly used to control ProTools via HUI. Five ProTools systems and an Avid Media Composer.
Outboard includes: Bricasti M7 and TC M6000 reverb; CEDAR DNS and a Junger b42 (four-channel dynamics processor, mainly used for de-essing dialog).
Welcome to the new Dolby Atmos mix theatre in Soundfirmâ€™s Melbourne studios. Story: Christopher Holder
Meyer Sound active loudspeaker system. 47 loudspeaker channels in all: LCR Acheron 80s behind the screen with five X-800c subs; four rows of eight HMS Series loudspeakers down the room/ceiling; five HMS10 speakers at the rear with two 500-HP subs.
Only three computer displays required thanks to KVMs to call up any video input, including ProTools, Media Composer or the IKIS Harrison automation screen.
Harrison Trion digital mixing console: 230 inputs, 72 input faders (four layers) and 16 layered remote faders: each channel has two inputs, 16 auxes, eight bands of EQ, gate and compressor.
When Roger Savage was preparing to move into bigger and better digs in South Melbourne he knew something was brewing at Dolby Labs. Regardless of any new surround format on the horizon Roger had already drawn up plans for a new Soundfirm flagship mix theatre. The aim: to have the premier film sound mix room in Australia. The fact that Dolby had plans of its own simply meant that Soundfirm could futureproof itself. The room is a beauty; the audio and vision is superb. From an audio perspective, Roger’s 2IC, rerecording mixer and sound designer Chris Goodes, had caught up with Meyer Sound’s cinema loudspeakers while on a study tour of the US. Dolby had arranged for a demo of Atmos at one of the Skywalker Ranch mix rooms equipped with Meyer’s loudspeakers. When the showreel sparked up, Chris emitted an audible gasp of admiration: “We’ve not got to the impressive bit yet Chris,” noted the Dolby dude. To which he replied: “It’s the dialog. I’ve never heard dialog sound so good.” (The Acheron’s 580Hz crossover point places most of the dialog in the horn, which makes it particularly well suited to cinema applications.) The die was cast: the new room would be a Meyer Sound room. It would also be a Harrison room. Roger has been a Harrison Consoles man for some time. The new Trion is a version of the company’s MPC5.
RETURN TO FADERS Atmos isn’t the only big change in audio post production. Brent Heber brings everyone up to date: Many will remember the Fairlight MFX or dSP Postation (both Australian innovations) with great affection. Digidesign’s ProTools MixPlus system effectively buried those post-specific hardware/software solutions. Mixing in the box became de rigueur — a consumer computer is now commonly the heart of our professional audio world, with enough grunt to mix and process hundreds of tracks with stunning clarity. Ironically, not so long after convincing the industry it didn’t need faders, Avid née Digidesign set about selling faders back to post studios, releasing the ProControl, Control24 and, most notably, the Icon surfaces in 2004. All this in 15 years. Icon has been widely adopted, with Avid’s latest Eucon-based System 6 the latest innovation.
ABOUT DOLBY ATMOS SURROUND FORMAT • The Atmos Rendering & Mastering Unit (RMU) accepts 10 bed tracks/stems (up to 9.1) which can be mixed/panned as per usual. • Additionally, as many as 118 individual sounds can be pulled out of the main mix and addressed as Objects to be steered around the room based on vector metadata. Atmos replay hardware in the cinema renders the info in real time for each Object related to the installed speaker layout. • Each speaker channel can be addressed discretely by an Object. • Each speaker is full range. • Each Atmos room or cinema needs to be Dolby accredited. Prior to use, a Dolby rep will ping the room which applies Lake processing across every channel — every Atmos room should (theoretically) sound the same regardless of size.
LOUDNESS: PEACE AT LAST A lot of research has been done into how the human ear perceives the loudness of an audio signal and once that was understood a dBFS-scale measurement was developed representing a far superior way of working than using RMS or peak scales; something more closely aligned with human hearing. The key difference is that loudness is measured over time: an ‘integration time’. Consequently you can have instantaneous measurement of loudness, short-term measurement (over say 2-5s) and then ‘integrated’ over the duration of your program material. Along the way, our headroom in the final product is also changing. For analogue transmission we always had to keep our audio below -10dBFS. With the digital age we can now open up those limiters all the way to -2dBFS, however this is measured as a ‘true peak’ or integrated peak, ie. how high the peak would really be in the analogue realm if the digital signal were put back together again. We can now mix a lot more dynamically for TV, aligning more closely with the cinema mixing experience. – Brent Heber
It’s a disarmingly unadorned console. It’s like the company had spent 90% of its budget on the best/brightest coders in the land, and then designed the aesthetics/GUI in an afternoon at the company BBQ. That said, the Trion is a sound-for-picture thoroughbred: Chris Goodes: “I’m a big fan of using the EQ and processing on the console. It’s so quick to call it up on the console. It’s all in line: compression, EQ, 16 aux and panning plus your bus outputs, which are all automatable — something ProTools still can’t do is automate your outputs. The automation is rock solid, fast and simple; very easy to edit then reconform the automation. “Also, Harrison is now releasing an Atmosspecific panning ‘plug-in’ that allows you to hit a button on the relevant channel which then auto reassigns that channel input on the router — takes it out of the main mix and automatically turns it into an Object for panning. That info gets stored as the metadata for that Object. It’ll make workflow so much simpler.” EMA_AT94_HR.pdf
Roger spec’ed his Trion to include an impressive complement of faders and allow the room to accommodate two or even three operators: Roger Savage: “You can have a sweetspot over 16 faders, and pull in any input across those faders. So if there aren’t two operators you can do that. The truth is, we have more faders than we need. But someone used to mixing in Hollywood will want two mix engineers, sometimes three. And if money is no object, naturally, that’s a good thing.” Wedged between the two halves of Trion is a Smart AV Tango controller. Roger uses his Tango to dig into ’Tools and make tweaks or write automation within stems. Roger calls it a ‘hybrid’ approach. You can have your ProTools sessions coming up as many or as little of the Trion’s input channels as you like, with the Tango bringing its motorised faders and touchscreen to bear on what’s ‘in the box’. Mind you, Tango uses the now-venerable HUI protocol to control ProTools.
WORKING REMOTELY Faster, cheaper internet has meant the widescale adoption of file sharing sites like Dropbox and Yousendit. While ISDN became a thing of the past for remote voiceovers and ADR. Source Connect was the final nail in the coffin and we could now start recording at decent quality from DAW-to-DAW over the web. Timecode lockup was added, dropout capturing/buffering was added and now it’s commonplace to record someone on the other side of the world using this sort of technology. The post pro world has shrunk considerably as a result and many freelancers are taking on work from home without the overheads of bricks and mortar. – Brent Heber
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THE ATMOS SPHERE
MANY DEATHS OF TAPE
Atmos is new and a pain in the bottom line to install into an existing movie theatre. Not so hard for new builds. Which goes to explain why there are so many Atmos-equipped theatres springing up in the land of the new build: China.
Timecode and the various niche machines that read it and acted on it were vital for post production 15 years ago. Sony announced the end of its DAT machine production in 2005 and most soon followed, allowing hard drive-based sound recorders to flourish. Now, a location sound recordist might hand over a USB stick, a hard drive or a DVD-RAM disk full of Broadcast Wave Files. Smooth sailing from hereon? Not quite. Even though DAT replacements have been around for years, most video software to this day have a bunch of gotchas about how they deal with location sound files, and film productions are still rife with workflow problems trying to get audio in sync in this new digital age. – Brent Heber
As it happens, Soundfirm is big in the People’s Republic. It has a Beijing branch office, in fact. Saying that, Soundfirm Melbourne houses the studio chain’s only Atmos room. So when the final mix of Filmko Entertainment’s big-budget, CGIheavy blockbuster, The Monkey King, arrived in the inbox, it headed south. Chris Goodes: “We got the bed tracks premixed in 9.1. We ran a ProTools HDX2 rig that was full: 512 voices coming into the console on 112 inputs. We had another ProTools system loaded with another 256 voices and other FX. “We mixed natively in Atmos from the beginning, which meant there was no need to undo the original surround sound mix. It also allowed us to have fun separating out elements such as the music. For example, we placed the choir in the ceiling at one point making more space on screen. “So the final mix was mostly 9.1 with the occasional Object movement. But those moments really pop when they’re happening.” They sure do. Fight scenes — of which there are plenty — are dynamic beyond anything you’ve heard in conventional surround. And loud. With nearly 50 full-range loudspeakers coming atcha full throttle, it makes for some intense sonic moments.
OPEN SOURCE SOUNDTRACK ENCODING Since 1976, Dolby Stereo and its later surround incarnations have ruled the cinema sound roost. The whole point of Dolby Stereo was quality control at both ends: noise reduction and encoding to put the soundtrack onto the sprocketed film, and then decoding hardware to read it and play it back in the cinema. In a masterstroke, Dolby only licensed the encoding devices to dub stages, and sent out a technician to operate it. So if you wanted to finish a project on film, you needed to pay Dolby for this service and we’re talking five figures. George Lucas emerged as an unlikely saviour. He chose to demonstrate a new digital format for cinema called the Digital Cinema Package (or DCP) by releasing Episode 1 The Phantom
Menace on this format, played digitally in four cinemas in the USA. DCP was originally developed as a package to distribute films digitally but the Society of Motion Picture & Television Engineers (SMPTE) saw its full potential. A DCP contains MXF media, with JPEG2000-encoded video and BWAV mono audio, 24-bit/48k. That’s right, the audio in the DCP is just uncompressed wav files — no more expensive Dolby licenses and equipment hire, any ol’ bedroom cowboy can now make a film soundtrack. The democratisation of film sound hasn’t been without its land mines. Interestingly, Dolby is doing its best to reinstate the old status quo with the introduction of Atmos. – Brent Heber
Before the final cinema mixes were due, the film studio needed a 5.1 version ready for Chinese new year. With time of the essence, Chris hit the ‘5.1 Fold Down’ button on the Dolby RMU and was pleasantly surprised. Chris Goodes: “The 5.1 auto fold down worked very well. We played through the whole film and it sounded great. Dolby had done a lot of work in the coding. Even perceptively, they’re doing some clever things with height — pychoacoustically it’s approximating some of the height information from the Atmos mix.” FUTURE READY?
Like any big generational change, it’ll take time for the Atmos juggernaut to truly gain momentum. Cinema take up in this part of the world is slow, and as a result local producers aren’t jumping out of their skin to pony-up for an Atmos mix. But as a sound professional or, indeed, as a film director, once you’ve heard what an Atmos mix is capable of, there’s no way of unhearing it. The creative possibilities are endless. Only time will tell just how much audiences will demand it. Roger Savage clearly sees it as a big part of his future: “We’re building an Atmos room in Soundfirm Beijing at the moment. It’s hard to find a room big enough in downtown Beijing. But that’s now happening. We believe in Atmos and are excited by its future.”
QUICKTIME IN NICK OF TIME In the days of Issue One of AudioTechnology, an audio post house would be routinely sent video on tape, Betacam, SP, digital, you name it. Some of these decks cost upwards of $70k, so the ‘price of admission’ kept novices out of the industry. Although QuickTime movies existed back then, the common trend was to plug in the tape and capture the images, convert that into a QuickTime movie, spot it into your workstation and start working. These days, we’re skipping the middle man and the picture editor exports all manner of video files for different people depending on their craft: the composer will get one, the audio post guys, the colour graders and all these 1s and 0s then go to air, often submitted to broadcast now as a digital file. – Brent Heber
Elmhirst’s credits include Adele, Amy Winehouse, The Black Keys, Cee Lo Green, Arcade Fire, Florence & The Machine, Mark Ronson, U2, Peter Gabriel, Rufus Wainwright, Goldfrapp, and many more. Elmhirst moved from his native England to New York in 2012, and has since conducted his mixes in Electric Lady Studios Studio C, where pride of place goes to his Neve VR72 console and favourite ATC SCM50 monitors. The Briton remains one of the prime mixers who continues to work on a large format analogue console, saying he likes to “marry the best of the old and new. I could not live without faders and outboard gear. But I could also not live without ProTools. For me, it’s about getting that combination to work.”
