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Visit www.turramusic.com.au • Call (02) 9449 8487 • Email hitech_sales@turramusic.com.au AT 5
COVER STORY
14
Recording the Teskey Brothers Live
ISSUE 64 CONTENTS
10
Sampa The Great Live
Beyond Music: Sound Recording for VR
20
Quick Mix: Chris Tanaka Canwell
Sennheiser HD25 Closed Back Headphones AT 6
44
Small Island, Big Song, Huge Story
8
30
Austrian Audio OC818 Dual 38 Diaphragm Microphone
Antelope Audio Discrete 8 Synergy Core
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INTERVIEW
THEQUICKMIX
with CHRIS TANAKA
CANWELL
Who are you currently touring with?
My main gig over the last few years has been mixing FOH for Vintage Trouble out of Los Angeles. We’ve toured all over the world, and most of my other work fell by the wayside once we teamed up! I also double-dipped as their Tour Manager for a while which was a great learning experience. Who are some other acts/bands that you have worked with?
The Jacksons are another group that I do FOH for when they play the States, as well as Evan Ross and his wife Ashlee Simpson. I also spent a few years as the house monitor guy at the Knitting Factory back in my hometown so I’ve had the opportunity to work with a bunch of bands, including The Neighbourhood (also from Los Angeles) who took me out on my first tour back in 2014. How long have you been doing live sound and what was your path to a career in audio engineering?
Coming up on 15 years now! Like a lot of engineers, I played in a band, tried to figure out how to record an album, and everything snowballed from there. I also attended a lot of bad sounding concerts and wanted to understand why they didn’t sound better.
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What is your favourite console and why?
Yamaha CL5. Not the flashiest choice, but in my opinion all of the current pro digital consoles sound good enough to mix a great show on. I’m typically mixing on the house desk with far less prep time than I’d like, so fast workflow is much more valuable to me than a slightly more pleasant tonal quality. Honorable mention to Avid’s Profile for similar ease of use, and Digico’s SD range for best overall sound! Have also heard great things about Yamaha’s new PM10 and Midas’ Heritage-D but haven’t gotten to try them yet. Favourite microphone or any other piece of kit?
For studio work, the most exciting mic that comes to mind is the Slate VMS... The flexibility of being able to switch between that many great sounding mics and preamps without committing is a game changer for me in the studio. It’s like having the flexibility of a guitar amp simulator, without all the tone-related downsides. As far as other pieces… for live work, the AmpClamps and the Kelly Shu kick mounts make life easy for me, for the drummer/ guitarist, and for the house crew. Most memorable gig or career highlight?
Montreux Jazz Festival was incredible. There was so much technological innovation and attention to detail, not only on the audio end but in the
whole experience for everyone involved. Hellfest was a dream come true for me as well; I got to see so many of my favourite bands, and I’ll never forget the site of 100,000 French metalheads enthusiastically singing along to a Vintage Trouble ballad when we played between Municipal Waste and Gojira! Finally, I did a show with Ashlee Simpson and Evan Ross at The Roxy in LA with Evan’s mom Diana sitting in the front row smiling. Definitely one of the most surreal and heartwarming gigs I’ve been a part of. What are three mixing techniques you regularly employ?
I love using delay whenever appropriate, and most modern consoles make it easy to tap in the proper subdivision for the repeats so I’m always tapping out the tempo and riding the sends/returns/ feedback level, almost like a nervous tic. Sidechain compression is a newer technique that I enjoy using whenever I’m on the right console for it; it really helps the vocals and low end sit right when you’ve exhausted all other options. Playlists for each song can be really useful when you get to mix the same band on the same console for a good amount of time. It’s easy to overdo and lose perspective, so I will usually only have the playlist to affect the release time on the master compressor, so that it grooves with the song tempo,
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audiotechnology.com and to cue up the specific FX for that song. By making those two things automatic and precise, it frees up precious mental energy that can be applied to fixing other potential issues throughout the show.
& YOU COULD WIN
What are three pieces of gear or features that have come out and been game changers for you?
Hate to say it, but the release of the X32 was a big one for me. It was the first digital console that I really got to know, and it opened a lot of doors for me. I liked being able to throw it in my car and take it to local gigs, and also use it as an in-ear (and occasionally FOH) console for the band that I play in. Around that time I started using tablet control regularly on consoles that support it, which was a huge game changer for tuning speakers. Being able to tweak parameters while standing in an ideal listening position is amazing! Lastly, the RF Venue spectrum analyser looks to be a massive game changer, although I haven’t actually picked one up yet! Unless you use one, or are at a big festival with a wireless coordinator (who is probably using one), choosing the frequencies for your wireless gear feels like you’re shooting blindly. I hope to have one next time I do a monitor gig where the band is on ears. How have your working methods changed since you began live sound mixing?
I’ve had enough time to develop a solid workflow, thankfully, so that most of the typical audio problems I used to have to deal with are solved before they materialise. More energy can be put into making sure that the show runs on time, that the bartenders can hear their drink orders, and that the other crew members and venue staff are having a good day. They’re all things that can get overlooked when you’re staring down at the console the whole time. Any tips/words of wisdom for someone starting out?
Keep an open mind and understand that you don’t know everything. There are valuable audio-related lessons to be learned every day, whether you’re stagehanding for a corporate gig or watching a local show in a tiny bar. For those looking to tour, I would recommend taking a good look at yourself and making sure that you’re a person who people will want to be around day-in and day-out. Find out what your worst traits are and work on them. Find a band whose music you can get into, whose members you can stand to be around, and really become a part of the team. Finally, to those engineers who are already touring – don’t double-dip as Tour Manager! It’s a great learning experience and everyone should do it briefly, but your mixes will suffer (as will the audience experience), the artist won’t be properly taken care of, and you’ll get burned out. There are many talented TM/FOH people out there and I respect what they do, but that’s my opinion and I’m sticking to it!
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FEATURE
It was a joyous and triumphant homecoming for an extravagantly-gifted hip hop talent.
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It’s strange to call it a ‘homecoming’ but for Sampa The Great, her Melbourne show at the Forum was just that — a hometown celebration. The Zambian-born rapper was raised in Botswana and jetted to California to study music for film, before moving to Sydney to pursue an SAE course in audio production. But since 2018, Melbourne has been home. Teaming with producer Alejandro ‘JJ’ Abapo to make ‘The Return’ LP, released late 2018 and garnering a Best Hip Hop Album ARIA nomination along the way. Sampa has toured the world. Mostly she travels light, flying in and out of festivals and short tours. There’s been a commensurately frugal approach to production but Sampa pulled out all stops for the sold out Forum show. Joining the core group of keys, guitar, bass, drums and BVs was a three-piece brass section, choir and guest artists. In all, front of house engineer, Dale Hughes had 67 inputs into his Digico SD10 and monitor engineer Adam Kaye had 24 mixes out of his SD10. With so many open mics, Dale was conscious he had to be careful in the forum — a venue he knew from previous gigs he’d mixed. “That was the reason for spec’ing KSL from JPJ Audio,” noted Dale. “With the KSL, and with everyone on ears it made for a really tidy stage sound.” QUIET PLEASE
JPJ’s new d&b KSL PA is voiced to sound much like its V and J Series brethren but has the big drawcard of superior rear-of-cabinet sound rejection — it’s eerily quiet behind a KSL hang. The result is an almost jazz-like quality to the FOH mix, where you can hear the air around the
instruments and vocals… albeit with a hip hopfriendly low-end that could raise the dead. “It’s all about the rapper in hip hop,” observes Dale. “Sampa is just rock solid. I’ve worked with Sampa and her band for two years, so I know every single word. I know exactly when she’s gonna yell or whisper or recite poetry or turn on a funny voice.” With the emphasis on Sampa’s vocal delivery, Dale spent time ensuring she had the right vocal mic. With Sampa’s full vocal delivery Dale knew he was after a dynamic capsule and dutifully picked up a Beta 58 as the first port of call. But after demo’ing the 58 and Beta 87, he gave the Shure KSM8 head a go, along with a Sennheiser option. Finally, Dale tested and settled on the Telefunken M81 capsule sourced from Travers Chesney Audio Services in Melbourne. “The M81 capsule achieves what I was looking for, with a low proximity effect. It’s a dynamic capsule, because Sampa’s loud and I don’t need a condenser picking up everything on stage. The M81 is bright — it’s a bright sounding mic in her ears — but I don’t find it harsh compared to other mics we’ve tried.” VOCAL DELIVERY
The show uses Shure ULX-D wireless, piping audio into the mixing console via AES digital. Dale uses a Rennaissance Compressor plug-in from a Waves pack rather than the desk’s onboard compressor: “I’m using the R Compressor to get some extra level on the lower-level parts but it’s not harsh sounding to my ear when she really does lean into it.” From there, the vocal goes into a Waves F6 dynamic EQ. “The F6 EQ is especially handy for getting the vocal sound I’m after in smaller venues,
with smaller PAs. For a larger gig such as this, the F6 is just doing some low mid attenuation. When she’s speaking it leaves more of the low mid in there, yet when she really puts some level into it, it pulls the low mids back a little.” The vocal then hits the Waves C6 which is doing some de-essing: “I’m just using the two floating bands to grab whatever is jumping out in certain songs — nothing too crazy.” Vocal clarity is key for Dale. He keeps Sampa’s channel locked in on the last fader of the console — he always knows where to find it and works that fader plenty. “It’s mainly a balance thing. I’m not so much into savage EQ — I try to let the vocal breathe and sound as natural as possible. In fact, I like to use less channel EQ and more system EQ. I like to get the PA sounding right then everything should sit in the mix with a HPF and maybe a little notch of 1dB here or there. “I like to use reverb as a space tool more than an effect,” continues Dale. “So I’ll try to place my backing vocals, for example, in a nice little package behind the main vocal. “I adhere to the old school method of determining how loud I want the vocal to be and work backwards from there — build my mix underneath that. That’s about a balance. “I like to mix the show and not rely on automation. I know the moments when I need to really push the guitar or bass. It’s a dynamic mix. I’m not ‘snapshotting’ every single song, locking myself into a box. I know how I want each song to sound and I use the faders to get there.”
DRUMS The drummer Sensible Jay doesn’t smash the kit, providing plent y of light and shade with his jazz-inspired breaks. The warped, multi-layered cymbal in the middle sounds uncannily like a TR808 clap sound — “this kit is like a real-life SPD!”. Dale mics the cymbals with Shure SM57s rather than using a pair of overhead condenser mics.
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KEYS Dale Hughes: “We’ve added the Rhodes for this show. It’s not something we’d normally tour with . Being a sold out hometown show, we thought we’d spring for a Rhodes.” ‘Spring’ being the operative word: “The Rhodes’ spring reverb amp does what spring reverb amps do when you jump on a riser they’re sitting on. We’ve now isolated that problem — that was a little learning curve for the morning!” The core of Alejandro ‘JJ’ Abapo’s sound is the Nord Electro 4. He’s added the Yamaha Montage 7 stage keyboard for this tour: “It’s been a great addition. Everyone’s eyes lit up when they were hearing the Montage 7 in their ears during production rehearsals.”