Beck’s 12th studio album, Morning Phase, is his take on seventies West Coast folk-rock albums. “The songs are coming out of a California tradition,” said Beck. “I’m hearing the Byrds, Crosby Stills and Nash, Gram Parsons, Neil Young — the bigger idea of what that sound is to me.” Its 13 lush and panoramic songs meander slowly over 47 minutes, mostly at ballad tempo, featuring a wide variety of acoustic instruments, including ukulele, charango, celeste, dulcimer, glockenspiel, piano, upright bass, cello, organ, pedal steel and strings. Beck’s hanging vocals float over soundscapes that have a lot of depth, courtesy of a crafty use of reverb and open dynamics. In many ways Morning Phase sounds like a continuation of Beck’s Nigel Godrich-produced 2002 album Sea Change, which featured a similarly introverted mood and instrumentation. But whereas Sea Change was full of heartbreak and desolation, Morning Phase is more hopeful, with the lyrics and the album’s general mood reflecting the optimism of the early morning. Partly inspired by Beck’s recovery from an incapacitating back injury where he couldn’t even play his guitar, the album’s theme is the return of the light after a period of darkness. THE TALENT ON CALL
Morning Phase was recorded by Darrell Thorpe over several years and at multiple locations. Thorpe, who was an assistant on Sea Change and engineered The Information (2006) and Modern Guilt (2008), helped Beck lay down some early material in Nashville in 2005. They returned in 2012 to record more tracks at Jack White’s Third Man studios. Following this, the majority of the album was recorded at various studios in Los Angeles, including the orchestral material which was arranged by Beck and his father David Campbell. In the summer of 2013, Beck, who produced Morning Phase himself, judged that the recordings for the album were almost complete, and called one of the world’s top mixers at his studio in Electric Lady in New York. “It was quite a surprising phone call,” remembers Tom Elmhirst. “Because I’m a fan of Beck’s work. He said he kept hearing things he liked, and found it was often me mixing, so he asked me whether I would mix his album. I mixed one song for him while he was still doing bits and pieces in Los Angeles with his dad. Then Beck came to New York during the summer to do some shows on the East Coast and we mixed the rest of the album AT 20
here in three weeks. He needed to be around because he’s very hands-on, very studio-savvy, very knowledgeable. Then there were quite a few revisions over the next two months when Beck was on tour in South America. “For me it was probably the most satisfying experience I’ve ever had in the studio, because he knows exactly what he’s doing and what he wants. I have never worked with an artist who can hear the things he does. And the basic tracks were amazingly well-recorded in great studios, so the multi-tracks were an absolute pleasure to work with.” MIX AS AN ALBUM
Given Elmhirst’s pedigree, for him to rank the Morning Phase sessions as “the most satisfying experience he ever had in the studio” is quite a statement. He only mixes music he enjoys, with a preference for mixing entire albums rather than single tracks. “I don’t like doing just individual tracks, unless it’s something special or for an interesting artist. I also like working with artists who have a career, and have been quite lucky to repeatedly mix for people who continue to make records, which is not a given nowadays. I also enjoy collaborations — like when working with Arcade Fire or U2 — where there are lots of people doing their part, and they may use a rough mix for the chorus and something I did for the verse. They jump around and are not precious.” Elmhirst has in recent months been working on U2’s forthcoming and again-postponed album, and also mixed many of the main songs on Arcade Fire’s Reflektor. His work on Morning Phase was also, to a large degree, a collaboration
Listen To Blue Moon from Beck’s Morning Phase youtu.be/WIWbgR4vYiw
with Beck. But with Elmhirst mixing the entire project because it was essential for the feel and flow of the album. “I enjoy mixing albums,” elaborates Elmhirst, “because when you start mixing, you don’t know what’s coming up. You may mix the first songs in one way, and by the time you’re four or five songs in you may have to think again a little bit. This may mean revisiting some of the earlier mixes. You’re shaping an entire album as a piece of art, and everything has to hang together.” READING THE PLAY
Elmhirst: “For me Morning Phase is a very visual album. It’s very cinematic. There’s a lot of talk about mornings, and there’s an undercurrent of the sea, like with the track Wave, which is just Beck and the orchestra. It was the first song for the album that I mixed, and it created a blueprint for the entire record. It sounds a bit like a West Coast seventies record, it has the solid drum sound you associate with Neil Young and the records from that era. One of the two drummers Beck used was James Gadson, who is 75 and has worked with Bill Withers, The Temptations, BB King, Quincy Jones and many others. Gadson really has his own sound, which is incredible. The contributions and mix of his and Joey Waronker’s drums were incredibly important, because the tempos were so slow and we had to keep the album as a whole moving. We were conscious of the risk the tunes would become dirge-like. Making sure we kept the groove was one way we counteracted that. “Another way was to make regular changes to the scenery. For example, on some songs the vocals were dry and placed in the middle, while
on others they were expansive and placed more left and right. Reverbs — coming mostly from the D-Verb Mark 1 — were an important aspect of the sound, and we varied them a lot to create depth. The song sequencing was also important, so you didn’t get the feeling that too many tracks were in the same tempo. Beck really cares about audio quality, and is very aware there are many audiophiles amongst his fans who will sit down in front of good speakers to listen to his albums. So no expense and time was spared to get the best possible result.” SINGLE ONCE IN A BLUE MOON
The song Blue Moon was released as the first single from Morning Phase, on January 20th of this year, a month before the album’s release. It’s one of the few mid-tempo tracks on the album, with a sparkling charango providing the early morning sunshine, backed by two acoustic guitars, a ukulele, glockenspiel, and multiple clavinets and pianos. Elmhirst: “The Blue Moon and Heart As A Drum tracks stood out because of their slightly higher tempo. But while we were mixing there was no awareness of what the single was going to be. In fact, I had conversations with Beck and his management about Morning Phase very much being an album and that it wouldn’t make a lot of sense listening to any of the songs in isolation.” Elmhirst had a point, because Blue Moon, and the album’s second single, Waking Light both failed to chart. By contrast, Morning Phase turned out to be Beck’s most successful album to date, in terms of chart positioning, reaching No. 2 in Canada, No. 3 in the US, No. 4 in the UK, and No. 5 in Australia, and the top 10 in a dozen other countries.
OUTBOARD Elmhirst is still one of the world’s prime exponents of mixing on a console, but also someone who aspires to “marry the best of the old and the new.” He makes extensive use of plug-ins in the mix session and then lays all the tracks out over his Neve board for compression and EQ — “I did a lot of reductive EQ, with a high Q and notching out frequencies” — and also applies a number of outboard effects. On lead vocals, Elmhirst used a Neve 1081 with a high pass at 82Hz and a little cut at 15kHz, an Altec A322C tube compressor and Pultec EQP-1A3. On the snare he used a dbx 160 compressor and a Neve 1081 EQ with a gentle shelf at 15kHz, a wide bump at 8.2kHz and a high-pass taking out any rumble below 82Hz. He also parallel compressed the drums using a Chandler TG1 set to limit with the input gain cranked right up. The silver face Urei 1176 compressor Elmhirst put on bass had a ratio of 8:1, with a slow attack and fast release. On the mix bus Elmhirst used a Chandler TG12345 with half dB steps to add a small 1dB bump at 70Hz and 4dB of lift at the top end around 20kHz, inserted a Manley Vari Mu with slow attack and medium recovery, and went back into the 24-bit/96k ProTools session via a Cranesong Hedd 192 with some tape sound dialled in. Elmhirst later sent the drum stem through his hardware EMT140 plate reverb, “because it needed to be a bit more expansive in certain sections of the song. I used the 140 quite a lot in some places.” AT 21
BLUE MOON SESSION
Elmhirst calls the Blue Moon session “full but also pretty straightforward.” At the top of the original mix session, is the rough mix (1832), followed by the mix print (Ref 1). The 10 tracks in pink are the other drum ‘bits and pieces’ followed by a bass track, two main acoustic guitar tracks and two acoustic guitar tracks for the break (all in green), the charango track (CMP), three ukulele tracks, the synth solo at the end of the song (kldbl), two clavinet tracks, four piano tracks, three glockenspiel tracks (2813, DV92), a double-tracked lead vocal that was split in three double tracks for different sections of the song, and Beck’s backing vocals. There are also a number of aux tracks in the session: BSU and BDTS just below the glockenspiel are for the vocals, and the three tracks at the bottom are more general effects tracks, respectively D-Verb, the Trillium Lane Labs Orban spring reverb plugin and the UAD EMT140 plate reverb plug-in.
DRUM BOUNCE ‘BND’ is a stereo bounce of the drum mix, which, said Elmhirst, “is how I received them. Some of these drum tracks were recorded three years ago. Most of the other songs had fairly typical drum tracks, with bass drum, snare and so on split out, but on this song I received the drums in bits and pieces and they needed fitting together a bit. I had the UAD Manley Massive Passive, UAD Shadow Hills mastering compressor, and the PSP Vintage Warmer on the drum stereo bounce track.”
It was probably the most satisfying experience I’ve ever had in the studio. I have never worked with an artist who can hear the things he does
— Tom Elmhirst
OUTRO DRUMS Elmhirst: “The ‘120tr’ track is the outro drums, and I had the Digirack EQ on there as a hi-pass, the Waves API EQ and the Waves MV2 compressor — I was trying to squash the track a bit, making it a bit denser to give it more power.”
VOCAL VERB Elmhirst: “I love the Orban plug-in, and now also have a real Orban. But I didn’t use a lot of spring reverb on this album, because it had a different kind of sound. As I said earlier, the D-Verb Mark 1 was quite a big part of this record. Beck is a big fan of that plug-in, as am I. It’s very transparent. I probably EQ-ed it a bit on the way back in, and particularly on the vocals I used very long reverb settings, in some cases as long as 11 to 12 seconds. The other two plug-ins I used on the vocals are the Waves De-esser and a bit of brightening from the Waves API 550a.” AT 22
PHASING OUT STEMS TECHNIQUE
Given today’s demand for instant, 24/7/365 recalls, Elmhirst also creates a stem mix in addition to the stereo mix. Virtually all console mixers do this today, as it allows them to quickly run off an alternative in-the-box mix by rebalancing the stem session. It’s always a compromise though, as mixing stems will never have the same interplay with the mix bus as rebalancing a complete mix. But on the flip side, recalling an entire’s studios outboard and console is unthinkable. Elmhirst, together with his assistant Ben Baptie, has worked out a rather surprising alternative to the instant recall conundrum: a seemingly counterintuitive method the mixer admits is “bonkers”. Instead of mixing stem sessions to obtain new stereo mixes, Elmhirst continues to work with his original reference stereo mix, massaging it in-the-box with in and out of phase stem tracks, until he and the client are 100% happy. Elmhirst: “I don’t like using stems. I want to be able to keep using my original mix, in part because the sound of the Manley compressor on the stereo bus is so important. But I still want to be able to play with the volumes. So purely by accident Ben worked out this reverse phase technique. We play the stem of the part we want to change at the same time as the reference mix, and when we reverse the phase of the stem track, using the Digidesign EQ, the part will simply disappear from the main mix. “For example, we put up a stem of the drums, flip its phase, and it cancels out the drums in the master mix. The drums will be totally gone. If we turn the stem track down, the drums will reappear in the mix. I still can’t quite get my head around how it works, but it does, and it’s pretty cool, because it allows me to keep the original mix with the Manley mix bus compression. Of course, to make the drums, or any other instrument, slightly louder, we simply add some of the drum stems with normal phase. “We’re not talking night and day changes to the mixes. The changes are scenic, like bringing the vocals up in certain places or the harmonies down, and so on. Also, if we wanted to change the sound of a part, we cancel a part by using reverse phase and then add in another stem with added EQ or effects. “We had eight revisions of Blue Moon, doing things like pushing the lead vocal up 0.25dB, pushing up the verse drums, a vocal mix with a low pass filter to brighten all vocals adding instrumental and cappella tracks, there’s stems coming in and out right through.”
DRUMMING UP WAKING LIGHT The final song on Morning Phase, Waking Light is one of the more expansive tracks on the album, complete with strings, two glockenspiel tracks, six clavinet tracks, five organ tracks, four piano tracks, triple-tracked chorus vocals, and a brief electric guitar cadenza at the end, culminating in a crashing, feedbacked guitar that finishes off the track. Elmhirst: “This was another one of the bigger mixes, and in the end it really lifts off. The instrumental outro is quite intense, and I used the Soundtoys PitchBlender plug-in on the guitars to create a Leslie effect. The drum treatments in ProTools were different, because I was dealing with individual drum parts.”
“The straight kick drum had the Waves Jack Joseph Puig drums plugin, while the Kick sub had the Waves Q10 EQ and Waves LoAir for some more sub, plus the Waves C1 gate.”
“The tom subgroup had the Fairchild 660 plug-in and the Kramer Pie compressor, and I reduced the sustain of the Floor Tom with the SPL Transient designer. I’m doing quite a lot on the toms.”
“I took some top end off the Snare with Lo-Fi, and used TapeHead to get some more drive on the combined Kick/ Snare track.”
“We even did a vocal edit to insert a lyric change. It also has a wet track, because Beck will have sent his vocal to an echo chamber, at Cello, East West or Capitol. He loves echo chambers. This whole record was so well-recorded that most of what I did was just balancing things.”
PC AUDIO There are testing times ahead, so why not invest in a few free PC audio utilities? Column: Martin Walker
Some people rely on their ears, but there’s no denying that having visual evidence of what various plug-ins are doing can be extremely useful, even if it confirms that a particular plug-in is doing nothing at all. So, with this mind I’ve decided to gather together some of the audio utilities and PC test techniques that have proved useful to me over the years. Christian Budde’s VST Plugin Analyzer (www.savioursofsoul.de/Christian/programs/ measurement-programs) is an extremely useful PC-only utility if you want to learn more about what particular VST plug-ins are doing to your sound. Widely used by plug-in developers to test their own creations, it can also be a wonderful educational tool for the end user. You just load in your favourite VST plug-in, load in one of its presets (or adjust the plug-in controls manually), and then run the various test options to see a plot of the frequency response (great for seeing why one EQ or preamp sounds different from another, examine any distortion (desirable or otherwise), explore the input/output characteristics of (for instance) compressors, and lots more. Sadly this utility only works with 32-bit plugins and does take a little effort to get to grips with, but it’s often helped me understand why a particular plug-in sounds as good as it does. ABX testing (aka double blind testing) is widely used in the audio industry to find if there really are any audible differences between two signals. The idea is that you can audition one or the other on demand (the A or B signals), followed by one of these two at random (X) to see if you judge which of the two it is. If X cannot be reliably identified after a number of these ABX tests (typically 9 out of 10 trials must be guessed correctly to be statistically significant) then it’s judged that there’s no perceptible difference between them. Software versions of the ABX test are available, and the easiest to use in my experience is part of the freeware foobar2000 audio player for Windows (www.foobar2000.org) developed by Peter Pawlowski. I use this player in preference to Microsoft’s own Media Player for all my general-purpose uncompressed and compressed AT 24
audio file listening, as its highly modular nature lets you add ‘components’ that, for example, support ASIO drivers for low-latency listening, bypass Microsoft’s Kernel Streaming to ensure bit-perfect digital transmission, play back unusual audio formats, decode various sequenced formats such as SIDchip, MOD game music and General MIDI files, perform bpm analysis, and a host of other options. The foobar2000 ABX Comparator add-on is a wonderful tool to dispel audio myths, or to prove whether or not you’re fooling yourself that one audio signal sounds better than another. Classic examples include comparing MP3 files at different rates with the original audio — some of you may be surprised that even if you can hear any difference between 256kbps and lossless audio, you may not be able to reliably decide which is which. Once you add a hardware send/ return loop into the proceedings via your audio interface you can even check out whether or not changing audio or even digital cables makes an audible difference that you can hear consistently and reliably. You can also use ABX tests to gradually ‘train’ your ears to hear smaller and smaller differences in amplitude, frequency, EQ, and so on. Christian Budde’s ABX Test for VST-Plugins (use above link) is a standalone utility that lets you load two VST plug-ins and an audio file to play through them both, to compare their effects. Although it sounds a bit like a party game for nerds, it really can root out whether you or your ears are being fooled by a fancy GUI or an expensive price tag. It’s also great for comparing the effects of two EQs, compressors, and so on. Another classic and very informative test is the antiphase null — clone any audio track onto another channel in your DAW, and then flip the polarity of one track — the two should cancel out completely, leaving digital silence during playback. Now add your choice of plug-in to one of the channels, and tweak that audio channel level slightly if required to compensate for any gain change due to the plug-in. Once you’ve got the lowest combined output signal, what’s left is whatever has been added by the plug-in (some
audio editors such as Steinberg’s Wavelab even build in such functions into their arsenal). The antiphase null can be a fascinating test, letting you hear exactly how preamp plug-ins alter your audio, whether or not they add extra harmonic content (distortion), or how compressors change the ‘envelope’ of the sound. This really can hone your listening and mixing skills. A variant of this technique is to pass each of the two channels through a different EQ plug-in, with their settings matched as closely as possible to minimise the difference signal. Eric Beam carried out quite a few such tests in Digital EQ Fact & Myth (rhythminmind.net/1313/?p=361) using various digital parametric EQ plug-ins. One EQ may be easier or more pleasing to use than another, but he came to the conclusion that most can be made to sound identical with just a little effort. The exceptions are those that model saturation or other ‘analogue-like’ nonlinearities, which will add a dash of character en route. Yet another revealing test for stereo signals is to download a freeware utility plug-in such as Kelly Industries Stereo Tools (www.kellyindustries. com/stereo_tools.html). Insert this into a stereo channel and then click on its Mono button to hear just the Mid (sum) signal, and then click on one or other of its channel invert buttons. Now you’ll be able to hear just the Side (difference) signal, with all the sounds normally located in the middle of the stereo image (commonly main vocals, bass, and kick drum) partly or even completely nulled out. What remains can be extremely revealing — harmony vocals, stereo early reflections and later reverb tails, delay bounces and other special effects, doubled guitars, extra keyboard parts, incidental percussion, distortion, compression artefacts. You can also glean a lot from commercial recordings by listening to the side signal alone, such as mixing tricks and stereo techniques that may be masked by the often stronger mid components, and it can also help reveal what perceptual encoding such as MP3 is doing to your music. Learning can be fun!
APPLE NOTES The ‘Trashcan’ Is Mighty Tiny Column: Anthony Garvin
Whilst collating the information about the history of the Mac for our 100th issue, the obvious became even more obvious to me — Apple as an organisation has permeated almost all areas of musical life. It’s not just their computers that are used heavily for music production, but apps on the iPhone and iPad have turned these devices into everyday pieces of music hardware too. Then of course there’s Apple’s own DAW, Logic. Plus, the iPod and iTunes have all become more or less the standard software and hardware to consume music on. Why bother, you ask? Well, if we audio producers don’t keep one eye trained on consumption trends, we could be barking up the wrong tree. Mac Notes has now become Apple Notes for this very reason. Of course, Mac computers are still going to a big focus of my columns — Mac Pro review part one below — I’ll just be including a bit of everything else, too. So, let’s start with a few goings-on in Apple land. CARPLAY
Apple’s Carplay is essentially an integration of iOS7 into a compatible car’s entertainment / navigation system. Music is a large part of this, with Spotify, Beats, iHeartRadio, Stitcher and Apple’s own Podcast apps already compatible, all controllable via the touchscreens, steering wheel controls and other buttons to control the iPhone directly. Meaning no more fiddling between devices to crank your tunes down the highway. Now we just need the cars to use Carplay with. Apparently Honda, Hyundai, Volvo, MercedesBenz and Ferrari will all have compatible cars out in 2014, with 13 other partners’ models in the works. And Pioneer just announced its first three aftermarket in-dash CarPlay systems, due towards the end of the year. CHANGES TO ITUNES?
The rumour mill has been turning a little recently, with ideas that Apple is seriously reviewing the way it runs iTunes. Last year, music download sales dropped for the first time, and at the same time, streaming services grew quite rapidly. Apple can’t ignore music AT 26
streaming any longer. iTunes Radio is a small step in this direction, but are we going to see a fully implemented iTunes/Apple streaming service this year? As a big fan of streaming myself, I hope so. It would be interesting to see Apple approach this market, with Spotify currently holding the lion’s share. It’s also rumoured Apple is considering an Android version of the iTunes app, to try and encapsulate users of the platform, which is now growing faster than the iPhone. THE NEW MAC PRO
Finally, three months since the release date (and just before submitting this column) I have got my hands on one of the new ‘trashcan’ Mac Pros. Over the next few issues, I’ll be sharing my thoughts from in-the-field use of the machine. So, here’s a taster. INITIAL SURPRISE
We all knew the new Mac Pro was going to be much smaller than the previous generation, however, its actual size is shocking. The Mac Pro’s diameter is less than an average tin of paint, at about 17cm. And it definitely wouldn’t make for a good trashcan, at 25cm high. In fact, I reckon you could stack the new Mac Pro two high and almost three wide, and you still wouldn’t be taking up the same amount of space as the old Mac Pro. Shocking! Now (without even the smallest bit of bias), I have to say the physical design of the unit is beautiful. Despite being small, the unit is solid and feels valuable. With an effortless click of a physical ‘unlock’ switch, the machined aluminium cylindrical cover simply slides off to reveal the various motherboards, ram and flash storage. One practical complaint is the power button nestled at the rear of the machine amongst the power, Ethernet and HDMI ports. Whilst hiding it on the rear may look good, it’s going to be a pain for users powering up on a day-to-day basis. Another design coup is the fan. Turning the unit on, this was the first thing I noticed (or didn’t notice?). It was dead quiet. I tried to measure the exact noise level of the fan, but it wasn’t above the noise floor in the (soundproof, very
quiet) studio. Unless I have my ear on top of the machine, I can’t hear it. As I punish it (or try to) with tasks over the coming weeks, I’ll be keeping an ear on this. Perhaps as it works harder, it will also work louder. USING THE MACHINE
Lucky for me, the Mac Pro I was sent is an 8– core 3GHz machine, with 32GB RAM, so it’s not exactly the base model (with a price tag of $8,668.99), but it’s not the fastest model either. As soon as I started using the machine, the ‘speed’ became very apparent. I say ‘speed’ because at this stage I’m not sure if it is due to the new flash storage, or the faster processor, or both. But for example, booting the computer takes under 12 seconds from pressing the power button to seeing the login screen. Other everyday tasks like opening applications and changing system preferences are also lightning fast — hitting option-F11 (to open System Preferences and go the Sound pane) happens instantly as I hit the keys. Unfortunately, with limited time between receiving the machine and writing this, I’ve only been able to run up Ableton Live to finish off some production tasks. It wasn’t anywhere near the most strenuous of sessions, but running Live at a buffer of 64 samples, with an Apogee Quartet, for a full day’s work of recording and editing was zippy and trouble-free. Doing a quick benchmark test, I nearly fell off my chair when I tested the internal drive’s speed. Using Blackmagic’s Disk Speed Test (as discussed in Issue 98) I found I could read and write at a speed of 950MB/s (as a comparison, my Retina Macbook Pro does 477MB/s read and 397MB/s write). Any lingering thoughts over whether a separate recording volume is a necessity can now be vanquished. Though with limited internal capacity (due to being flash storage), you’ll still need external drives for archiving or long-term storage. I have this Mac Pro in my possession for the next six weeks or so, so over the next couple of issues I’ll be looking further in depth at just how well this machine performs in real life music production scenarios. Stay tuned.