IN EARS Dale Hughes: “Moving an entire band to IEMs for a tour is a pretty daunting task. People are often brought in specifically to handle that move. [Monitor Engineer] Adam has done a really, really good job transitioning everybody. Sampa has been on ears for longer. I encouraged her to adopt in-ears to deal with the unpredictable nature of festival monitor setups and operators.” Adam Kaye: “This show was a challenge. We had guest vocalists who hadn’t sound checked. So I pulled up a nice-sounding mix as a starting point: kick and snare for them to follow; bass guitar for a full sound, lots of their vocal and some BVs to help them pitch. I’ll put a medium hall reverb on their vocal and some crowd mics as well. “Sampa likes a bright in-ear mix. I give her that with some 6k, but I’ll use a dynamic EQ to ensure her vocals are never harsh in her ears, when she’s really belting it out.”
MONITOR WORLD Monitor engineer, Adam Kaye, has a business called Compact Audio Touring Systems (CATS) which supplies the tour’s mixing and monitoring package. In most venues this involves a Digico SD11 front and back with a waves pack at FOH. Adam has an Australian distribution agreement with Dave Rat’s company SoundTools, which makes stagebox systems using Cat5 cable, saving on the weight of heavy snakes. “We have a 48-channel patch that packs into one Pelican and weighs only 28kg,” notes Adam. In this case SoundTools’ system runs analogue audio down Cat5 – three lines of Cat5 for the 12-way boxes. AT 12
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antelopeaudio.com AT 13
FEATURE
The Teskey Brothers are known for the sweet old-school sound of their recordings. AT’s Greg Walker catches up with them at Melbourne’s Forum, while recording their shows with three different recording rigs — including an allanalogue rig… Story: Greg Walker
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Why not go to analogue tape as well for a possible vinyl release?
Walking into the Forum Melbourne on a cold Spring afternoon I’m looking for Naomune Anzai (Nao to his friends), production manager and FOH mix engineer for the Teskey Brothers. Recently renovated and restored to its original beauty, the Forum’s inside is resplendent in gothic glory with Greco-Roman spires and turrets looming against a starry dark blue ‘sky’ – reminding me yet again of what a special venue it is. Underneath the proscenium arch this evening something equally special is taking place: the Teskey Brothers are playing a fourth sold-out show, and behind the scenes there are no less than three recording rigs capturing every note – one of them recording to analogue tape. If there’s any band that can take on this challenge successfully, it’s the Teskey Brothers with their hard-wired classic soul sound and passion for old-school sonics and recording techniques. Nao Anzai is a key figure in the Teskey Brothers story, having shown Sam Teskey how to get the best out of a Tascam eight-track tape recorder many years ago, and thereby firing his interest in analogue recording. While acting as a mentor in the early days, Nao is very much steering the overall Teskey production ship and brings a flexible yet purist approach to FOH mixing. He specifies a fully analogue Midas Heritage console for as many shows as he can, including this run of dates at the Forum. I arrived before Nao and had the fortune of taking a tour around the stage with Sam Teskey, looking at the numerous mics in use. After that, we ended up in the small makeshift control room where Alex Bennett had set up the analogue-only recording rig. We were soon joined by Matt Duffy, who was running the main digital recording system, and, of course, Nao Anzai. As we began what turned out to be a long conversation about gear, recording techniques and the joys and challenges of tracking to analogue tape, I was struck by the warmth and respect in the house. These people obviously enjoy each other’s company and feed off each other’s skillsets in a very positive way. It seems like a perfect blend of personalities with Sam’s enthusiasm and know-how, Nao’s experience and deep technical knowledge, Matt’s proficiency in location recording, and Alex’s fearless commitment to an all-analogue approach. There’s
a lot to discuss and also a lot of practical ‘runson-the-board’ know-how to tap into... THREE RECORDING RIGS
While it’s not unusual to record a live show these days, there can’t be many bands (or crews for that matter) up for the challenge of running three rigs at once; let alone one of them using the one-inch eight-track analogue tape format... Greg Walker: Why are you using three recording rigs? Sam Teskey: We initially planned to record the Forum concerts digitally from the live mics through the FOH console, to accompany the filming we were doing. There were some compromises involved in that approach, however, so we decided to use it as a back-up rig and bring in a separate digital system dedicated to recording for the filming. That rig takes a split from the live mics. It’s helpful for the filming because we can make a really hi-fi mix from it and sync and edit things easily. Greg Walker: What were the compromises with recording from the FOH console? Nao Anzai: It’s mostly about the track count... From the Midas Heritage FOH console I’ve got a total of 16 tracks going into Reaper through a UAD Apollo Quad interface, with a Presonus Digimax providing the extra ADAT inputs. I’m running four channels Direct Out of the console, along with 12 group outputs that include, of course, some submixes. I’ve also got a pair of Røde NT3s at the back balcony for room mics, taking my total track count up to 18. Matt Duffy: In comparison, I’ve got every line off stage coming into its own Pro Tools channel, along with my own room mics. I’ve got 26 channels going into two Focusrite Clarett interfaces, and another eight channels (mainly spares) coming through an old Zoom R16 interface. So all up I’ve got 34 channels of recording, with no groups or submixing... Greg Walker: What made you choose the Claretts, Matt? Matt Duffy: There’s not that many decent interfaces on the market that have lots of channels; there’s a big gap between the portable rigs and non-moveable studio set-ups. I do a lot of location recordings where I’ll go out to a house in the middle of nowhere with a band, and the Clarett did
a great job – the preamps are really good. So when I wanted to expand my set-up I bought another one. They’re really stable and easy to set up together. Greg Walker: Sam, why did you decide to go with the analogue recording rig as well? Sam Teskey: We’d already decided to have the two digital recording rigs for the purposes of the video, as we’ve just talked about, but from a purely recording point of view we thought “why not go to analogue tape as well for a possible vinyl release?” And instead of splitting all the lines again for the analogue recording rig – which would have meant about four different splits from the live mics [FOH, monitors, digital recording rig, analogue recording rig] – we did it the old fashioned way with totally separate mics and lines for the analogue recording rig, and doing things like taping two vocal mics together. It looks like the old Jimi Hendrix live footage with all these recording mics taped on top of the PA mics! Alex Bennett: And because the analogue rig is using totally separate mics, I can choose mics and placements specifically for the desired sound without worrying about feedback or spill. Greg Walker: Is the analogue recording rig sharing any of the live mics? Sam Teskey: There are a few extras like the banjo and acoustic guitar that only appear for one song so we take a split off the live mics for those things, but that’s about it. We decided that having one or two more mics on stage just for those short moments might be one or two mics too many... Alex Bennett: I’ve only got eight tracks for recording, but I’ve got 24 lines coming off stage. I’m taking 12 of those lines into an old ‘70s Chilton console and submixing them to four buses to record on to four tracks: drums, backing vocals, horns and keyboards. The bass and the lead vocal go directly to their own tracks, while the guitars are mixed and matched through preamps to the last two tracks. Sometimes I have to do some fast re-patching for the banjo and other things that only appear in one song. Matt Duffy: He’s mixing some things down to one channel, re-patching things between songs and also changing reels. He’s really been working flat out! Sam Teskey: Every night we’ve been changing the setlist to keep it pretty fresh for the band, but that’s been a bit of a headache for everyone on the crew. Before we go onstage we’ll tell Alex “we’re AT 15
I’ve only got eight tracks for recording, but I’ve got 24 lines coming off stage
going to change things a bit so you’ll have to repatch an instrument or change a reel here instead of there.” We add ‘reel change’ to the set list so we all know when he’s changing a reel and can give him time to do it. It’s been challenging, but we’ve become a tightly oiled machine. ANALOGUE RECORDING RIG
When the Teskey Brothers decided to bring in an analogue recording rig, they decided to go very analogue – right down to using vintage consoles, preamps and compressors, custom-made equipment, analogue tape, and signal paths full of tubes and transformers... Greg Walker: Tell me about the analogue recording rig, Alex... Alex Bennett: A lot of the gear here is mine. We hired the 12-channel Chilton QM3 console off a guy who was selling it. He was happy to
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hire it to us, and it’s perfect for our needs. The rack on the lower left was wheeled straight out of Sam’s studio, and I also grabbed a few other things from his place. I’m using his Chandler preamps on the guitars. Sam Teskey: I love the Chandlers on guitars. In the studio I set it so it hits the tape hard and when you really rip into the guitar you can hear the top end break up a little. The other thing I’ve been doing with the Chandlers is plugging the guitar straight into one unit, getting that real direct electric sound and then patching it straight into the other channel, then into the tape machine. You get some really crazy, huge and weird sounds doing that, and you’ve got so much control with the EQs. Greg Walker: What about the Giles Audio valve preamp? Alex Bennett: That’s on the bass and it’s going into the Giles Audio valve compressor. It’s quite
a slow compressor, but I think it’s got about the right timing for most of the Teskey Brothers’ songs. I’m slowing it down maybe one or two clicks for a couple of songs. Sam Teskey: It’s also really handy because you can turn off the compressor circuit and just use the extra gain stage. It’s a handy tool in the studio. It’s really quick and easy to just click it off and crunch things up a bit — especially with something like the Giles Audio preamp that only has an input gain control; the compressor gives you much more output gain control, so they’re handy together. Greg Walker: I see a Universal Audio LA610 channel strip there. What is that being used for? Alex Bennett: That’s also Sam’s. I’m using it for the trumpets; there’s a ribbon mic out there for the trumpets, and the LA610 let’s me give it just a bit of high shelf EQ brightness that works well on brass. Sam Teskey: That’s something I’ve always had. It often ends up on Josh’s vocal just because it’s so easy to dial up. Our first album was all done on that. The Giles Audio preamp has a bit more fidelity and a big valve sound, but the Universal Audio LA610 is more of a character thing. Greg Walker: What else do you have in this rig? Alex Bennett: I use a lot of Ekadek gear, made by Greg Brice; he’s a friend of mine who lives in New Zealand. He customised my mixing console, and he’s made a lot of boxes for my studio. We’re testing out a prototype preamp of his on this gig; it’s kind of a Neve-ish vibe with a 1080-style EQ section. [Points to items in rack.] He also built this VariMu compressor and this two-channel valve preamp with an EQ stage — it’s got tons of gain and switchable input impedances so you can choose the right impedance for the right ribbon mic. Also, it has no phantom power so you can’t blow up your old ribbons! Greg Walker: And the tape machine is yours? Alex Bennett: Yeah. It’s an MCI JH 114 oneinch eight-track analogue tape recorder. We had a balanced/unbalanced issue while connecting it to
the Chilton console but I got to the bottom of that. The console has transformers on the microphone inputs but not on the outputs, which is no big deal – the preamps are nice and that’s all that matters to me. The bussing section is good, too. It’s really the perfect console format for this job. Greg Walker: I assume you’re using a fair bit of tape... Alex Bennett: Over the four shows we’ll end up going through 12 reels of tape. It’s RTM 911 formulation, which is the modern equivalent of the old 456 tape. I’m running it at 15ips, with IEC equalisation at 410nW/m, and I’m not using any noise reduction. Greg Walker: 12 reels? That’s three reels per show. With all the submixing, patching and reel changes required, I guess you’re busy the whole time! Alex Bennett: Yeah. Although the submixing is pretty intuitive. The drum bus doesn’t change too much but every now and then I’ll need more or less kick, give it some more bottom end or whatever. The drums are being captured with just four mics and they’re going down to one track on tape, so I’ll ride the subgroup fader depending on the dynamics of Liam’s playing; feed it a bit more in the quieter sections, or give it more headroom if I know he’s about to go wild. I’m trying not to crush
the busses too much, I want to keep it snappy in the dynamics. Do you want to hear something? [At this point Alex tees up a reel with a freshly tracked version of ‘So Caught Up’ from the night before. The track that greets our ears through the small Genelec monitors sounds great; it’s got a deep throaty analogue character to it, and Alex has captured the quality and energy of the sounds beautifully.] Sam Teskey: If the performances are there I reckon we’ve got it! MICROPHONES
With 34 lines coming into the digital recording rig along with the additional mics in place for the analogue recording rig, there’s obviously a lot of mics on stage. The following focuses on some of the more interesting choices, and particularly those being used for the analogue recording... Greg Walker: On stage I saw some ribbon mics that looked like they were for the brass section... Sam Teskey: Yeah. They’re only for the recording; one of them looks like a cheese grater! Alex Bennett: There’s an RCA 77DX and a Steanes Sound Systems R47 – that’s the one Sam called a ‘cheese grater’! (laughs) It’s actually an old Australian ribbon mic, manufactured in Melbourne
MCI JH 114 one-inch eight-track analogue tape recorder. Chilton QM3 12-4-2 recording/mixing console. Ekadek FerralGrapha two channel preamp. Ekadek and Giles Audio preamps and Preamps, processors & mixers for processors. analogue recording rig. Alex Bennett, Sam Tesky and Nao Anzai in the makeshift analogue control room. Note absorption and tape boxes on rear wall! Analogue recording mics: AKG D12 (kick), Sennheiser 441 (snare), RFT Gefell MV101s (toms/ cymbals). AKG D12 for analogue recording of bass amp. Shure SM57 and DI for FOH.
in the ‘40s or ‘50s. The two trumpets go into the RCA, and the trombone goes into the Steanes. Those ribbons are also acting like room mics for me because they’re quite far away from the horns and their rear lobes face the audience. Nao Anzai: As an indication of distance, for FOH I’m using a Shure Beta 57 for the trumpets and a Peavy PVM520i for the trombone, and Alex’s mics are about a metre further back than mine. Alex Bennett: They’re working out great because I need to keep it simple and I don’t have room for the dedicated room mics that Matt is using in the digital recording system. Greg Walker: What are the room mics you’re using, Matt? AT 17
Matt Duffy: They’re Electro-Voice 635A omnidirectional dynamics. Classic ‘70s mics... Alex Bennett: They’re the same era as the Electro-Voice RE16s I’ve got strapped on to the vocal mics for the analogue recording. Sam Teskey: There’s a lot of Electro-Voice mics on stage. For tonight we’ve got two guest backing vocalists, and each of them has a Shure SM58 for the live mix with an Electro-Voice mic taped on to it for the analogue tape recording. Electro-Voice’s RE16 and DS35 are very similar mics and we’re using them as recording mics for the vocals across the front – Alex tried a few different options and liked them the best. Backing vocals for the drums and bass are AKG D190Cs, another old classic. Greg Walker: Speaking of classic AKGs, I saw some D12s on stage... Sam Teskey: Yes. We’re using a D12 on the bass amp, and another on the kick. Alex Bennett: The D12 on the kick is for the analogue recording, along with a Sennheiser 441 on snare – l love that mic! I’ve also got two modified RFT Gefell MV101s set up in an oldschool placement where they’re capturing a lot of rack tom and floor tom while also doing the overhead job. Sam Teskey: When you’ve only got four mics to record a kit it’s important to get that balance right. Nao Anzai: It also helps to have a super balanced drummer like Liam! On most of the quiet songs I’m just using my kick drum mic (Shure Beta 52) and my overheads (Shure KSM32s) through FOH and it’s sounding great. But I can only do that because Liam is a super balanced drummer. I don’t need to worry about cymbal spill or whatever, everything is completely fine because of his playing. Greg Walker: What about the guitar amps? I saw some vintage Sennheiser 421s there – the old white ones – and some of those Shure ribbons, the black and red ones... Sam Teskey: I love those old white 421s on my guitar amp, they sound completely different to the black ones. They’re great for guitar amps. Nao Anzai: Agreed! I’ve got one of those early white 421s; they’re great mics. They were completely discrete and hand-made. Later, after the white ones got a good reputation, they moved to a bigger factory and mass produced the black ones. The white ones were almost more like a prototype, made in small runs by hand. Sam Teskey: My favourite for guitar amps in the studio is a Coles 4038 ribbon off the cab a bit, but they’re a really sensitive mic and you can’t use them up close to an amp in a live setting because they break up. That’s why we’re using the modern Shure KSM313 ribbon mics – the black and red ones. Nao Anzai: The best thing about those Shure ribbons is that they’re built tough for live performance. It’s really hard to break one. I didn’t want to use ribbons for these gigs but they said “no, these ones are fine.” I tested them and they’ve been great. I’m also miking the guitar amps with Shure SM57s. Greg Walker: What about the keyboards? Sam Teskey: We’ve got an old organ using a Leslie speaker, but there’s also a modern keyboard, AT 18
a Nord; we put that through a Fender amp to match the tones a little. Alex Bennett: We couldn’t just DI the Nord because both it and the organ [miked up through the Leslie] are going to the same tape track, and they had completely different textures. Running the Nord through the amp brings it a little closer to the texture of the Leslie. For recording, I miked that amp with a Sennheiser 441, and the Leslie with an ElectroVoice RE20. The RE20 also acts as a bit of a room mic because it faces the audience. Sam Teskey: Because we’ve got mono drums for the analogue recording hopefully these more roomy mics – from the brass on one side of the stage and the keyboard amps on the other side – will open up the stereo image a bit more, so it’ll be punchy in the middle and more spacious on the sides. GOING ANALOGUE...
Sam Teskey, Alex Bennett and Nao Anzai talk about tape recorders, tape formulations, mixing and mastering... Greg Walker: How did this passion for using analogue tape begin, Sam? Sam Teskey: Around 13 years ago, when my setup was pretty much just a home studio, I borrowed an eight-track Tascam tape machine and Nao showed us how to sync it to Cubase and Nuendo. I was learning heaps by watching Nao work with the tape machine, and after a while he happened to be selling one of his eight-tracks. I bought it and used it for years. Later on I bought the Studer 24-track we use now, which is quite a hi-fi machine, but it brought with it a whole new level of learning, maintenance and calibration that I needed to understand, whereas the Tascam was just really solid and simple – I think I only had it serviced once a year. Nao Anzai: I used that Tascam machine for mastering; for bouncing mixes from the computer to tape. I would set it up so that the first two tracks were hitting at a moderate level with noise reduction, the second two tracks at super hot levels with noise reduction, the third two tracks at moderate levels with no noise reduction, and the last two tracks at super hot levels with no noise reduction. Then I’d bounce it all back into the computer and rebalance the song part by part – in the quiet parts I’d use the noise reduction tracks more, then blend the other tracks up in the mix for the louder sections and get more tape compression as well. Before I had a decent master compressor, I used the tape machine like that. I stopped using analogue tape for mastering about five years ago when I couldn’t get any more of the original 499 and 456 tapes... Alex Bennett: You don’t like the new tapes they’re making now? Nao Anzai: I think they’re fine for tracking, but mastering is a different thing; also a lot of my clients now don’t like the hiss. The other thing that changed for me was Universal Audio’s Ampex ATR102 tape machine plug-in. With that you can decide, on the fly, which tape formulation is better for which song, which tape speed is better, whether or not you want a transformer output, and so on.
Song by song you can just switch these things on or off. This completely blew my mind, because when you’re using a real analogue tape machine these are the things you have to decide before you start the mastering session. Calibration took a long time so you couldn’t make changes on the fly, but with the plug-in now you can and this has given me a huge advantage. Sam Teskey: Plug-ins are good, but I love Alex’s approach to recording and mixing too. I really admire the fact that he doesn’t use computers at all. When we started working together I said to him, ‘Do some mixes, we’ll listen to the digital bounces and give you some feedback’, but it’s much harder to work in that digital ‘mix, save, revise’ manner with a fully analogue system. So Alex would say, ‘Just come to the studio for three days and we’ll mix it all together’... Alex Bennett: Then it’s done! Sam Teskey: Yeah, then it’s done! Greg Walker: Speaking of mixing... Sam, when this all wraps up will you be mixing the analogue recordings yourself? Sam Teskey: No I’m going to keep it in Alex’s hands because he’s the mastermind of this whole analogue set-up. He’ll take it back to his studio in Castlemaine where there’s no computer, and he’ll mix it direct down to 1/4-inch tape. I’ll probably go there for a few days and work on the mixes with him, acting more as a musician. We’ll mix as much of the good stuff as we can and then take it to Crystal Mastering in Melbourne to transfer it to digital format so we can share the mixes between ourselves, listen to them and decide which ones we want for the live album. We can only fit 45 minutes on a vinyl release, but if the recordings go really well there’s room to maybe do a deluxe edition or a double LP. Or maybe we’ll just release the full version on Spotify... But all of that is for later; for now I’m really focusing on the shows and keeping my head out of this other stuff as much as I can... BACK TO WORK
The discussion comes to a timely halt when soundcheck is called. I spend a bit more time at front of house with Nao, where he points out a portable Zoom recorder that he’s rigged up to catch extra crowd ambience for his Reaper recordings – another example of the happy blend of technologies being roped in to achieve the desired result. Later that night the Teskey Brothers play a soulful show to another sold-out audience. The live sound is sweet and full and has a relaxed quality that is pretty rare at a live gig these days. It’s not super loud or sub-heavy, inviting the audience in rather than beating them into submission. Nao is perched at the FOH mixing position and, true to his word, the drums do sound great with just the overhead mics and a bit of kick. It’s yet another example of how the band and crew back their audio philosophies with carefully considered actions that really do buck the trend of contemporary studio and live production. As the awards and accolades continue to accumulate there’s no doubting it’s an approach that’s paid off big time.
The F6. A new era in recording 32-Bit Float Recording: Capture The Calm & The Storm In the field, fluctuating dynamics can prove challenging. Loud sounds can cause clipping and quiet sounds can get lost in the noise. The F6 uses 32-bit float recording with dual A/D converters to capture both explosive and subtle sounds at full audio quality – without ever adjusting gain!
Mic Preamps: 6 pro-grade pres feature a super-low noise floor (-127dBu EIN) and up to 75dB of gain.
Precision Time Code: On or off, the F6’s Temperature Compensated Crystal Oscillator (TCXO) generates time code at 0.2ppm accuracy.
Advanced Look-Ahead Hybrid Limiters: By adding a 1ms delay, the limiters ‘look ahead’ anticipating clipping before it’s recorded.
Zoom AutoMix: lets you keep your eyes on the action by automatically adjusting the levels of your mix to reduce ambient sound.
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TUTORIAL
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Have you ever thought about working in audio for games or virtual reality? Did you know that movie trailers are not made by the same people who make the movies? Sound Recordist and Designer Stephan Schütze provides some background to how things are done for these fascinating aspects of the audio industry. Tutorial: Stephan Schütze
The process of laying out tracks and building up sonic worlds allows for lots of creativity, but the underlying consideration is communicating with the audience about things seen and unseen.