There isn’t a monitor controller worth owning which can pass audio information as cleanly as the Symphony I/O
APOGEE SYMPHONY I/O Symphony now comes pre-configured, while still remaining the most flexible converter on the market.
NEED TO KNOW
Review: Brad Watts
PRICE Systems starting at $2995 Thunderbridge: $785 PCIe Card: $1275
CONTACT Sound Distribution: (02) 8007 3327 or firstname.lastname@example.org
PROS Vast array of I/O options. Superlative operation with Mac OSX Delightful sound quality. Attractive entry price point. Multiple host options.
CONS There is a fan in it. Mac OSX-centric operation.
SUMMARY The Symphony’s finesse of control is matched by its do-anything flexibility and premium converter quality. A fully-modular converter system that will literally suit any studio — as long as it’s running a Mac.
For nigh on a quarter of a century, Apogee has sat at the pinnacle of audio-to-digital conversion technology. The definition of the term apogee itself denotes: ‘the highest point in the development of something; a climax or culmination.’ Indeed a great choice to name a company aiming for some of the most sophisticated recording I/O you can buy. Since this century’s commencement, the company has broadened its range to include entry-level interfaces to suit all recording budgets, but this hasn’t negated development of top-of-the-line interfaces for the most demanding professional situations. A few years back now, Apogee dropped its AD- and DA-16X, Rosetta, and long-standing AD8000 ranges, replacing all these units with a Swiss Army knife approach — the Symphony I/O. And while the army knife analogy might imply a makeshift resource for when real tools are unavailable, the Symphony I/O is quite the opposite — it’ll tackle its duties with aplomb. So much so that it’s feasibly the final word in multichannel digital audio I/O available. MIX IT UP
The Symphony I/O is quite a sexy piece of electronics. Yes, it’s a ‘looker’, but I’m more specifically referring to the extensive array of configuration options. The Symphony I/O initially was released as a basic mainframe or chassis, into which you could then install the appropriate I/O options. These are quite varied, allowing for pretty much any patching situation. Two card slots can accept any mixture of the following cards: The A2x6 incorporates two XLR inputs, AES digital I/O, optical I/O for ADAT, SMUX and S/ PDIF, and six balanced analogue outputs as a D-Sub connector. The A8x8 offers eight analogue I/O (balanced) and 8 AES-EBU on D-Sub connectors, plus the other optical and S/PDIF digital I/O as on the A2x6. The A16x16 provides 16 balanced analogue I/O as two sets of D-Sub connectors, along with a S/ PDIF coax I/O for up to 192k. The A8MP option for the upper slot includes eight built-in mic preamps offering a hefty 85dB of gain, each with selectable 48V phantom power, Soft Limit and phase invert, and eight software assignable analogue insert points. These are presented as discrete sends and returns, again as D-Sub connectors. It also includes four high impedance, high-level instrument inputs (as jacks). Additional cards include simple AI16 and AO16 cards, with 16 inputs and 16 outputs respectively. So you see, between all these options are all the permutations you could hope for in an audio interface; from the serious home user through to high-end studios. Oh, and if 32 I/Os isn’t enough (two 16I/O cards), two Symphony I/O units can be ganged together from a single Symphony
64 PCIe card for 64 I/O. Incidentally, the main chassis of the Symphony I/O includes a USB port which acts as both a conduit for control of services to the unit from the host computer, and as an audio interface when the device is running in USB mode (however, you’ll still need an I/O card in the unit to get audio in and out). To make things a bit simpler, these days the Symphony I/O is released in various ‘preconfigurations’ rather than simply the chassis and then adding cards. Those ‘factory’ configurations include: Symphony I/O 2x6 for $2995, the Symphony I/O 8x8 at $3695, the Symphony I/O 16x16 at $4895, and the Symphony I/O 8x8 + 8MP for a cool $5495 (which is the one sitting here on my desk). When you consider the pricing structure, it’s not that unwieldy — three grand for the entrance fee and $4.9k for 16 I/O. Now for the sake of clarity with all these I/O options, I’ll also point out the various integration options. Given who the Symphony I/O will appeal to, there are a number of host systems you can attach this unit to. Firstly, as alluded to earlier, the Symphony I/O will function via USB — all good — handy for connecting to any Apple laptop, and especially handy if using the Symphony I/O in standalone mode and requiring access to the mic preamps of the 8MP card. Secondly, the unit will, of course, converse with Apogee PCIe Symphony 64 cards, which are connected via what Apogee refer to as a ‘PC-32’ cable. Thirdly, you can connect the unit to ProTools HD, Native, and HDX cards, as the PC-32 cable is exactly the same as the Avid/ Digidesign interface cable. This is quite pertinent as those studios running ProTools hardwarebased systems such as HD, HDX, and Native, can use the same Symphony I/O interface when switching between ProTools and a completely native DAW platform such as Logic Pro or Digital Performer via Symphony 64 cards. In a pre‘trashcan’ Mac Pro you could feasibly run an HD2 system alongside a Symphony 64 card, switching between the two platforms with a simple cable swap (I’m yet to find a manufacturer supplying a hardware switch to do this — but wouldn’t that be just dandy). It’d be a clear cut method of platform swapping while keeping the same A/D D/A hardware patched permanently in place. THUNDERSTRUCK
Apogee hasn’t left the recent Mac Pro or nonPCIe-endowed Macs, such as iMacs, out of the Symphonic equation. For connecting these host machines to a Symphony I/O, the answer lies with the Symphony 64 ThunderBridge. In essence, this unit takes over in the absence of PCIe slots and an installed Symphony 64 card, providing two PC32 ports for connection of up to two fully populated Symphony I/O units. That’s a lot of I/O in a very small space, and remember it’s ThunderBolt, so you could quickly swap between MacBook and Mac Pro/iMac systems with a cable swap. Piece of cake! What’s more, the ThunderBridge unit is nearly $500 less than the Symphony 64 card.
The Symphony I/O is quite a joy to drive. Most of the settings are accessed via the Apogee Maestro software control software, including control of the eight mic preamps (with that hefty 85dB of gain). The mic pres can also be grouped for ganged level control, and these retain any offsets from initial gain settings. And, the review unit also provided insert points which could be shunted across mic pre inputs at a moment’s notice. All inputs offer limiting at -2 and -4dBFS, along with Apogee’s ‘Soft Saturate’ and ‘Soft Crush’ algorithms. I/O points can be set for various levels individually and can be trimmed in 0.1dB steps within ±2dB. For precision setup you’d look no further. The Symphony I/O actually sent me scurrying about looking for a better monitoring path, leading me to the conclusion that there isn’t a monitor controller worth owning which can pass audio information as cleanly as the Symphony I/O. With a number of high-end interfaces at my disposal during the last couple of months I’ve been re-evaluating my monitoring path. I’ve come to the realisation the only option is to come directly from the audio interface to the monitors, as no monitor controller can come close to the spec provided by units such as the Symphony I/O and, indeed, the Prism Sound units I reviewed recently. With a dynamic range of 120dB (A weighted) on the way in (A/D) with THD+N -113dB (0.00024%), and a D/A spec offering a dynamic range of 129dB and THD+N of -117dB (0.00014%), adding another veil of analogue switching, connectors, and associated resistance simply isn’t going to cut it. As it transpires, the Symphony I/O is perfectly set up for directly connecting to powered monitors, and behaves admirably when switching inputs and I/O (from the front panel or Apogee’s Maestro software for OSX) as there’s soft muting on outputs so as not to give your monitors a sudden belting when switching and powering up. As for the actual sound quality, the Symphony I/O is up there with the best of them. The sound is precise, musical, warm and inviting. When directly compared with the Prism Titan for example, I’d put the Titan above the Symphony I/O in terms of presentation of sound stage, with the Apogee sounding slightly veiled in comparison, but remember we’re talking another couple of grand in price discrepancy here, and this also needs to be tempered with the fact the Symphony I/O is a far more versatile unit, with more mic preamps if required and far more potential I/O configurations. In some ways this is an unfair comparison but it gives you an idea of where the Symphony I/O sits in the grander scheme. As a workhorse studio and recording I/O the Apogee certainly wins in the price/ performance ratio stakes.
PRISM SOUND TITAN AUDIO INTERFACE Titan is part of a whole new breed of Prism Sound interfaces that will convert you to high quality sound. Review: Brad Watts
Titan’s ivory enamel-esque front panel is especially nice — and feels like it’ll handle everyday use without complaint — if relatively sparse. The main volume control/data knob is surrounded by 15 LED indicators, and pushing the control mutes the master output. Also on the front panel are two high impedance instrument inputs, and two headphone outputs with dedicated volume controls. Aside from metering for the eight analogue inputs or outputs (switchable), master output level, and ancillary LEDs indicating mic preamp settings, that’s about it.
NEED TO KNOW
Prism Sound is a benchmark for digital audio recording and playback. There’ll be no argument there. For years the company released industrial strength A to D and D to A converters which kept the bar at prodigious heights. Like all best-in-class technology, however, Prism Sound converters were always far too expensive for the aspiring recording enthusiast. Thankfully the cost of Prism Sound’s technology has lowered appreciably in recent years, with the company’s initial foray into ‘affordable’ converters being the Orpheus back in 2008. While still a top-shelf set of converters with an equally top-shelf asking price, the Orpheus was streets more affordable than the Dream series of converters the company rose to
PRICE $6460 CONTACT CDA Pro Audio: (02) 9330 1750 or email@example.com
prominence with — those units will set you back a staggering $13,000. Now you’re back in your chair you’ll be refreshed to learn Prism Sound’s latest offering doesn’t come near those price figures, although it certainly isn’t a budget interface. The Titan is a continuation of Prism Sound’s USB-based interfaces reviewed a couple of issues back such as the Lyra and the Lyra 2. However, connectivity with the Titan is more than mere USB, and includes some versatile options for connecting to various systems. Pricing is described as ‘accessible’, which is a fair assessment considering the Prism Sound pedigree, and especially when compared with Prism’s high-falutin’, and for most, inaccessible Dream ADA-8XR.
PROS Impeccable audio reproduction and recording. More bells and whistles than you can think of. Outstanding build. Small footprint. USB connectivity. Expandable.
CONS I could say price, but you get what you pay for.
SUMMARY Prism Sound is still building gear in Ol’ Blighty worthy of Her Majesty. It’s a case of ‘get what you pay for’ with the Titan. It’s still a whole lot less than you used to have to dish out to convert to Prism Sound, but with specs that’ll no doubt cream your old gear.
ROUND THE CLOCK
The plentiful I/O section also has ancillary connectors such as ADAT optical I/O for addition of a further eight streams at 48k, or four streams at 96k. An RCA in and out provide either S/PDIF digital or AES/EBU when using the same port with the supplied S/PDIF to AES adaptor. This also offers sample rate conversion on either the way in or out (not both simultaneously). And there’s wordclock I/O for syncing with anything else in the world.
Out back there’s I/O for eight analogue audio streams. Eight TRS balanced outputs, four TRS balanced inputs and four combo TRS/XLR inputs. These last four address the four built-in microphone preamps offering 65dB of gain, 48V power, 80Hz high-pass filter, phase reverse, and a pad. Each pair of channels can also be set for M/S encoding, and Channels 1 and 2 can be set to use an RIAA EQ for inputting vinyl turntables. The remaining four TRS analogue inputs also sport high-pass filters and phase reversal, and can be individually set for -10dBV/+4dBu levels, as can the outputs for all eight analogue outputs.
EXPANDABLE, NOT EXPENDABLE The jewel in the I/O crown is an MDIO expansion slot. What’s this all about then? Well, should you want to use the Titan as say, the front end of a ProTools HDX system, the addition of a suitable card to this slot will make it possible. This card is due to land by mid-2014, with other cards offering alternates such as AES/EBU I/O and Thunderbolt connectivity to follow thereafter.
QUEEN OF CONVERSION
In use the Titan arcs up immediately. No install issues at all. Connection is via USB 2. Yep! All this I/O is shunted to your laptop or desktop CPU via the truly universal serial bus. I had no trouble whatsoever recording 18 tracks at 44.1k to a machine running OS X 10.8.4 with a 3.2GHz i5 processor. 14 tracks went down happily at 96k, and 10 tracks at 192k. The Titan swallowed it all whole at a buffer size of 128 samples. Speaking of platforms, Windows (Vista, 7 and 8, 32- and 64-bit, ASIO and WDM drivers), as well as OS X 10.4.11 and later (Intel) are all supported. Once configured with a computer, Titan can also operate as a stand-alone unit using analogue, ADAT, S/PDIF or AES/EBU I/O. I must say, until recently I’d never expected USB 2 (and 3) audio interfaces to be this bulletproof, but it appears the slow demise of Firewire has increased manufacturers’ development processes toward this ubiquitous bus system. That said, the MDIO slot could be fitted with a Thunderbolt interface option, eradicating bottleneck concerns pretty much completely.
Oh yes indeedy, this is a quality piece of work. The Titan is an interface to keep the most demanding recordist huffing upon their knuckles as they tackle a multitude of live and tracking style projects with ease and finesse. Not only does the recording and reproduction quality bear this out, but also the vast array of professional options. Obviously clock and jitter are important issues to consider with an interface of this ilk, and Prism Sound goes to great lengths to outline the technology behind its clocking technology in the unusually well presented and informative manual, along with explanations behind the various dithering algorithms included in the unit. Really, the feature list keeps on rolling with the Titan.
Equally versatile is the unit’s ability to have I/O mixed and routed allowing low-latency mixes to be bused to any of the unit’s outputs, including the digital outputs and headphone outputs. This abundance of functionality is accessed via a software control panel that appears to work flawlessly upon first install. As you can see, this is a professional recordists’ piece of interfacing, covering every possible function you could reasonably expect, and unlike many ‘pro’ interfaces, most functions are available across all analogue inputs — for example, the highpass filters and phase reversal included on the TRS inputs should your outboard preamps not include these features. Considerate indeed.
For those who get a kick out of comparing specs, true to form, Prism Sound actually publish comprehensive specifications, while so many other manufacturers aren’t game to follow suit. This in itself is a testament to the quality and conviction behind both Prism’s products and ethics — three resounding cheers. The Titan’s analogue outputs provide a dynamic range of 115dB and an impeccable THD figure of 0.00045%. That’s a seriously low distortion figure. Frequency response is well documented; with the high definition sample rates such as 96k and 192k rolling off a mere 0.05dB at 32kHz and 3dB at 47.8 and 76kHz respectively. For garden variety 44.1k and 48k rates, rolloff of 0.5dB occurs at 21.4 and 23.2kHz — damn close to the theoretical limits of each format.
MADE IN ENGLAND
The Prism Sound Titan is manufactured in England — not a phrase you read on a lot of product these days. I almost expected to read ‘By appointment to Her Majesty’ alongside it. Call me old fashioned if you wish, but there seems an extra air of authority to the Titan’s build quality not regularly seen in Chinese manufacture.
Benchmark is getting rather long in the tooth but spec-wise they’re alarmingly similar. That established, specs are evidently not the be-all and end-all as an audition comparison between these units bears out quite clearly. In other words; the Titan creamed the Benchmark. There’s much more depth to the stereo image, bottom end is far more present and ‘authoritative’, and high frequency reproduction is smooth and blissful. The Titan is an absolute joy to listen to, and those mic preamps? They’re sublimely top-notch recording channels. Also top notch is the ‘Overkiller’ circuit on each of the inputs, which can be individually instigated on each channel. This is basically an extremely fast progressive limiter which will buffer peaks to a degree of about 10dB. LED indicators on the front panel let you know when the Overkillers are actually limiting. Would I own a Titan? You can bet your bum I would. Yet while the Titan is still up there price-wise, it’s very competitive when compared to flagship units from other premier interface manufacturers. What sets the Titan apart from many other units is its extreme versatility — with a remit ranging from small on-site ensemble recordings through to acting as the main interface in bespoke recording environments. In short, a stunning piece of machinery for the recording perfectionist.
Now while on the subject of specs, it’s interesting to note the published specs of the Titan are pretty close to the published specs of the Benchmark DAC1 I use for regular monitoring. Sure the AT 31
‘Concentric Summation Array’ technology and a new phase plug ensure that the two columns of three mid frequency cone transducers sum coherently with the HF section.
The full width of the enclosure serves as a horn in the horizontal plane, for an even horizontal coverage. It also means it can smoothly handover to an adjacent column.
Simple bar-and-pin rigging doesn’t need to win any design awards — there’s only one configuration.
Outer slots provide enclosure venting for better LF.
Built-in microphone for in-the-field diagnostics. Anya’s HF horn uses a proprietary loading technology to reduce apparent source spacing for more accurate adaptive control. Each module deploys 14 independently-powered and processed HF compression drivers.
Slotted vents provide a degree of loading to the dual 15-inch LF transducers and shift the apparent LF sources farther apart, extending horizontal pattern control to minimise the build up of LF energy.
EAW ANYA With its ‘hang it straight then steer it later’ philosophy, Anya’s approach to pattern control leads the way.
NEED TO KNOW
Story: Christopher Holder
CONTACT Production Audio & Video Technology (03) 9264 800 firstname.lastname@example.org www.productionaudio.com.au Production Audio has a demo system in stock and will be road-testing it, with a launch event coming soon. Be sure to get in contact to register your interest in ANYA.
ANYA VITAL STATISTICS Transducers: LF 2 × 15-in cone, 4-inch voice coil 6 × 5-inch cone, 38mm voice coil 14 × 1-inch exit, 35mm voice coil Class-D Amps: 2 x 1700W, 6 x 350W, 14 x 350W (Max output) Horizontal Beam Width: 70° per column Operating Range: 35Hz to 18kHz
There are many ways to launch a product. If you’re Richard Branson, you’re likely to wear a kilt, be surrounded by a bevy of comely beauties, and, with the guarantee of some kinda stunt, attract half the world’s media. More traditionally for our industry, you rock up to a trade show and put out a press release. Anya was effectively launched at last year’s Coachella festival, when Rat Sound took some pre-release product and let it loose at the Palm Springs music festival. And I recall that my first ‘look’ at Anya was via a Dave Rat Instagram. Not sure if EAW has officially launched Anya yet, but gradually more info trickles out of Whitinsville. There’s the whiff of nonchalance about its introduction. That’s appearances anyway. I’ve no doubt EAW is anything but nonchalant about Anya. By its own lofty standards, EAW has been in the wilderness for 10 years or more. EAW is accustomed to being at the top of the heap and having popular, sophisticated gear that rocks. Anya may well put it back on top again and lead the way for some time to come. Anya embodies the logical extension of the latest thinking behind line array and how to attain the best pattern control. Essentially, the idea is: the more channels of amplification and processing you can squeeze into a box, the more control you have over what comes out of the box. Martin Audio’s MLA series ratcheted up the stakes by making all of its driver components individually addressable. In so doing, MLA can fine-tune
its coverage pattern better than your average tri-amped line array. Still, to achieve the coverage required (to simultaneously get way up the back and cover the first row), MLA needs to be splayed in the now-familiar J hang. Why not flat and straight? Let’s get into the wayback machine and find out:
ANYA LINGO Adaptive Performance: Being able to actively change the coverage pattern without rehanging the array. Resolution 2: Software that manages the coverage and directivity. Adaptive Healing: Each Anya element packs a mic to make sure its output isn’t out of whack.
MORE PLEASE, OF EVERYTHING
If we go back to the dawn of line array, it was recognised by acoustician Harry Olsen (mid-last century) that stacking speaker drivers produced pattern control. But it would take technological advances and some real ingenuity for line array theory to be truly useful in concert PA systems. Then L-Acoustics’ Christian Heil cracked the code, and devised the WST HF waveguide to produce full-range coupling. (Still confused? Have a scoot through Dave Rat’s box item.) Ever since, all modern ‘line source’ arrays have followed a similar approach, until now. It’s taken another technological leap to allow Anya to exist. HF devices, digital amps and processing have all either got: better, cheaper, more powerful, or all of the above. So whereas V-DOSC (with its 2 x 15s, 4 x 7s and 2 x HFs) has/had eight drivers, Anya has 22 — and each has its own amp channel and is addressed by its own processing channel. Crucially, of those 22 devices, 14 are its brand new (one-inch exit/38mm voice coil) HF compression drivers, loaded on a HF horn that expands to fill nearly the entire face of the enclosure.
NEW GEAR’S RESOLUTION
With its 2 x 15s, Anya is a large-format line array system but its ‘resolution’ is far greater than similar products. By which I mean, with all those individually addressed drivers, the analogy is somewhat like building your model out of Lego instead of Duplo — the degree of fine-tuning is vastly increased with Lego because the building blocks are smaller and you have more of them to play with in the space allowed. What’s more, with the extra ‘resolution’ afforded by the extra ‘circuits’ per box, Anya’s response is potentially a lot more natural. Traditionally, line arrays need serious DSP turbo-charging and a whole bunch of compensation elsewhere to get those two or three HF units in a box to be heard 100m+ away, while ensuring the overall response to sound half-natural. With Anya, the high resolution of transducers allows for the use of drivers that aren’t working as hard or being asked to work beyond their normal limits. “Gone are the break up modes of large format compression drivers,” says EAW’s CEO Jeff Rocha. Rocha also explains: “The basic un-adapted Anya acoustics package is outstanding,” he said. “So beyond the application of EAW Focusing [which irons out the speaker box frequency/phase anomalies], only minimal processing is required to deliver some radical changes to the shape of the wavefront. Despite the abundance of onboard DSP, Anya does not suffer from the tonal degradation associated with overly processed sound systems because the relative change from device-to-device is very small.” ROOM OUT OF THE EQUATION
These ‘radical’ changes to the wavefront mean you no longer need to splay the array to achieve the required coverage. Anya is hung flat. Any coverage alteration is handled electronically, and not mechanically. With the extra resolution, the precision you can bring to the changes in coverage pattern is quite extraordinary. Jeff Rocha again: “Anya tailors coverage so precisely to the audience area that the impact of room acoustics on fidelity is significantly reduced. Taken all together, these factors produce an infinitely scalable, large-format system that has the impact of a big PA but the fidelity of an enormous pair of studio monitors.” And look, I know he would say that. He’s EAW’s boss and one of the masterminds behind Anya and, as yet, not many outside of California have heard Anya in full flight. AT 33
One person who has heard Anya in action more than most is Dave Rat. Again, Dave isn’t totally disinterested. He’s good mates with EAW and was sold on Anya early. But he’s primarily a bloke who runs a rental company (Rat Sound) who can ill afford some vanity-led flight of technological fancy. RATTING ON ANYA
Dave Rat: I’ve worked with EAW on the Microwedge floor monitors. They licensed that product and they’ve been manufacturing and selling that, and the relationship has been great. So it wasn’t a complete surprise to be contacted about the Anya project. To be clear: nothing about Anya is my idea. EAW came up with the idea; they’re the brilliant MIT graduates; they’ve got all the technical stuff down… But when it comes to putting those ideas into the real world; making it interface with the human race… that’s what I’m good at. AT: What was your initial ‘real world’ perspective? Dave Rat: ‘That’s never going to happen!’ My next reaction was: ‘Sure, I’ll consult on that! What’s the worst thing that could happen? It’ll never come to fruition but it’ll be interesting to watch them try.’ Next thing I know: ‘Alright, we’re ready. Come listen to a box.’ I thought: ‘Holy shit! This thing is real?!’ AT: Are advances in component technology just as much responsible for Anya as clever thinking? Dave Rat: If what you’re saying is: the drivers have got small enough, light enough and loud enough; that the Class-D amplifiers got small enough, light enough and cheap enough to put 22 into a cab; and the digital processing costs have come down enough and are high quality enough… sure. All that created the mechanical ability for Anya to exist. But then they needed to create the software and processing in order to bring it to life. AT: With so much potential control, I’m guessing the software runs quite deep?