In over 20 years of working as a location recordist and sound designer, I have recorded and produced audio for interactive media, sound libraries, movie trailers, film, virtual reality and augmented reality. Yes, that is as fun as it sounds! In doing so, I’ve discovered that each of these formats has different needs and requires a different approach to creating the content. In this article, I’m going to share what I’ve learnt. CAPTURE
The most important thing when recording is to capture a great sound; we have to hunt down those rare and interesting sound opportunities like a photographer chasing the perfect sunset. Location recording is much more than just pointing a microphone at something noisy – it is about discovering a world so rich with incredible sound sources that we will never run out of things to record. The source sound is everything, and whatever can capture that sound becomes the best tool at the time. I’ve had people laugh when they learn that I sometimes use a Zoom H1, and I chuckle back knowing that my H1 recordings are in the official Captain Marvel movie trailer. Remember, Academy Award-winning Hollywood sound designer Ben Burtt (Star Wars, Indiana Jones, etc.) did some of his best work with analogue tape and razor blades... The core consideration is to start with clean usable sounds, and I’ve always strived to achieve this while recording. Our recording skills are critical to what we do, but the software that supports us frees our creativity. Advances in
technology, such as iZotope’s RX, mean that I can record kookaburras on my porch and not worry too much about a bit of wind noise in the trees because I can remove it later. Once we have a selection of amazing raw material for a particular project, we need to prepare it for the platform we’re working on. Each platform has its own requirements, as I’ll describe below. GAMES
Games are interactive, making them quite different to other forms of media. Interactive formats are constructed to behave much like the real world does, so there can never be a single ‘final’ mix for a game; the mix is created automatically by the game’s ‘sound engine’, in real time, in response to the player’s actions. That means you can’t just drop in a convenient stereo ambience of a forest, for example. The player is likely to be walking through that forest and can choose their own path, and that will determine the sounds that are heard and how they are mixed together. In the forest example, we need to have individual bird sounds placed in the trees that the player moves between. The stereo or surround impression the player hears is from multiple mono sounds positioned in a 3D world. So when the player walks between two trees the bird sounds will be heard on either side, and these sounds will grow louder as the player approaches them and softer as the player moves away from them. All of that real-time manipulation is done by the game’s sound engine. The audio team can establish guidelines for how loud each individual sound or group of sounds is, how they blend
together and so on, but it is the actions of the player and their specific position at any one time that defines the mix they will hear. This makes balancing game audio hugely challenging because you cannot blend individual sound elements to cover gaps or weak transitions. Most of the individual sound elements are exposed to the player, and the player’s journey through the game world determines how they are blended together. Obviously, all sounds used within a game need to be clean and very isolated. Going back to my
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forest example, I’ll need separate bird sounds, a separate wind sound, a separate stream sound, and separate versions of any other sound elements; they all need to be highly isolated, and placed individually. When I prepare bird sounds to use in a game, for example, I need to cut them up into individual calls and place them into the desired place within the game’s 3D world. A common tool for this purpose is Wwise, produced by Canadian company Audiokinetic. Wwise allows me to take a selection of individual bird sounds and define their behaviour so that I can create a volumetric area in which birds will twitter randomly in real time. I essentially simulate real-world bird behaviour inside the game world. VIRTUAL & AUGMENTED REALITY
The New Reality formats are hugely different to every other platform; even games, which they share some similarities with. The key reason for this is the desire to create a realistic spherical sound field for the audience. Remember, we are placing our audience inside the virtual world; unlike a game, they are no longer just looking into that world through a screen, they are immersed within it. The various technologies we use to achieve this attempt to simulate how we localise sound in the real world. They simulate the interaural amplitude differences, the interaural time and phase differences, and the filtering of sounds reaching us from different directions, all to create a realistic immersive sound experience. The technology is still evolving, but that’s what it aims to do. There are many companies trying to create the perfect technology solution for this functionality. Companies such as Two Big Ears were doing an excellent job of this, and that is why they were bought by Facebook – their technology now forms the basis of the Oculus’ audio system. At this point in time we do not fully understand the science of how humans triangulate sound sources, so it is no wonder that the different technology solutions vary in effectiveness. The important thing about all of the above is that the amplitude and frequency makeup of a sound are critical aspects of how our brains calculate where a sound is coming from. This means we cannot just boost the midrange to give a sound more presence or add a sub channel for added impact, because any enhancement like this could interfere with the directional nature of spatial positioning. Essentially, if we want a sound to be located 45 degrees to the audience’s right and 15 degrees above the horizon, any alterations we make to that sound may influence the perception of where it originates from. Elements that might add excitement to a project, such as enhanced low frequency content, need to be carefully designed to highlight the audio without working against the illusion of realism. As with game audio, I start with clean and well-isolated mono files and place them into the simulated ‘world’, allowing the technology to handle distance attenuation and occlusion filtering. Then I layer in combinations of stereo, 3D, ambisonic and even binaural sound material AT 22
to achieve the end result. For example, VR Regatta, the VR sailing game for Oculus Rift, uses an array of over a dozen layers that play simultaneously in real time – just to create the spherical wind sounds! The layers are all interacting in real time within the spherical sound field; in many ways, it is more like a live performance than a pre-produced product. Much of what makes this content sound great is achieved during implementation. Virtual Reality (VR) and Augmented Reality (AR) are both going to be incredible formats in the future, and I honestly think no one has gotten close to realising their potential yet. Audio production for these formats is difficult and challenging, but it is also incredibly satisfying as we are literally creating a whole new form of media and discovering what can be effective. Good quality isolated audio is critical to the immersive effectiveness of these formats. If you plan on working in the VR, AR or 360 video space, I highly recommend you check out some of the discussion groups on Facebook that talk about the processes. The technology is changing so quickly that it can be a huge benefit to keep in touch with others working in this field, and
seek their advice when necessary. We are all fairly new to these formats, and sharing the knowledge helps everyone. TRAILERS
The approach to creating great audio for movie trailers is very different to games, VR, AR, and, surprisingly, film. It is odd and a little counterintuitive to discover that the trailer for a film will have a very different sound production process than the film it is promoting. For a start, it’s a completely different team that works on the trailers. Movie trailers, especially in Hollywood, are big business – there are studios dedicated entirely to producing high-impact trailers! And that is the core of it: it’s a specialised process because film trailers have 30 to 60 seconds to grab the audience’s attention and get them excited and engaged. While the visuals and dialogue for a trailer are cuts from the film, the sound and music are created specifically for the trailer. Sound Effects (SFX) for trailers are super hyped. If the dial goes up to 10, then trailer SFX sit somewhere around 12 or 13! There is certainly an
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element of the loudness wars here, but it is more nuanced than just turning up the volume and compressing the hell out of everything. The sounds themselves need to simultaneously achieve two things: they must be punchy and really cut through, but they also need to stay well out of the way of the trailer’s voiceover and music. It takes many, many layers of sounds to achieve the end result, and the editors do an incredible job of blending all this together for the audience. As someone who has been recording and using my own raw material for 20 years, I know my content really well. Despite this, there are a handful of my sounds in the trailer for the latest Fast & Furious movie (‘Hobbs & Shaw’) that I cannot even recognise because of how densely the sound has been layered and mixed to get that high impact end result. When creating trailer SFX there is a significant emphasis on mid-range content, which is often boosted to what would normally be considered silly levels. The trailer music usually occupies much of the frequency range we want to use; it dominates the very low and very high end of the spectrum, forcing us to tailor the SFX to have maximum impact inside of those two extremes. When preparing the sounds, I have a template where I layer my own sounds and do a lot of work to boost that mid-range frequency content. I find this tricky because the easiest way to make a sound cut through a mix is to add or boost high-frequency content so the crispness of the sound carries it through. Without being able to rely on the lovely high frequencies, I need to paint with a broad brush across the middle of the frequency spectrum. Compression is essential, but I do a lot of that work manually. If I am layering eight to 10 sound files I will build a custom volume curve for each sound, and tune each one so that I hear the exact elements I want to hear at the exact time I want to hear them. I mentioned earlier that game sound consists of many different layers that are all automatically mixed together by the game engine to provide a real-time experience. Film trailers are the polar opposite: the layers are not just hammered together, they are surgically grafted with each other to allow each element to do its best work and then move out of the way for the next element. AT 24
The challenges are very different for trailers, but the intensity of the audio content you are working with really creates excitement. Working on trailers is fun but very challenging, and, of course, it’s always cool to hear your content on the big screen! Just be careful of your ears because that high intensity can be really fatiguing. FILM
I am not going to include television as I have not worked for TV beyond commercials, and my film experience is not as significant as my game experience. Nonetheless, there are some fundamental differences between working with sound for linear media such as film, and for nonlinear media such as games, and these differences influence my approach to each platform. The main thing for me with film is to achieve and maintain sonic consistency over time. One of the key aspects of film production I have encountered is the room tone/atmos and ambience setup. It’s a little like the audio equivalent of colour grading, where you have to take all of these individual shots, from different angles and with different lighting, and try to create a consistent lighting look across all the edits. It’s the same with the room tones and atmos. You need to remove all the background content from the dialogue tracks, as it is often really different and distracting to the audience, and replace it with a nice consistent tone across each scene. It’s like applying a smooth undercoat before you start painting. With film, I tend to build up sounds from the rear to the front. I start with long, smooth, clean edits and transitions, and use spot sound effects to provide support for actions and events. Again, all the content I am editing and preparing needs to be super clean because, unlike nonlinear media that is mixed in real time, film has no limits to the number of sounds you can combine simultaneously. Layers and layers can be combined, but every single layer has the potential to contribute to the overall level of unwanted noise. Combining many layers risks becoming an utter mess of unwanted noise if the sounds are not super clean to begin with. Unlike non-linear media, you have much more
control in film because you can set a scene exactly how you want it to sound and it will play back the same every time. After years of working in games, this is very different; you can push things to their limits and know they will never exceed those limits. This is why films can be dynamically more intense than games. The interactive and immersive nature of games is where they get their impact, not from a super tight mix. SEEN & UNSEEN
The process of laying out tracks and building up sonic worlds allows for lots of creativity, but the underlying consideration is communicating with the audience about things seen and unseen. Audio is our principal emotional sense, and the soundtrack for any media needs to enhance and support the narrative of the dialogue and music while also transporting the audience into the worlds we create. Each delivery format has its own challenges, strengths and weaknesses. Understanding the best way to approach each format allows us to enjoy the process of creating an audio world, rather than fighting with the content to make it fit. Never be afraid to try new things or crazy ideas. So much of the best creative work comes from folk who think and act outside the box. We all work in sound because we could not imagine ourselves doing anything else! As someone who records raw material for sound libraries and also uses that material in sound production, I’m experienced with the entire workflow – from capture of raw sounds to delivery of finished product. This provides me with a useful perspective, and I honestly think it makes me better at both jobs. When recording raw sounds on location, I consider all the ways I might want to manipulate those sounds as part of production. I will often record sounds that are very ordinary in their raw format, but I know are going to be a great basis for sound design. Likewise, when doing sound design I am aware of the limitations of recording on location and just how hard it can be to capture really clean content. I am going to explore these concepts in more detail in future issues of AudioTechnology. Stay tuned!