Dave Rat: EAW historically over-complicates things. And I think they were wary of making that mistake here. The software simplifies the process rather than opens a can of worms. Effectively, what you do is give the software the room dimensions and it will then distribute the energy as uniformly as possible and with as equal a frequency response as possible. AT: I guess the theory is: with great power comes great responsibility… or at least more ways in which the PA can go spectacularly wrong. Dave Rat: And there are some very clever fail safes. For example, every cab has infrared sensors. So Anya is ‘self aware’. The cab at the bottom of the array knows that, because it can’t sense another below it. And so on. The array is selfdetermining which mean the processing can’t go spectacularly wrong. There’s also a microphone in every box that allows it to do a self-test. The speaker makes noise and analyses itself and lets you know if there’s something wrong. AT: How will the ‘hang it, then steer it’ capabilities be so much more useful than model it/splay it/ hang it? Dave Rat: Here’s an example: Say you have Anya installed into a theatre and you’ve not sold the balcony tonight. Without moving anything you can steer the entire array to the floor area. Maybe there’s a late rush on tickets and they open the balcony in the middle of the show. No problem, you can steer the entire PA to include the balcony. Or, say you’ve got a festival rig that’s designed to cover 10,000 people in a paddock. But 15,000 show up. You can expand the coverage of the rig without lifting a finger. Or, say you’ve got noise complaints during an outdoor gig. You can reduce the long-throw portion of the PA in real-time while you’ve got the official on the two-way radio. All without compromising the show, or lowering/raising the PA.
AT: What does Anya sound like? Dave Rat: The short answer is: it sounds great. But I’ll tell you this anecdote about the sound during testing: We had the rig set up in a factory space. I was listening intently to see if I could hear anything odd; any aberrations. I’m extremely familiar with the sound of the Microwedge, so I had a Microwedge 15 set up in the same space as the two Anya boxes. Just for comparison’s sake we tried to EQ Anya to replicate as best we could the sound of the Microwedge. Without much luck. Anyway, we were listening to Anya, hearing it in different coverage configurations. Set to full 180°, where it was shooting straight up and straight down, it turned the space into a big echo chamber — it was almost unintelligible. And then I said, “Now just cover up to the mix position,” and all of a sudden the sound became clear and tight. The difference was dramatic and then I said, “Wait a minute, can you set the coverage of the Anya to match the Microwedge’s at 60° conical?” So the EAW guys set it up and I got goose bumps, because suddenly when the coverage patterns were mirrored, and even though the systems were unrelated in every other way, they sounded way more similar than when we were trying to match them tonally with EQ. I realised what we as engineers often think is EQ-based, and what our measurement systems tell us is EQbased, is probably just as much coverage based. It’s the way our brain interprets room reflections and the character of the room that’s being excited. And with Anya, being able to control its dispersion in real-time, well, suddenly that’s a capability you can’t un-hear — once you’ve heard it, you’re hooked. AT: So you’re saying that a PA will often compare more poorly/favourably because of how much or little it’s exciting the room, rather than any inherent ‘sound’? Dave Rat: Anya isn’t the only great-sounding
DAVE RAT: HOW ANYA’S HF SECTION CHANGES EVERYTHING Dave Rat: The question is: why can’t all line arrays perform in this way? What is unique about Anya that means it has this capability other line arrays don’t? Why can’t we hang a V-DOSC rig and address five different amps and put an amplifier on every driver all the way through and gain the steering control? The answer: EAW solved a very complex problem in a very unique way. I’ll give you an illustration: Say you have a column of 15-inch speakers all stacked up that you’re running from a low frequency of 20-30Hz up to around 120Hz. Next, you delay each speaker, from the top one down. The timing of the delay exactly matches the driver distance (centre to centre). The result of that is: the sound of the 15s pointing straight down. If you decrease the delay times you could have them all focus on a single, distant point. Keep decreasing the delay time and you’ll have the array point straight up. None of this is new, it’s basic beam steering. But as the wavelength get smaller, it gets harder. So, AT 34
let’s take those 15s again and change the upper frequency from 120Hz to 1200Hz. Instantly, you have a problem. The wavelength of 1200Hz is around one foot/12 inches, and the 15-inch drivers are at least 15-inches apart. The rule of thumb for the speed of sound is: one foot per millisecond. So let’s go back to delaying our 15-inch array downwards by applying a delay of the speaker step distance (15 inches), which is 1.5ms. Do that and you can produce an end-fire array down, but as you try and sweep that forward (like we did earlier) by decreasing the delay, you get 180° phase shifts, resulting in all sorts of cancellations. The problem: the drivers are too far apart — the time in milliseconds is too long versus the frequency’s wavelength. The rule of thumb for beam steering and control is max out at a time delay four-times the step length. Which means for 15-inch speakers you have effective control up to 120Hz. With 10-inch drivers its up to 300Hz. With eight-inch speakers you might
get up to 500Hz and 800Hz for a six-inch speaker. By ‘control’, I mean, at these maximum frequencies with these drivers you can point an array straight up or down and at no point are the drivers 180° out of polarity with the delay that you’re imposing. Like I said, that’s old news, that works with any line array. But what happens when you get to the HF drivers? At 12kHz you have a one-inch wavelength, which means you can’t delay any more than 0.1ms before you run into 180° phase shifts. If the maximum time delay between two consecutive stacked HF drivers is 0.1ms, the most you can do is point up 5° and down 5°, without running into all kinds of cancellation issues. Anya’s answer was to stack more drivers: 14 one-inch drivers, in fact. And at only one-inch apart, when you delay them by 0.1ms they shoot straight down — without any phase cancellation in the forward field. So, in summary, if the ‘mechanical resolution distance’ of the drivers is high enough, you’ve got yourself a full-range steerable system.
True To Your Music Wi-Fidelity
PA on the market. There are a number. But if we were able to eliminate reflections in a room, every room would sound the same and perfect. That’s where Anya comes to the rescue. With reflections, the sound system is one output, the floor is one reflection, left wall, right wall, ceiling and then rear wall. That’s five different versions of the system we’re hearing, then double that per hang. There are 10 different primary reflections showing up in a square-box-type room. In an arena, with all its concave surfaces, who knows?! So will people like the sound of Anya more than K1 or the J line? People have their preferences. But Anya has the capability to reduce one of the primary issues, unwanted reflections, more significantly than any other system we have, in real time, resulting in a better sounding show than we can get with other systems. OH… & ANOTHER THING
The gripe many have had about large-format line array (and why Nexo chose to pursue its modular STM rather than yet another big three-way box) is that even though they’re ideal for outdoor festivals, in most other applications they’re unwieldy and inflexible — you’re probably better off with a point ’n’ shoot PA. With most line array elements covering less than 15° in the vertical, you need at least five a side to cover an average-
sized room, when in every other regard that’s just way too much PA. Anya doesn’t work this way. If the room needs only two boxes a side, then hang two boxes. Or think of it this way: without the use of electronic beam steering, you need a good number of line array elements — say five or six — to get any of the pattern control goodness in the mids and lows afforded by line array theory. Put only a couple of boxes up and you might get highly directional horn-loaded highs and high/ mids but the lows are all-but omnidirectional. Not so with Anya. With its electronic beam steering and processing, you have full-band control. If EAW gets the package right — and let’s hope, given the company’s pedigree, that’s a ‘given’ — then it’s hard to see how system designers, installers and rental companies won’t immediately be sold on the benefits of Anya. I’ve no doubt it’ll be a pricey option, but you’re not compelled to buy into a hefty minimum to have any system at all — two a side would make for a very handy club system, for example. Is it the next big thing? Just perhaps. Anya will at the very least give the rusted-on EAW-philes — of which there are many — something to cheer about.
Epic digital wireless technology, now amazingly affordable. Introducing the DWZ Series with affordable 2.4 GHz technology. You get solid-gold, 24-bit linear PCM digital audio to keep you sounding your best. You get robust transmission and easy channel selection. Even multi-way powering and digital EQ are available. And Sony has pre-assembled DWZ packages for guitar/bass, vocals, wind instruments, presentation and speech. Sony’s DWZ Series. Sound like a million bucks without spending it. pro.sony.com.au/audio
ON THE CARDS
Based on Korg’s EDS-I (Enhanced Definition Sythesis – Integrated) technology, the Kross offers 512 pre-loaded sounds, 265 user programmable patches, 80-voice polyphony, a 16-track sequencer as well as arpeggiator, MIDI and audio recording (16-bit/48k PCM) and a range of insert and master effects.
Separate transport and record buttons for audio and MIDI recording and a handy tap tempo function round out what is a wellthought out and surprisingly comprehensive control set. You can also load MIDI song templates and quantise onboard Kross. And when you’re done with a song or idea, it can be stored on a Kross-formatted SD card.
A couple of rotary controllers give you access to sounds arranged in 10 pre-programmed soundbanks and one user bank. Backlit navigation buttons each side of the small, but well-lit LED screen, allow you to combine, edit and effect those sounds. And the usual suspects; pitch and modulation wheels, octave up and down buttons, and master volume are combined with comprehensive step sequencer controls, including a 16-button backlit sequencer array, for plenty of control.
BACK & A KROSS
The 61-key model is a lightweight portable affair weighing in at 4.3kg and measuring a svelte 91 mm at its thickest. And this is where the Kross comes into its own. While you can draw power from a wall-wart, Kross can also draw juice from six AA alkaline batteries to give you about five hours of operation. Underneath, there’s a nifty carry bar in a cut-away at rear of the keyboard. It’s the ultimate portable workstation.
Round the back things are pretty simple with audio outputs provided by a pair of unbalanced 1/4-inch jacks. You can record external audio via 1/8-inch stereo mini jack input or a non-phantom-powered 1/4-inch microphone input jack. MIDI to your computer is on a standard square USB socket.
KORG KROSS KEYBOARD WORKSTATION Korg workstation goodness in a package you can carry around town.
NEED TO KNOW
Review: Greg Walker
PRICE Expect to pay 61-key $899 88-key $1449 CONTACT CMI Music & Audio: (03) 9315 2244 or email@example.com
PROS Comprehensive sound library Flexible patch editing tools including effects Built-in sequencer, MIDI and audio recording Lightweight pick-up-and-go design
CONS Playing action on the 61-key model underwhelming Limited real-time hands-on controls
SUMMARY The Korg Kross offers excellent sounds and a load of extra workstation features in a compact, welldesigned package. A keyboard that’ll have you up and running in no time.
Korg’s little playstation comes in synth-action 61-key and weighted 88-key iterations, and while the Kross may look like a bit of a toy thanks to its lightweight frame, it actually packs a pretty hefty punch while covering a lot of musical bases. KROSS PURPOSES
The Kross offers a remarkably large palette of sounds that belies its diminutive frame. In contrast to a ‘stage’ keyboard like the Korg SV-1 (that utilises a small number of piano and organ sounds and then provides extensive tone shaping and modulation controls), the Kross gives you a ton of different ready-made sounds to play with — many of them based on its elder brother Krome’s sound library. The piano sounds are rich and varied and I particularly liked some of the non-standard tonalities such as the undamped and honky tonk settings. Likewise the organ and electric piano offerings provide a stack of different tonal options that should cater to almost every style and situation while the Mellotron emulations are very useable indeed.
Are your wireless mics ready for the Digital Dividend ?
Further afield are 49 types of tasty mallet and bell sounds including very handy vibraphone, marimba, steel drum and gamelan emulations, toy pianos and other tinkly things. The string and brass sounds are adequate but somewhat less convincing. The synth options will be a bit hit and miss depending on what you’re after, leaning generally towards the dance and atmospheric side of the spectrum. Drum sounds are very useable and range over a good number of styles, with the hip-hop and electro kits being particularly fun to play with. There’s even a bunch of sound effects hidden away at the back end of the drum section so you can throw a few FX curve balls if required. All in all it’s a great set of sounds for such a modest-looking keyboard, and while patch tweaking requires a good knowledge of the nested menus inside the program patches, you can spend some time with the Kross and really bend it to your will. Ergonomically the Kross works well, with all buttons and dials having a solid no-nonsense feel to them, though my main criticism is the playability of the synth-action keyboard. It feels just a little clunky on more complex material and adjustments to the MIDI sensitivity settings helped somewhat without totally putting my fingers at ease. TICKS & KROSSES
Korg has got a lot of things right with this little keyboard. The basic patch sounds are very useful and there’s a great variety to choose from. The onboard sequencing and recording functions will appeal to a subset of users who would use the Kross more as a standalone music machine, whereas others will find it a handy addition to an existing studio or live set-up where it doubles as a powerful sound source and MIDI controller. For live performance there’s an easy out-of-the-box playability to the Kross, though the sparsity of real time controllers may inhibit some. For serious live use I’d recommend the weighted 88-key model for a better playing experience. In the studio I regularly found myself using the Kross on overdubs that otherwise would have been taken care of by software instruments. Almost invariably the Kross sounds had the edge in terms of punch, character and playability. For the kind of film and TV work that I sometimes do, the range and character of the Kross was also a handy adjunct to the standard sound libraries I use. Overall the Korg Kross is easy to navigate, offers surprisingly flexible sound shaping tools under the bonnet and above all is a fun keyboard to play because it sounds good.
OURS ARE ! By the end of 2014, all analogue TV transmitters will be turned off and all digital TV transmitters will have changed frequency. The band between 694 MHz and 820 MHz will be cleared of all users so it can be used for mobile data services. Check your wireless microphone systems now ! If they operate between 694 MHz and 820 MHz you need to start planning to operate between 520 MHz and 694 MHz before the end of 2014.
Make certain your systems are ready! visit
digitaldividend.com.au for more information
NEKTAR PANORAMA P6 Studio or performance, Panorama is bristling with MIDI control. Reason users (in particular): feel the love. Review: Christopher Holder
CUBASE/NUENDO Leaving aside the Reason schtick for a minute, Panorama’s Cubase/Nuendo integration is sophisticated and Steinberg-ers ought to look into just what the P4 and P6 can do for them. There’s a huge list of plug-ins and virtual instrument maps, and when you toggle to the Panorama’s Mixer view you’ve got extensive control over the Cubase mixer (levels, EQ and an insert menu). As I mentioned, I didn’t run Panorama with Cubase but there’s no question that Nektar is toiling hard to provide deeper and deeper Steinberg integration.
Reason users have plenty of cause to feel smug. We’re producers of music from first principles. We have a grasp of the dots on a stave; we’re not afraid to dig into synth parameters and indeed flip them around for re-patching; we’re engineers, producers and musicians and we’re proud. [Pause a moment for high fives and fist bumps.] But we’re not feeling the love from those who make hardware controllers. Meanwhile, those trumped-up DJ wannabes, throwing loops together, not knowing their rituendo from their Nintendo, have a smorgasbord of possible external control.
NEED TO KNOW
Novation, Akai, Ableton themselves, Tenori and others have been indulging the Live community
PRICE P4 (49-note): $599 P6 (67-note): $699 P1 (Control Surface): $499
CONTACT Sound & Music: (03) 9555 8081 sound-music.com
with arrays of buttons and pots that provide these pseudo-musical dolts with hands-on control. Frustrating? It’s enough to make me smash one of my Stradavarii. But, fear not, fellow Reason scholars, help is at hand. A new and enlightened company called Nektar out of the US has clarity where others don’t. Nektar is a friend of Reason, and, as such, is your friend too. COMPATIBILITY
I’ve had the Panorama P6 over the Christmas break and it’s been a delight. Like an early Santa present, I plucked the P6 out of the box, popped the USB cable into the back of my Mac, booted up Reason, and with very little ado, was
PROS Deep, solid DAW integration – especially Reason & newcomer Bitwig Loads of knobs, buttons and faders ALPS motorised fader
CONS Short-throw, 45mm faders a bit ‘tacky’
immediately cueing up new sounds, tapping away at the drum pads, and tweaking levels like the natural extension of Reason I hoped it would be. Did I audition the Panorama with other DAWs? No. And I’m unapologetic. Not to say you can’t. Nektar can be adopted by other DAW tribes. The Nektar literature outlines compatibility with Logic, Cubase/Nuendo, ProTools, [gag] Live, or even as a generic MIDI controller, but the real juice is in its deep Reason control. INSPIRATION FROM CONTROL
I know I’m preaching to the choir here, but Reason’s sound banks run deep. I’m endlessly impressed by Propellerhead’s synths, and,
SUMMARY Nektar is a control specialist. The P4/P6 is a no-brainer for Reason users, while increasingly compelling for Cubase and Logic people. Can a ‘dumb’ MIDI controller be inspiring? Nektar proves it can.
okay, always lusting after the next Rack Extension. But I’m sure I speak for others here when I talk about cueing up sounds. Got a groove happening and auditioning leads? Often it’s not the original preset that screams ‘use me’, it’s how it responds to modulation, to how it sounds with extreme resonance, or a different LFO shape/depth. With the P6 I found myself, not only cueing up a new synth patch from the keyboard but instantly grabbing the P6 parameter pots to hear how the patch responded. Nektar makes this easy by automatically mapping parameters and providing you with different parameter combinations. Adjust to taste. The 3.5-inch TFT is bright and easy enough to read and the naming of the parameters pleasantly non-cryptic. It’s a joy to riffle the Reason sound ‘deck’ with the P6 and find what you need or a good place to start your own sound creation.
Going digital ain’t so bad...
Meanwhile, when you’re deep into editing a synth patch, for example, you might find a more arcane parameter (dunno, maybe the resonance on your Malström’s comb filter) is really driving the sound. It’s easy to map that control to the Panorama pots via the Control Edit menu. Ditto on/off toggle switches. Like that particular map? Save it as a preset. FADER MOTORING
That’s touching on the main compositional meat and spuds of the P6 but undoubtedly the show stealer is the long-throw ALPS motorised fader. For the fader to function you need more than PC bus power. But when you do plug the P6 into the mains it comes alive, literally — giving a little startup waggle to let you know it’s reporting for duty. Once a channel is selected in your Mixer section the motorised fader is all yours. I don’t know anyone who likes performing real-time fader riding with a mouse — either they’ve never used a genuine high-quality fader before or they’re too young to feel the first signs of mouse-hand RSI. The P6 fader feels great and any former, too-much-bother, reticence to ride my levels have evaporated. I’ll happily do channel-by-channel passes, fine-tuning my levels, enjoying the extra feel and resolution of the ALPS unit. SWEET NEKTAR
When I first clapped eyes on the Panorama P6 I knew I wanted one. It’s a sexy-looking keyboard with its high-gloss livery, and I like how the bottom note sits proud of the mod and pitch bend wheel section. In the same way that I feel a deep affinity to Reason, I knew Nektar understood my demands. Which is all a bunch of heart-over-head baloney really, given Nektar was untried, and programming deep, reliable DAW compatibility is a tough gig. Fortunately, Nektar has come through with the goods. Am I blown away by every aspect? Of course not. You only have to head to the Nektar forum to see the niggles any product (so dependent on software) like this experiences. And sure, it’s built to a price and the pots/pads/generic faders aren’t always the height of luxury, but crucially, the important stuff is all there — it’s responsive, the semi-weighted keyboard manual has a pleasant playability, and the Reason integration is all-but faultless. Are you, like I was, working with a generic MIDI controller, relying on your qwerty/mouse as much as your keyboard? Stop it! I urge all Reason users to rally behind Nektar and buy a Panorama. As for other DAW users? Well, if you must, but I rather like the fact we’ve got something all to ourselves for a change.
THE DISPLAY SAYS IT ALL A 3.5-inch colour TFT display acts as the nerve centre. There are four navigation buttons that take you to various modes. Mostly you’ll be toggling between Transport (where you can, for example, quickly dial up new left/right locators for looping — which saves so much faffing about); and the Instrument screen, where key parameters are mapped to the available pots. I like the way the screen and controls feel like they’re keeping up with your ideas… latency isn’t an issue. Spent hours working with your customised setup combo? Then save it as a profile — there are 20 ‘presets’ you can fill.
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The meter section uses a multi-coloured LED per channel to indicate signal level (green, yellow and red) and track arming (flashing red). These LEDs also turn solid yellow to indicate the pair of channels currently being monitored on the PFL bus and sent to the headphone output. The more detailed dual 12-LED peak meter indicates the level of the PFL bus.
MADI RECORDER & DIRECTOUT ANDIAMO.MC PREAMPS Remote recording? Remote control is the way to go.
NEED TO KNOW
Review: Matt Dever
PRICE JoeCo.MADI: $6580 ex GST JoeCo.MADI Player Upgrade Option: $580 ex GST DirectOut Andiamo.MC (SC/SC or SC/BNC): $11,735 ex GST Breakout Panels: 16 XLR analogue I/O: $495 ex GST 8 XLR AES3 I/O: $495 ex GST 16 AES on BNC: $995 ex GST 8 analogue I/O on D-Sub: $155 ex GST 8 AES I/O on D-Sub: $165 ex GST
CONTACT Pro Audio Technology: (02) 9476 1272 or firstname.lastname@example.org
PROS Huge track count in a small rack High quality audio Recording directly to a portable HDD is convenient DirectOut gear built like a tank CONS Audible clicks when changing Andiamo gain
SUMMARY JoeCo’s reputation for building rock solid recorders, and DirectOut Andiamo’s road ready build quality, makes for an extremely reliable remote recording setup. Not only that, you can control each remotely. Hello, under-stage recording rigs.
NEED MORE CHANNELS?
Dual RCA connections, labelled Ext Clk, form an S/PDIF link that passes control data and timecode information between BlackBox recorders, allowing a number of units to be daisy-chained and sync’d to achieve even higher track counts.
The 64 inputs and outputs are accessed via dual MADI connections; 64 I/O via optical connections and 64 I/O via coaxial connections. This allows the recorder to be used with a growing range of digital consoles, audio interfaces and preamplifiers that feature the MADI standard. Even when a digital console does not come with MADI straight out of the box, it can often be installed with an optional I/O card.
The idea of having 64 channels of standalone recording in a 1U device is a relatively new and exciting one. So the idea of strapping 64 broadcast-quality microphone preamplifiers to it, fitting it all into a 10U case and controlling it with an iPad is just crazy, right? Not if you can get your hands on a JoeCo BlackBox and a pair of DirectOut Andiamo.MCs! Lucky for me I was sent the BlackBox MADI recorder and an Andiamo.MC just in time for a gig with a lot of channels and a spare multicore split available. Let the games begin! NAVIGATING THE BLACKBOX
JoeCo’s BlackBox Recorder has been around since 2009, with various versions available that differ in I/O connectivity. The MADI version on review here is one of the newest additions to the family, more than doubling the track count of its older siblings. It’s a lot of channels to fit into 1RU, and what’s more, the whole thing weighs 2.3kg. While it initially seemed a little lightweight compared to the stocky Andiamos. It’s the perfect travel companion. Working with MADI is very straightforward, especially for anyone who has used ADAT or most other digital audio standards; simply decide which device you want to be the master and set the others to slave. The clocking is generally looked after by the MADI connection – configuration done! As far as recorders go, the BlackBox has a fairly unique user interface. A touch-sensitive jogwheel and seven buttons to the left of a small,
The analogue D-Sub input around the back provides eight channels of analogue balanced line inputs. These eat up eight of your MADI channels, but are designed for patching in local audience mics when using legacy MADI consoles which only allow 56 channels of MADI.