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Mark Woods goes backstage to see how Sennheiser came to the rescue for a show that involves dialogue, singing, dancing, gymnastics and a live band. Story: Mark Woods
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Theatrical productions are tough audio. Mixing bands has its own challenges, but at least the vocalists sing into the mics. I had cause to reflect on this while visiting Sydney’s State Theatre to see the non-stop whirl of songs, dance and gymnastics known as Bring It On: The Musical. Based on the loyalties and rivalries of competing cheerleaders, the show is focused on the performers and the choreography. The audio production includes 22 wireless headset channels and a live band. Sound designer Greg Ginger from Outlook Communications gave me a look around. HIDING THE BAND
The show needs all the floor space it can get so the stage at the State Theatre was extended forward 4m, covering the orchestra pit. The six-piece band was set up downstairs and backstage in the Green Room, leaving it looking like a combination of a Green Room, a rehearsal studio and a live recording room. It took a bit of soundchecking to get levels comfortable in the room, with everyone so up close and personal. The guitar amp ended up in a road case, and I think Greg Ginger wanted to put the drum kit in one too but couldn’t find a case big enough, so he dampened the skins with some cotton wool as a compromise. The benefits of the band being hidden away and removed from the stage were studio-like separation from the stage sound, and complete control in the FOH. The band had a conductor, and followed the show via headphone monitoring and a screen. The mix required 24 channels to cover the band, with lots of Sennheiser e614 small-diaphragm condensers to be seen. The 24 channels were sent to a Digico SD5 FOH console, mixed in with the on-stage performers’ headset mics and some manually-triggered FX from QLab, then processed through a BSS Soundweb BLU-100 and fed to the house speaker system. The performers on stage heard themselves and the band through four little Funktion-One F55 dual five-inch speakers [[08]] attached to the front lighting truss, with another four on the middle truss. Positioned about 6m above the stage — just out of reach of flying cheerleaders — they weren’t close or loud, but supplied an even spread of sound that was just enough for the performers to perform happily.
With most of the cast in cheerleader outfits, hiding the wireless bodypack transmitters and keeping them in place as the performers jumped around could be an even bigger problem.
Green room converted into performance space.
Green Room (& below) converted into performance space.
ON-STAGE PERFORMERS
The person who invents invisible wireless microphones is going to do very well, but until then it’s omni-directional headset mics and bodypack transmitters for theatrical performers. The average rock opera might get away with headset mics somewhere down the sides of the performers’ faces; it’s not an ideal place to pick up vocals in itself, but it’s a luxury compared to Bring It On where the mics are hidden away on the performers’ hairlines or in wigs. The mics aren’t the only things that need to be hidden. With most of the cast in cheerleader outfits, hiding the wireless bodypack transmitters and keeping them in place as the performers jumped around could be an even bigger problem. Regular bodypacks (typically powered by a couple AT 27
of AA batteries) would have been awkward under the tight costumes, and, with 22 wireless channels and a couple of spares, would have required a carton of fresh AAs for every show. Sennheiser’s Digital 6000 system came to the rescue. Their new SK6212 bodypack transmitter, released earlier this year, is so small that it’ll change expectations for these devices. It feels strong, the replaceable antenna is thin and flexible, and I like the look and feel of the rounded corners and sides — it shouldn’t break or dig in if a performer falls over and lands on it. An additional inner seal helps to repel moisture, and Greg noted its effectiveness in this sweatier than average application. Bring It On: The Musical is a fairly simple but boisterous and dynamic show. The audio production had to work within the constraints
imposed by the physicality of the show, and relied heavily on the quality, stability and manageability of the large number of wireless channels. The singers on stage were the stars of the show and, apart from my concerns about mics in wigs and the lack of opportunity for any sort of mic technique, it worked remarkably well at showtime. The cast were experienced musical theatre performers with strong voices and good projection. The 2000-person capacity State Theatre has excellent acoustics and the PA was wellcontrolled in the highs and high-mids where it might feed back, so despite the inherently risky combination of a loud show, 22 omnis and a PA, there was no feedback or instability. The band sounded great, they were mixed to suit the vocals and I didn’t miss seeing them.
Funktion-One F55s mounted on the overhead trusses provided on-stage monitoring while remaining clear of flying cheerleaders
Sennheiser e614 small-diaphragm condenser microphones were used on nearly everything for the band. They had no problem handling the peaky cymbals/percussion, and their tight supercardioid pickup pattern helped with separation in the crowded green room.
Sennheiser’s SK6212 digital bodypack transmitter With 22 wireless mics on stage, plus spares, it pays to be organised.
Sennheiser’s EM6000 two-channel receiver and LM6000 charging module.
SENNHEISER’S 6000 SERIES The 6000 series as used in Bring It On consisted of three components: the SK6212 digital bodypack transmitters, the EM6000 digital receivers, and the LM6000 charging module. Matchbox-sized at 63 x 47 x 20mm, the SK6212 bodypack is about half the size we’re used to. Weight is also about half, at 112g total with the BA62 lithium-polymer rechargeable battery on board. LiPo batteries are expensive but light, and the BA62s allow an astonishing 12 hours of operation between charges. They can be removed and recharged in the Digital 6000 rackmount charger via the dedicated LM6062 charging module. The long battery life made everything easier for Bring It On, especially on days with two shows. Greg showed me one unit that had been on for four hours and still indicated a full battery. LiPo batteries do wear out eventually, but Sennheiser says users can expect well over 500 duty cycles before replacement. The OLED screen on the front of the transmitter is small but clear. Above the screen sits a single sunken Power/Mute button that requires determination and fingernails to activate; a short press gets Mute, while pressing for a few seconds turns the unit off. It turns back on almost instantly. Below the screen are three slightly recessed butAT 28
tons to access the menu options. The screen turns off automatically after a few seconds of inactivity, but pressing any key brings it back with a display of the operating frequency (or a user name), the battery level and the time of day; a nice touch. Keep pressing and you get an audio level meter. The operating frequency can be changed from within the bodypack, and the power/mute LED light can be turned off if it’s distracting. The EM6000 is a two-channel 1U digital receiver that presents itself as a premium product with solid construction and a quality feel to the controls. Operating over a wide 470MHz to 714MHz switching range, the Digital 6000 system is free from intermodulation between channels so it can use a simple equidistant frequency grid with a minimum 400kHz spacing for maximum channel density and stability. The sharp OLED screen’s default display for each channel contains everything you need at a glance, including a Link Quality Indicator (LQI) meter, with more details and editing available by turning the control knob to access the menus. As soon as you stray from the default display the Esc button lights up, offering a quick way back. Around the back are the expected XLR and 6.35mm analogue audio outputs, joined by digital
AES-3 and RJ-45 Dante outputs (optional card). External sync is possible via I/O BNC connectors and the RF antennas have BNC output splits for daisy-chaining up to eight EM6000s. Greg made good use of these options by sending the Dante output to FOH, with the analogue output used for backup, and the headphone output used for switching monitoring. The Digital 6000 series uses the same Sennheiser Digital Audio Codec (SeDAC) found in the top-spec Digital 9000 Series, and Greg found the audio quality to be outstanding (particularly the D/A conversion) with a wide frequency response and dynamic range. Latency is quoted at 3ms for all output formats and was not noticeable in use, nor was any of the error correction or audio error masking that could be happening in the background. The sound design for Bring It On required a wireless microphone system that was sonically accurate, flat-battery-proof and discrete. Just the brief for the Sennheiser Digital 6000. Priced for professional applications, it’s been embraced by production and hire companies with good reason. Sennheiser is the long-standing market leader in the wireless segment and the Digital 6000 Series is as good as you can get.
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FEATURE
SMALL ISLAND BIG SONG HUGE STORY
Tim Cole and BaoBao Chen set off on a recording adventure that would take them from Madagascar to Easter Island. Since then they’ve released a multi award-winning album, uploaded a series of music videos, and made a movie that’s screening at festivals around the world. Story: Greg Simmons
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So we hung the mics from trees, or found sticks to tape them to
While recording traditional music in Vanuatu, producer/engineer Tim Cole learnt of a heritage shared by island communities throughout Asia, the Pacific Ocean and the Indian Ocean. Most of those communities were established by ancient seafarers; the indigenous people of Taiwan who, thousands of years ago, pushed their boats out to sea and headed for the horizon. Inspired by this, Tim thought “What an incredible wealth of cultures, music and instruments to base an album on.” A year later he was in Australia’s remote Central Desert recording Inmas (songlines) of the Pintupi Elders. Their ancient songs were full of information about living on the land, maintaining social systems in harmony with nature, and more. That night, while listening to the IPCC report on climate change, a tragic irony struck him: the people who know how to live sustainably on the land will be the first to lose their land through the unsustainable practices of the rest of the world. The developed world is going to lose the knowledge it needs to hear, at the time when it needs to hear it. This was all the motivation Tim needed to act on his previous inspiration. Teaming up with BaoBao Chen, they headed to the shores of Taiwan to retrace the paths of the ancient seafarers. They visited 16 different island nations and collaborated with over 100 artists to create a series of contemporary ‘Oceanic Songlines’ for a sustainable future. The resulting album won the ‘Best Asia & Pacific Album’ award at Songlines Music Awards (UK), the ‘German Record Critic’s Award’ (Preis der Deutschen Schallplattenkritik), and was nominated for ‘Best Concept Album’ at the American Independent Music Awards. They’re currently taking the movie of their musical adventure around the world festival circuit.