A headphone output monitors the inputs either in mono, stereo, or an internally generated mixdown (with individual level and pan on every channel). The source and output level is controlled from the jog-wheel and Back button on the front panel. It’s worth reading the manual to get the most out of the monitoring.
full-colour screen. The jog-wheel and buttons provide control over all the parameters on the device, including all menu functions and headphone monitoring options. The main screen displays useful information regarding the current file, remaining disk space (displayed in time rather than gigabytes, which is handy) and clock synchronisation or lack thereof. Hitting the Menu button allows you to delve into the settings of the device, such as track arming, synchronisation settings and disk utilities. The feel of the touch-sensitive controls is similar to a smart-phone, or perhaps not quite as sensitive. The system generally works well, however damp or greasy fingers can make things tricky. The saving grace here is that once you have everything set up, there’s little need to access the menu. BLACKBOX REMOTE
The test rig was shipped with the JoeCo Remote hardware, which essentially creates a Wi-Fi hotspot to connect an iPad to the BlackBox. The Remote app is very elegant; it provides high-resolution metering of all 64 channels simultaneously, which you can’t get on the unit itself. It also allows wireless control of the recorder, including all settings and transport controls. I used the iPad app during the test recording, where I was also mixing front of house. The app put all of the recorder controls and metering within arm’s reach, allowing me to focus on the mix and occasionallyglance at the recorder… great! Setting up the remote was simple enough, however the iPad needs some specific network
settings in order to function, which is where the manual came into play. Once set up, the BlackBox essentially commandeers the iPad. You have to be connected to the BlackBox network to control the recorder, so you need to set up proper network infrastructure to flick between your BlackBox app and your digital console control app. The Remote links to the recorder via a serial cable connection, but also requires a separate DC power input. Placing the Remote as high as possible helped maximise the Wi-Fi range, so it was best to leave the unit loose in the rack ready to position it high. However, the two cables always got tangled and its reach was limited by the length of the power cable. It would make for a much sleeker solution if the Remote unit could be powered by the transmission cable. According to JoeCo’s Joe Bull, this was an acknowledged tradeoff. While he agrees the unit would have been sleeker if powered from the 9-pin cable, it would have required existing users to return their units for an upgrade, as standard 9-pin connectors don’t provide power. Knowing how crucial JoeCo’s recorders are to its customers, as it would have been too much downtime, and a great inconvenience. CRASH TEST!
When I use a recorder in a live situation, I always like to know what happens if power is lost while recording; I need to know that the files will be recoverable if something goes wrong. I was pleased to find the BlackBox deals with this situation very elegantly indeed: it’s called Safe’n’Sound Record Recovery.
The Andiamo.MC I/O is laid out very nicely considering the unit packs 32 preamps. There are eight D-Sub connectors for the analogue I/O, dual redundant MADI connections (available in either two optical ports, or optical and coaxial), dual redundant power inlets, wordclock I/O, a USB port, and finally a serial port labelled GPO.
The General Purpose Outlet port was really the only thing out of the ordinary on the back panel. It’s actually a pair of remote switchable 12V power outlets that allow you to, for example, switch a ‘recording’ light on and off from the Andiamo Remote software. Speaking of powered pairs, dual redundant power supplies only adds to the Andiamo’s impeccable quality.
If the unit loses power, or even if the disk is unplugged while recording, the files will remain on the disk but they will not have been closed properly. The files will appear as 4GB each regardless of length and won’t work in many DAWs. Once the BlackBox is powered up again, or the disk is plugged back in, Safe’n’Sound will automatically detect that the recording wasn’t closed properly and repair the files. DIRECT IN WITH DIRECTOUT
The BlackBox MADI recorder is perfectly complemented by the selection of boxes from DirectOut Technologies; including analogue (mic or line) to MADI, AES to MADI and ADAT to MADI. The test rig came with an Andiamo.MC, which is the 32 I/O microphone preamplifier version. The gear from DirectOut has that ‘built to last’ look and feel to it. The Andiamo is heavy and void of anything that can break off or be easily scratched. The dual-redundant power supplies are another clear indication of the build quality of these units — a very impressive feature indeed! The front panel is almost purely for visual feedback, featuring 137 LEDs to indicate everything from input signal level to clock sync information. Four buttons provide control over the whole unit by cycling through each function, however this is not terribly user friendly and AT 42
can be quite time consuming to use if you have to change a lot of settings. I also noticed that adjusting the gain of a preamp using the front panel buttons actually feeds an audible click through the channel. This is where the Andiamo Remote saves the day! ANDIAMO: HERE WE GO
The Andiamo Remote software works on a different principle to the BlackBox Remote software. The Andiamo is attached via USB to a PC loaded with the Remote software, which controls the device — no fancy Wi-Fi or iPad app here! The software takes some concentration to install and set up, however the instructions are simple and there were no issues with my install to a Windows XP netbook. Once installed, the software provides you with fast and comprehensive control over all settings on the unit, such as channel gain, pads, patching and phantom power. You can also control utility settings such as clock source, MADI format and even the fan speed. I will point out that, unlike the physical buttons, changing the gain using the remote software does not produce any audible clicks through the channel. If I owned an Andiamo, I would definitely keep a little netbook in the back of the rack to control it. I was even able to control the unit with the iPad by using a VNC app to control the netbook!
The Andiamo came with a complete set of XLR breakout panels, giving 32 inputs and outputs on XLR connections. The breakout panels are available in a few different configurations and are also of high build quality to match the main unit.
DirectOut has done well to fit 32 microphone preamps into a 2U case, but how do they perform? The results of my test recording were very impressive, however given that it was a noisy rock concert, I can’t be as critical of the sound as I could if I used the unit in the studio. Something that did impress me was on an acoustic guitar DI channel. It was padded for the main act but the support acts acoustic was much lower in level — to the point where it almost didn’t meter at all in my DAW. I was able to gain this channel up enough for it to be usable, which I think says a lot about how quiet the preamps are. A SOLID TEAM
The BlackBox MADI and Andiamo.MC team up perfectly to create an incredibly powerful mobile recording system. There aren’t too many situations where you need 64 channels to record a live gig, however, it certainly opens up some possibilities. Something that comes to mind is recording multiple stages at a festival to the same recorder. MADI cables can be long, so you could have an Andiamo unit at each stage running back to the BlackBox at a central location. I was more than happy with the performance of both units, although I was more impressed with the build quality of the Andiamo.MC overall. Both units are expensive, but you really need to think about it on a ‘per channel’ basis to be fair.
LAST WORD with
Wally De Backer & Franc Tetaz: Recording Studios & Gear Lust Photo: Hugh Hamilton
Franc: L.A. is the same as anywhere else in the world; people don’t have the budgets they used to. I was talking to [iconoclastic engineer/producer] Nick Launay the other week. He’s worked in the same L.A. studio for many years, and he said, ‘That’s it, I’m giving in, I’m going to work from home.’ Something he’s never done before.
The Grammy-winning Wally De Backer (AKA Gotye) life turned upside down with the global success of Someone That I Used to Know. He now has time to work on new music from the studio he’s completing on his parents’ property in southern coastal Victoria. His coconspirator, producer Franc Tetaz, rode the success of Someone to move home to L.A. where he works as a freelance producer. He maintains Moose Mastering in Melbourne. Last Word will play host to Wally and Franc’s ramblings about the future of audio for the next few issues.
I’ve been doing some work at Capitol and a house engineer told me he had two days last week where his job was to watch two guys sit on laptops. They could be doing that anywhere! Wally: That’s the future of the commercial studios. A big empty room with a laptop and a desk! Franc: We can laugh but for many forms of music, this is where things are headed. But I’m pretty pragmatic about this stuff. I’ve never had a recording studio per se because I wouldn’t use it enough to cover its overheads. It seems a waste of money to work hard just so you can have a huge amount of gear lying around. I’m better off keeping my kids in new shoes. Saying that, if I use a certain item of gear every day, yes, I should have it — otherwise it’s a luxury. Wally: I’m slowly making my space more workable, more practical. Some of those immediate things I’m combatting are temperature fluctuations through the seasons in Melbourne which aren’t great for gear, and exposure to external noise. I’m running out of space pretty quickly, but that’s because I’m buying too much gear, in contrast to Frank’s minimalism. In the end it’s my MacBook Pro, with Ableton Live and ProTools, that represents my essential rig for recording, editing, arranging and mixing — it’s indispensable. I’ll happily use mics as they’re available and work with whatever preamps or space is available. For example, I did some recording with Nick Launay, and he swears by his Beyerdynamic M88 as his go-to dynamic mic and always carries one or two of those around. But that’s because he’s so familiar with the sound of that mic and what it brings to the session, and he can’t be without them. I’m not at that point that I need a toolkit of mics and preamps always in my back pocket. I’m still quite happy to take that approach of: ‘Can I make it sound how I want with what I have?’ If I can’t, then sure, maybe I have to go somewhere else.
and hearing something that ‘sounds fine’. The experiences I’ve had in super-schmick studios have tended to be a bit bland. It sounds fine, but there’s not something about it; something truly interesting. Wally: I think there’s something in that for home studio producers. You’re far more likely to finish with something interesting when you’re pushing the workable boundaries of budget gear — and I think ‘interesting and gnarly’ trumps ‘pristine and bland’ every time. Franc: My laptop now provides me with the most powerful studio setup I’ve ever had. When I think of my plug-ins — I’ve got a suite of UAD plugs that sound awesome, and I’ve been using them a lot — I’m amazed by how technology has come such a long way. My setup comprises a laptop along with a chassis, and I’ll have a couple of my favourite microphones and preamps. That combination, with a great pair of monitors, gives me a hugely powerful setup, and what’s more, it’s not tied to a physical space. Wally: But, here’s a question for you Frank. Until you have that 10-strong Stylophone collection, will you really have it all? Franc: No, I’m not going to have it all. But where I want to be is in a place where it’s all about creativity. Knowing the sound I want for a song and not actually needing to grapple with the gear to achieve it — pure imagination. Wally: I’m not sure human interfaces have progressed as much as they could have. When I’ve had the chance to play a Yamaha CS-80 [classic mega-synth] I love the ribbon controller on that keyboard and the expressiveness of it. So between that and having a play of things like an Ondes Martenot [early electronic instrument], it really made me realise just how limited most keyboard-based contemporary synth technology is in terms of giving you options for expression as a musician, rather than forcing you to think like a producer. Even a theremin with its voltage control… there’s a lot you can do waving your hand in the air in a magnetic field as opposed to having your hand on a knob, however big.
If anything, my growing collection of gear is all about the trimmings. I’m really intrigued by interesting sounds and equipment that’s fun to come to grips with. None of it’s indispensable, yet I’ve just gotta have it!
As far as innovative interfacing goes in software, I was pretty excited about the potential of the audio-to-MIDI conversion Ableton has begun to incorporate into Live 9. I’ve tried it on a few things. If I remember correctly, I was trying to render Steve Gadd’s drum lick from 50 Ways to Leave Your Lover on a Roland 606. It wasn’t doing it particularly successfully, but it was interesting trying!
Franc: I love studios or spaces where I’m just a bit uncomfortable or pushing hard for something I need — it always ends up more interesting than just turning on a switch
Next issue Wally and Franc will explore the various inadequate ways producers make money from music… and why Wally isn’t a ‘gazillionaire’.
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