In the following interview, Tim relates their story of tropical islands, typhoons, bush mechanics and audio derring-do... EQUIPMENT
Greg Simmons: Thanks for stopping by, Tim. AudioTechnology is a magazine for sound engineers and recording musicians, so we’d better start with the recording equipment you used... Tim Cole: Sure. I started with a few mics I’d collected over the years. I had a Neumann TLM102 studio condenser mic and a Shure SM87 live condenser mic, and with those two I knew I could pull a good vocal sound. I also had a little AKG 408 condenser mic that was handy because I could clip it on to things. I didn’t have a stereo pair, and I didn’t have a field recorder. I’d been working at CAAMA [Central Australia Aboriginal Media Association] before starting this project, and as a farewell gift they gave me a Zoom H6; it provided me with an XY pair of cardioids, and it also became my field recorder. GS: You made this entire multi-tracked album using a Zoom H6 as your recorder? TC: Yes. All the recordings were made through the Zoom H6; sometimes on its own with a guide track, other times as a USB interface recording directly into Protools on my Macbook Pro. So in terms of mic preamps and converters, everything passed through the H6. GS: Any other gear? TC: We planned to film as well, so I bought a lapel mic that could connect directly to the camera; it’s called a ‘MyMic’, it has a little gooseneck and was incredibly handy. It turned out to be the main vocal mic for many of the performances – but only after a lot of EQ! Then I also got a Zoom H1; I wanted to record surround sound at the same time,
so I’d often put the H1 off in the distance to capture the left and right rear channels. About halfway through the project we got a sponsorship from Røde, who were really happy to help us. They gave us a matched pair of NT5s and the Stereo Bar, an NTG2 shotgun mic, and a Stereo Video Mic to put on the camera for ambience. And, of course, their windjammers for everything! GS: What about monitoring? TC: I wanted to do all the post-production in the field, as we went from island to island, and I knew the biggest problem I was going to have from an audio point of view was sitting the bottom end accurately. I ended up with the iLoud monitor from IK Multimedia, and a pair of Sennheiser HD8 DJ headphones. Between them I knew I could sit the bottom end close enough for the mixes and videos I was going to make in the field, and do the finetuning later in mastering. MAKING & BREAKING RULES
GS: This was an ambitious project that aimed to combine instruments that evolved in different parts of the world and with different musical aesthetics and tunings. Despite that, you’ve managed to blend them all together and create a consistent feel throughout the album. How did you achieve that? TC: When producing albums, I think it’s really great to set rules that define the approach you’re taking. Those rules help to provide consistency and they also help to define the album’s concept. I love concept albums, you know? Albums that are works of art, with a theme that bonds all the different parts together. The Small Island Big Song album is a concept album, of course, so I created some rules to hopefully tie all the different parts together while giving it the most substance and the most integrity; trying to keep it very honest and genuine. AT 31
GS: Are you referring to the producing/ engineering rules described on the Small Island Big Song website? TC: Yes. They were the rules for producing and engineering: only traditional instruments and languages could be recorded; all recordings had to be captured acoustically with no pickups or DIs; all recordings had to be driven by feel with no metronome or locking to a grid; and all takes had to be filmed. Also, all the artists involved had to be first nations people who were connected together through the seafaring heritage that is behind the whole project, and who are keeping their traditional language and instruments alive through their musical practice. GS: Did you ever have to break the rules? TC: Sometimes... There’s a song called Gasikara that we recorded with Airileke. He’s a traditional drummer who has studied throughout Papua New Guinea and the Pacific, but he’s also a hip-hop producer. He said, “I’m gonna give you beats”, which was against our rules, but he recorded himself playing traditional patterns from the village of Gabagaba — which means ‘Drum Drum’ in Motu, which is his language. His beats contained the garamut slit drums, the paté log drums, and the kundu and warup skin drums; they’re all traditional drums from that region and they were all played and sampled by him. We broke the rules there because, you know, these cultures haven’t somehow been frozen in time since Western colonisation; they’re alive now with a vision of their future, and they’re still pursuing new ideas and adapting new technologies — AT 32
The only tools I let myself use for mixing were EQ and compression.
including sampling, loops and beats. That’s why Airileke said he was going to give us beats: he was essentially making a statement about that... GS: I read on your website that you let the artists choose the location to record at. What if the chosen place was too difficult to get a good sound, or had lighting or other visual problems? TC: That was always a possibility but the goal of the project was to create something potent, something rich in the sonic and visual textures of those locations and the cultures that have the longest living relationships with them. So we let the artists guide the process; where we would record, what music they would perform and what instruments they would use — as long as they stayed within the rules. Then it would be up to us to make it work. GS: One of the videos shows a number of women standing in a river making music by splashing the water. It has a driving rhythmic pulse
that seems to come from the water itself, and there are vocals but I cannot see any microphones in the video. How did you make that work? TC: That was the Leweton Cultural Group from Vanuatu; it’s taken from an earlier project that helped to inspire Small Island Big Song — it’s called the Vanuatu Women’s Water Music. I was using different equipment then. I had a couple of Mann M1 microphones sitting in the bushes, and they were the key microphones picking up the water splashing. GS: How did you capture their voices? TC: Their voices weren’t loud enough in the recording from the river, so back at the village I played it through my monitors and asked them to sing along with it, close miked. That allowed me to add another layer of them singing, but cleaner and closer. GS: What about that driving rhythmic pulse? TC: I did some pre-recording as well. They often performed this same piece of music for tourists, but in a concrete swimming pool rather than the river. To record it – believe it or not – I dropped a Tandy PZM microphone straight in to the water. GS: Deliberately? TC: Yeah, I just dropped it straight in! I’d done it before in a swimming pool and in a bath, because I didn’t have a hydrophone. That Tandy PZM worked underwater with no protection, just like that — although it doesn’t work any more after that recording! I also used a Shure SM57 in a condom, all taped up and water sealed. From those mics I got that underwater sound; it’s a kind of explosive ‘boom’ that happens when the women slap a
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When producing albums, I think it’s really great to set rules that define the approach you’re taking
handful of air into the water. I post-synced that sound to the movement in the video to recreate the low frequency rhythm. MIC TREE
GS: We’re heading into some interesting recording territory here... Many of the videos show the microphone taped to a stick or a length of bamboo, or hanging from a tree. Was that just to look good for the video? TC: Not at all. That was actually a practical consideration. We could only bring what we could carry on the plane without incurring excess weight charges. Everything was weighed and budgeted because we were recording audio and capturing video at the same time, and there were only two of us. That meant no microphone stands, because they were too heavy. So we hung the mics from trees, or found sticks to tape them to. Sometimes we borrowed stands from the artists... GS: I asked you about that because anyone who sees the video with the woman standing on the beach singing into a Neumann that’s taped to a stick embedded in the sand, with waves washing up around it, would naturally think you’re either crazy or it was just for looks. TC: That’s the actual take you’re hearing in the song, breaking waves and all! GS: That ocean looks and sounds pretty wild... TC: It brings up one of the other reasons we were doing it this way; we want to talk about these AT 34
environments and what’s happening to them. The woman on the beach is Ado Kaliting Pacidal from the Pangcah/Amis people of the east coast of Taiwan. It was just after a typhoon so the ocean is very rough, and that’s why she chose that location at that time. Extreme weather events are becoming the norm on the east coast of Taiwan, where that part of the song was recorded and filmed. It’s a direct effect of climate change, and there she is standing on the beach after a typhoon, representing her culture, singing, and drawing that experience of extreme weather events into the song and the video. We’re not spelling it out directly, but it’s there to see and it’s part of the sub-text of the whole Small Island Big Song project. GS: Knowing that the ocean is rough because it’s just after a typhoon adds an extra layer of drama to that scene... TC: Another one was filming Koyawa’s vocals for the song Alie Sike. There were a few options there; we were looking at this beautiful beach with all the outriggers in the background, but Koyawa said, “My outrigger is moored in the mangroves, I’d like to record there...” Mangroves are one of the world’s most precious but maligned environments. They’re vitally important to our ecosystems, they store two to four times as much carbon as any other environment, but we’re pulling them down to reclaim land with ocean views. GS: What mic did you use for that one? TC: That was the Neumann TLM102 hanging
from a tree, it’s about 60cm to a metre in front of him. If you look carefully at the video you can see it. On that session I accidentally left all the windjammers behind, so I took off one of my socks and wrapped it over the mic. That’s the real take you’re seeing in the video and hearing in the mix. MIXING RULES
GS: That brings us to mixing. You made rules for recording to maintain a consistent feel throughout the album. Did you make similar rules for mixing? TC: Yes. The only tools I let myself use for mixing were EQ and compression... I chose not to use any spatial effects; there are no added delays, no reverbs, nothing like that, even though I love mixing with that stuff. GS: The album doesn’t sound like it was mixed with only EQ and compression. It sounds much more ‘produced’, with layers of atmospheres from all the different locations you recorded in. What apps and plugins did you use for mixing? TC: I mixed the album in Pro Tools on my MacBook Pro. For the compression and the broader EQ work I mostly used Slate Digital’s plugins: Earth, Air, the VCA compressor and the Neve EQ. I just love how broad the low mids are on that Neve plug-in for boosting the substance and body. For the precision corrective EQ I used Pro Tool’s EQ, just to knock out some frequencies that were getting in the way. I used iZotope’s RX for cleaning up the tracks and removing unwanted sounds.
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I love concept albums, you know? Albums that are works of art, with a theme that bonds all the different parts together
I’m using compression all over the place, but it’s usually just ratios of 2:1 and often I’d use really sharp attacks and long releases to pull things back about 2dB overall, just to pull the transients in. I got into a habit of doing that. I also made a choice not to take the levels too hot, and I kept the mixes all around -14 LUFS. For the whole album I tried to let it breathe as much as I could. GS: That brings us to mastering... TC: I did all of the mixing as we travelled from island to island, using the iLoud and the Sennheiser HD8s to get them as close as I could to sounding right. Then I said to BaoBao, “I have to spend a day in a studio somewhere to run these mixes through proper monitors and check that all of my mix information below 100Hz is true.” She said, “What’s the best studio in the world to do this?” We ended up at Peter Gabriel’s Real World! GS: An engineer’s dream, and a very appropriate choice! TC: Yes, it was around the time of my birthday, too, so it was a great present! We got to sit in the same room as some of the world’s greatest while listening to our mixes through really true monitors and passing them through some very nice outboard gear, like the Massenburg EQ, just for a bit of broad tweaking. We also got to do tape passes, pre and post, so I had both to play with. We did all of this with Oli Jacobs, the in-house engineer at Real World.
together on numerous songs. How did you deal with the contractual and legal aspects of that? TC: I’m glad you asked about that, because it’s a really important consideration. We spent a year developing that side of this project; looking into Fair Trade Music and the United Nations Declaration on the Rights of Indigenous Peoples, trying to find the best ethical structure for Small Island Big Song in terms of distributing any profits among the artists, figuring out the best way to handle copyright and content in relation to working with indigenous cultures and intangible cultural heritages, making collaborations between really well-known artists and very grassroots artists, and so on. It took a lot of time and research. We sought out a lawyer who specialises in these types of sensitive crosscultural projects, and we eventually came up with something in which every contributor and culture is treated fairly and respectfully. GS: I asked about that because people sometimes assume they can use recordings of indigenous music for nothing, and even profit from it with no obligations. I think it’s because no-one seems to ‘own’ this music in the Western sense of ‘ownership’, which overlooks the possibility that a culture itself could be an owner. I’m glad to hear that you gave it such thorough consideration. TC: Having all the legalities and contracts sorted out not only keeps everything fair and equitable, it also brings a lot of weight and substance to the whole process.
LEGALITIES
CLOSING
GS: What a way to finish the album! We’ve talked a lot about the rules you made for the musicians, and for recording and mixing. I’d like to ask you about rules of another kind: you were recording numerous artists from numerous countries, collaborating
GS: The effects of climate change on traditional cultures, especially those on islands, is a recurring theme throughout the Small Island Big Song material and also throughout this interview. It’s a good topic to finish with. Do you have any closing
MASTERING
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thoughts about that for our readers? TC: I’m not a factual documentary maker; I’m a music producer and I’m a sound engineer. So, as music producers and sound engineers, what can we do to bring these issues of climate change to people’s minds? We can record and produce meaningful music from people who have got something to say about it. That’s what we did, and we did it in the places where the effects of climate change are already visible. All the other stuff is great, too; I’m really excited about producing quality music, working with amazing artists playing incredible instruments with all these different tonal qualities, and creating incredible art... But at the same time I want to drop a boulder into the world’s psyche and let the ripples go out, to raise as much awareness as we can about the climate change problem, you know? GS: Absolutely. You said earlier that you wanted to create something potent and rich with this project, and I think you’ve achieved that. It’s sonically and visually compelling, and carries with it an important and timely message for the world. Thank you very much for taking the time to talk about it, Tim, and all the best at the film festivals. TC: And thank you for the chance to share the technical side and the legal structure; for me it’s just as important as the content.
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REVIEW
AUSTRIAN AUDIO OC818 Dual Diaphragm Condenser Microphone By combining the precise replication of a classic capsule with digital manipulation of polar responses, has this new Viennese company succeeded in bringing the essence of a classic mic into the 21st century?
NEED TO KNOW
Review: Greg Walker
PRICE OC818 Microphone: RRP $2099 OCR8 Bluetooth Dongle: RRP $299 CONTACT Group Technologies Australia (03) 9354 9133 sales@grouptechnologies.com.au PROS • Great studio all-rounder with excellent sonics • Dual diaphragm design • Comprehensive filter, pad and polar pattern options • Remote digital control adds continuously variable polar options • Innovative and powerful PolarDesigner plug-in delivers multiband polar shaping, EQ and spill minimisation • Mounting and windshield options supplied • Designed and built in Vienna, Austria CONS • USB Dongle for Polar Pilot app not supplied • No frequency response charts published • Rear mounted mini-plug a little fiddly SUMMARY Austrian Audio’s OC818 marries hand-built European pedigree with cutting edge digital technology to produce a top quality microphone that shines on almost any source. The mic’s smooth and classy sonics are augmented by multiband polar mix processing and remote set-up control to deliver a very desirable general-purpose recording solution.
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When AKG decided to take the path-mosttrodden and move production to Asia, a core group of engineers, designers and management decided to stay put in Austria and keep the dream of superior Teutonic microphone manufacturing alive. With several filing cabinets worth of new design ideas, an ambitious R&D program and plenty of organisational nous, Austrian Audio was born. The company’s first offering is the OC818, a microphone that blends classic capsule design with bleeding edge digital innovation to create something quite unique. MAKING HISTORY
When AKG moved away from their iconic 1950s valve-based microphone designs and released the C414 in 1971, they probably didn’t imagine that it would still be in production (albeit with a confusingly large amount of subsequent modifications) some 48 years later. The original C414, as well as early versions of its immediate successor the C414 EB, utilised the CK12 capsule. Along with Neumann’s M7 this is one of the undisputed ‘holy grail’ microphone capsules from the golden era of European microphone design. CK12 capsules were difficult to build due to their brass-mounted element and were time-consuming to service and maintain, but sounded like a million bucks. At some undefined point during the 1970s AKG quietly switched over to a more stable nylonmounted element that made manufacturing easier. The mic lost a little of its magic, but continued to be an extremely capable audio tool. FUTURE PAST
Drawing on this history, the engineers at Austrian Audio set out to recapture a little of that lost early C414 magic. Just to be clear, the resulting OC818 mic is a modern re-imagining of the original 414. It is not a slavish part-for-part clone (DIY-inclined owners of AKG 414 models will not be able to retrofit the Austrian Audio CKR12 capsules into their mics), and nor is it a compromised ‘built to a price point’ knock-off. Many months were spent reworking the new Austrian Audio CKR12 capsule until a satisfactory meeting of sonic excellence and manufacturing repeatability was obtained. Each hand-assembled, tested and fine-tuned OC818 features a ceramic mounting for an edgeterminated capsule that provides stability while delivering sonic results carefully measured against selected original CK12 examples. The design team also completely overhauled the original C414’s internals, adding some very 21st century features. The OC818 features a one-piece body cradling a dual diaphragm capsule allowing two-channel recording. Microprocessor controlled polar pattern responses can be adjusted remotely via an innovative Bluetooth-based app, while a dual-channel plug-in allows for multiband polar pattern shaping of dual-channel recordings at the mix stage. These last features firmly push the mic into new recording and post-production terrain. THE FULL PACKAGE
Unpacking the OC818 for the first time, I was surprised by its modest size and weight. While
PolarDesigner plug-in: a deep, powerful piece of software that can have a massive impact on the tone and character of your source material.
not really a small microphone, it is one of the more petite fully-featured large-diaphragm condensers on the market – especially given its dual diaphragm capsule. It’s an effortlessly stylish design with a tall head basket and a short onepiece base showing superb build quality in tasteful low-gloss silver with a subtle red ‘waist band’. The rear mini-plug socket for the second output is cleverly protected by a modest rubber plug until such time as it is required. The OC818 has a look that is very much its own, but gives a nod to its AKG roots with sliding switches for polar pattern, pad and bass filter selection at the front of the mic a-la the 414. Four standard polar patterns (figure eight, hypercardioid, cardioid and omni) are offered plus a fifth ‘black-dot’ preset option (factory default setting is broad cardioid). There are three bass roll-off filter settings; 12dB/octave at either 40Hz or 80Hz, and a 160Hz setting that drops from 6dB/octave below 160Hz to 12dB/octave below 80Hz. The filter points are well chosen and suit a wide variety of recording situations, while the -10dB and -20dB pad settings help the OC818 cope with loud SPL levels from sources such as drums and amplifiers. (A more affordable single diaphragm cardioid pattern version of this mic, the OC18, is also offered for those who want the sonic benefits of their CKR12 capsule without the extra polar pattern options.) The OC818 ships in a small but sturdy flight case with two different mounts. The first is a basic clip that holds the mic by the neck near the XLR input. The second is a robust suspension mount that provides good support and isolation while being quick and easy to set up and adjust. A short adaptor plugs into the mini-plug socket at the rear and terminates in a male XLR, while the supplied thick black windsock can be fitted snugly onto the OC818 for gusty recording assignments. Disappointingly, there was no frequency response graph supplied for the review model and none displayed on the company’s website, either. In every other aspect the microphone and accessories display superior quality of materials, design smarts and build execution and the whole package inspires confidence in the way that proper professional kit should.
On vocals the mic was great, delivering what I was increasingly coming to associate with it: superb tonal balance with trouble free low-end and silky highs that took EQ very well...
PROOF OF CONCEPT
Melbourne singer-songwriter Dan Flynn arrived at my studio shortly after the OC818 did, so we wasted no time putting the mic to work on some acoustic guitar beds. Dan plays a lovely Martin with a strong finger picking style, and we quickly discovered that the Austrian Audio mic and the Martin were a match made in heaven. With the bass roll-off filter set for a little rumble protection below 40Hz and the mic aimed at the 12th fret from a slightly elevated angle about one metre out in front, the sound coming back off tape made us both extremely happy. The articulation was smooth and even right across the spectrum with a lovely strong boom-free bottom end, great woody midrange and sweet natural-sounding highs. Subsequent tracking gave us excellent results in this application with a variety of guitars in both soft picking and manic strumming contexts. This turned out to be a sign of things to come, and over two days we used the OC818 as our go-to mic on a variety of overdubs. On vocals the mic was great, delivering what I was increasingly coming to associate with it: superb tonal balance with trouble-free low-end and silky highs that took EQ very well where required. I was also impressed with the overall sonic ‘image’ the mic presented, and the smooth way it handled transients and offaxis sources. With an output impedance of 275 ohms and a moderate sensitivity of 13mV/Pa, the OC818 allows preamps to run at reasonably high gain levels. Through various tube and solid-state preamps, it proved itself to be a reliable workhorse studio tool. Having established the OC818’s bona fides in basic mono tracking mode, it was time to investigate its dual channel possibilities and the intriguing digital accessories. POLAR EXPLORATION
Austrian Audio’s free Polar Pilot app is designed to expand the capabilities of the basic onboard mic settings. It lets users control the filters, pads and polar patterns of the OC818 remotely via a portable device such as a smartphone or tablet. To use the app you’ll need to purchase the OCR8 AT 39
dongle, then pair it via Bluetooth to your device of choice. The app itself is easy to load and navigate, with the key feature being that it allows fine-tuning between the fixed polar patterns accessible from the mic’s onboard switches. You can dial up a slightly cardioid omni for example, or a figure-of-eight skewed towards the front lobe. Such fine controls won’t be a selling point for everyone I suspect, but they certainly do expand the potential of what is already a very capable microphone. A favourite polar setting can be saved inside the mic as a preset, there’s an in-app 60-second clipping monitor/ logger, and the status of all the mic’s settings can be checked remotely from your device. POLE DANCING
While the Polar Pilot app is fairly straight-forward in terms of its operational scope, the same cannot be said of the PolarDesigner plug-in. This is a really deep, powerful piece of software that can have a massive impact on the tone and character of your source material. Its operation it is not unlike a multiband compressor, except with polar patterns instead of dynamics as its building blocks. As a post-production tool the free and open-source PolarDesigner provides an impressive range of creative sound shaping options as well as intelligent tools to tackle unwanted spill and proximity effect. The plug-in is instanced on a pair of tracks recorded from the front and back diaphragms of the OC818 (or other dual source mic, though the results may not be as optimised) and presented as a stereo track for the plug-in to work its magic. The GUI is clearly laid out and is dominated by the multiband polar pattern and frequency chart to the right. Sliders underneath each band’s graphic display provide continuously variable control of polar pattern and gain addition/reduction. Each band can also be soloed or muted here. On the left are controls for selecting the number of bands (up to five bands are available but you can select as few as one), equalisation and proximity effect as well as intelligent spill targeting and minimisation. Reverse polar patterns can be allowed or forbidden and presets can be saved and loaded. It’s hard not to feel like a kid in a lolly shop when you first pull up the PolarDesigner plug-in, such is the smorgasbord of possibilities on offer. Firstly, you can choose how many bands you want to address and adjust the crossover points to taste. Then, of course, there’s the small matter of deciding which polar pattern delivers the best results in each band. Gain addition and subtraction in each band can be adjusted too. Muting and soloing the bands helps you hear exactly what the individual processes are doing. The diffuse field/free-field idealised response equalisation options provide subtle differences in tonal voicing and the proximity effect control certainly does what it says on the tin. The more exotic spill-minimisation process is a two-stage affair where the desired signal is fed in and ‘maximise target’ is activated. The plug also wants you to feed in undesired spill after which you can satisfyingly hit the ‘terminate spill’ button. It all gets very Arnold Schwarzenegger here, and the end results vary but are often surprisingly effective. AT 40
I was also impressed with the overall sonic ‘image’ the mic presented, and the smooth way it handled transients and off-axis sources.
CAREFUL WHERE YOU POINT THAT THING!
While all these controls are extremely welcome to experienced engineers, a few words of caution primarily aimed at newer players are warranted. The first thing to note is that after a few minutes of twiddling PolarDesigner’s parameters it is very easy to end up with a sound that is somewhat compromised and a million miles away from what you originally tracked. Big variations in polar patterns from band to band come at a cost and if you’ve got all five bands engaged and furiously tweaked you’ll be introducing a lot of processing anomalies at the crossover points and a general ‘pinching’ of the sound will be evident, despite Austrian Audio’s claims of phase linearity. ‘Termination’ of spill can also really chop out significant areas of your tone, and if they are mission critical areas you might be better off learning to love the spill you had. Having said all that, there are incredible upsides to the PolarDesigner and I think it’s a brilliant bit of audio design that will have a host of applications. The one that immediately sprang to my mind was the eternal spill battle between vocal and acoustic guitar mics during live takes. I did some guitar miking tests while singing with the mic’s front capsule aimed at the guitar and the null point aimed at my face. I found that tuning the ‘presence band’ (3kHz to 8kHz) of the OC818 to a tight figure-eight pattern kept vocal sibilance to a minimum, while the cardioid pattern below 3kHz gave me a full-bodied lower midrange and a little more air and ‘bloom’ from the instrument. After much tinkering I found the most natural results occurred using the minimum amount of bands for the job. The PolarDesigner plug-in certainly opens up a plethora of possible ways to optimise results in such situations and encourages thinking ahead with mic positioning to make the most out of the plug-in’s sonic possibilities. Perhaps best of all, the key variables of the PolarDesigner are all accessible via automation. This means you can selectively morph polar patterns, gain levels and proximity effect during a mix. For example a cardioid live vocal take can be shifted to omni for an instrumental section to add extra room energy, a piano overdub can slide towards omni for a more roomy background feel during vocal sections and focus back in to hyper-cardioid for its solo. Drum overheads and distance-miked amps can be pushed back into the room or forward into sharper focus.
Really, the sky’s the limit and of course any music technology this powerful is begging to be creatively misused by artists and engineers looking for a new edge. BALANCING ACT
Over the next few weeks I gave the OC818 a series of heavy workouts on vocals, mono drum overheads, electric and double bass, electric guitars, strings, percussion and piano. Whether in its single- or dual-channel configuration, the mic repaid me with great tonal balance and a natural yet pleasing sonic quality that always seemed to fit effortlessly into overdub contexts regardless of what the existing material was. As the days went by I realised I wasn’t reaching for EQ much either. The last mic to catch my ear like this was the Soyuz SU17 and that’s a valve model worth more than twice the price of the Austrian Audio — the common denominator being that these are both hand-built designs made by audio obsessives with decades of European experience behind them. Just to make sure I had fully taken the OC818’s measure, I tracked an entire song using just this mic (a process that I find exaggerates any weaknesses in a microphone as problematic frequencies and resonances build up). Sure enough the mic came through with flying colours yet again. The track stood up on its own so well it didn’t require any real help with EQ at all, something of a rarity in my experience. This last bit of tracking re-confirmed my high opinion of the mic and sent me checking my bank balance — always a danger for reviewers! The consistent quality of the sounds the OC818 delivers over a wide variety of applications is a real testament to Austrian Audio’s new capsule design – the results really are quite stunning. With its easy authority on key instruments and vocals, the smooth way it handles transients along with its generally un-hyped but sweet tonal signature, the OC818 is one of the best-sounding new microphones I’ve had the pleasure of reviewing over the past 15 years. Austrian Audio has delivered one heck of a mainstay studio mic here at a very reasonable price point. Throw in the innovative and comprehensive digital control over multi-band polar patterns and you can be sure the competition will be sitting up straight in their chairs. Time will tell, but with the Austrian Audio OC818 my feeling is that we may indeed be witnessing the birth of a modern classic.
We’ve just unboxed a new addition to our family Hello HARMAN Professional Solutions!
MadisonAV appointed the Australian distributor of HARMAN Professional Solutions. We’re thrilled to be the newly appointed distributor of AMX, SVSI, and install ranges of AKG, BSS, Crown and JBL Professional in Australia. So in the spirit of great changes, Madison Technologies’ dedicated AV business will now be trading as MadisonAV. Sales Enquiries 1800 00 77 80 www.madisonav.com.au AT 41
REVIEW
ANTELOPE AUDIO DISCRETE 8 SYNERGY CORE
Thunderbolt/USB Interface Antelope takes interface processing to new heights by ‘synergising’ DSP and FPGA in the one box.
NEED TO KNOW
Review: Preshan John
PRICE $2650 (expect to pay) CONTACT Turramurra Music: (02) 9449 8487 or hitech_sales@turramusic.com.au
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PROS • DSP effects sound great • Outstanding DAC conversion • Loads of I/O in small form factor
CONS • Software support a little unstable at times
SUMMARY A 32-input interface is just the beginning — the real value of the Antelope Discrete 8 Synergy Core is in its substantial DSP power, in addition to the brand’s established use of FPGA chips. The onboard effects sound incredible, as does the hardware itself. Behind the unassuming exterior is an extremely capable interface.
I would be happy to track all my sessions through these plug-ins. They’re that good.
I remember the day my Antelope Audio Zen Tour desktop interface arrived in a parcel and I rushed home to set it up in my small studio setup. Hearing audio for the first time through the Zen Tour’s pristine conversion was a revelatory moment, for me — like a blanket was lifted from my studio monitors and I was finally listening to the real thing. I’m a believer in the Antelope difference. Besides the impeccable conversion, another Antelope selling point is its use of built-in FPGA processing chips. Interfaces such as the Zen Tour, Orion Studio and Goliath HD all possess internal processing power to run Antelope plug-ins in realtime thanks to these powerful chips. Now with a new family called Synergy Core, Antelope is exploring the world of DSP. Unlike FPGA chips, digital signal processors are a form of microprocessor which excel in handling larger amounts of processing in real-time — think: UAD interface systems. We received the Synergy Core Discrete 8 interface for review. Two DSP chips within can run up to 64 instances of effects, at any sample rate; to give a stated “eight times the processing power of the Discrete Basic”. FPGA technology still makes an appearance — hence the ‘Synergy’ title. Capitalising on the strengths of both processing types, the single FPGA chip takes care of internal I/O routing and recreation of studio effects and mic models for use with Antelope’s Edge and Verge modelling microphones. The Discrete 8 Synergy Core offers both USB and Thunderbolt connectivity (including Thunderbolt for Windows) and includes eight built-in, digitally-controlled transistor preamps. Key information is displayed on a screen along with a row of eight detented gain knobs, two mic/ line/instrument inputs, two headphone outputs and a pushable control knob. You also get two ADAT inputs/outputs, two reamp outputs, eight DCcoupled line outputs via DB25, and SPDIF in and out. Considering the numerous I/O options, DSP power and high end specs, the Discrete 8 Synergy Core is a formidable beast for a 1U interface. GRUNT WORK: BOOT CAMP
Despite all that power under the bonnet, trying to boot the thing up wasn’t straighforward. I followed the instructions on Antelope’s website to start the Launcher app which automatically detected the connected Discrete 8 Synergy Core. After picking the latest firmware version and clicking update, the Launcher kept stalling at a certain point, freezing up the GUI on both the
application and the interface’s screen. I tried all the usual troubleshooting hacks to no avail. After six attempts I took up my case with Antelope. After going back and forth with tech support, the only way I could update the firmware was by running the Launcher on another computer — and only then did the interface leave its frozen lock-out mode and accept the firmware update. Unfortunately, this kind of drama isn’t unheard of with Antelope products. I had a similar experience with my Zen Tour. Clearly, it’s something Antelope needs to address. RECORDING: 888
Once the Discrete 8 was firing, my first proper hitout was at a local performing arts centre for a large live recording session. The full band lineup called for no less than 24 inputs, so I supplemented the eight built-in pres with another two eightchannel preamps via the dual ADAT inputs. A baptism of fire, to be sure, however, its performance contributed to the resounding success of the session. A well-considered routing program makes a huge difference when handling 24 inputs. The Antelope software is very functional with an easyto-read GUI and cleaner layout than the version that comes with my Zen Tour. Gains for the eight onboard preamps can be set along the top section while levels for a few separate mixes are set with the faders in the middle section. Whipping up a couple of monitor mixes was easy. A little switch icon toggles between displaying the built-in eight or full 32 channels of I/O. DSP + FPGA
The real story behind a Synergy Core product is the processing and effects. The Discrete 8 includes 36 DSP-powered effects (54 plus AFX2DAW with the introductory offer). These are available across the eight built-in preamps. You can build virtual channel strips within the control panel software in the row marked ‘AFX’, or select from a list of presets. As the name suggests, the AFX2DAW plugin allows you to run Antelope’s FPGA software as inserts within your DAW session without burdening your computer’s CPU. Clicking in the AFX box of a track opens a separate window in which the actual channel strip can be configured (fancy GUI and all). An at-a-glance view is provided in the main window showing the chain of effects on each track, along with a BP switch to bypass all processing. The effects menu leaves little to be desired. I can’t get enough of the old-school models under Vintage
EQs, Vintage Compressors and Vintage Preamps. All possess a distinct and usable character that’s convincingly analogue. More utilitarian tools are available too, like a parametric EQ and noise gate. In the tracking room, Antelope’s DSP effects are nothing short of stunning. I put a couple of the vintage compressors and preamps to task on a basic acoustic guitar and vocal tune. The result was similar to taking a 2D photo and putting it through a lenticular lens, imbuing it with a sense of depth and three-dimensionality. In fact, it contributed so much warmth and richness to my recordings that I sorely missed it when bypassed. One of my favourite features of the Zen Tour is AuraVerb, Antelope’s ‘comfort’ reverb, which also runs off internal DSP grunt. AuraVerb is on tap for all 32 inputs of the Synergy Core Discrete 8. Access it by clicking the AuraVerb logo at the bottom right of the control panel, then adjusting individual send amounts. Tonally the reverb can be tweaked to your liking with plenty of flexibility (Colour, Predelay, Early and Late Reflections Gain controls, Richness, Reverb Time and Room Size). A touch of reverb in headphones while tracking is a sure way to please a vocalist. RUN DSP
The promise of mass DSP power is why most people, myself included, will be interested in Antelope’s Synergy Core products. In real world scenarios it means you’re putting better-sounding tracks to tape because they’re running through high quality emulations of vintage gear, all of which is supplied to the artist with zero latency. In I/O and horsepower terms, you won’t find another system for the same price — a UAD system, Antelope’s direct competition, will set you back significantly more money. Of course, none of that power would matter if the effects were second class — and they’re not. I would be happy to track all my sessions through these plug-ins. They’re that good. I really wish my experience in launching the device wasn’t quite so painful, otherwise I’d have absolutely no hesitancy in totally recommending the Discrete 8 Synergy Core. But a little like childbirth (so I’m told), the birth pangs are soon forgotten once you’re using the interface and enjoying its power and sound. With this amount of power on tap, it represents massive value, and as long as Antelope commits to ironing out the wrinkles in its hardware/software integration, this rig demands consideration.
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REVIEW
SENNHEISER HD 25 LIGHT Closed-Back Headphones The sound of a classic at an accessible pricepoint, Sennheiser’s HD 25 Light proves its worth as more than just a DJ headphone. Review: Preshan John
TAKE YOUR PICK The Sennheiser HD 25s are available in three offerings. While the new HD 25 Light is cheapest, the established HD 25 model sports a single sided cable, split headband and rotatable ear cup. The HD 25 Plus is the deluxe package — same headphones but with an extra cable, extra ear pads and carry pouch thrown in.
NEED TO KNOW
Lightweight and unobtrusive, Sennheiser’s HD 25 on-ear headphones may not impress with their looks but effortlessly impress with their sound. Those little earcups deliver big sound and a thumping bottom end without sucking anywhere nearly as much amp juice as your reference open-backs. In fact Sennheiser boasts of the HD25’s ability to handle sky high SPLs without exploding, making it an ideal tool for audio monitoring in various applications – from a recording engineer to a cameraman to a DJ. Sennheiser has long considered the HD25 a ‘DJ headphone’ because, besides the punchy bass, it has a rotatable ear cup and both earcups slide straight off the headband to allow for walkie-talkie style one ear listening. Newly added to the family at NAMM 2020 is the HD25 Light – the cheapest route to HD25 sound but with a simplified headband that still lets the earcups slide off, but not rotate.
PRICE $179 RRP
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CONTACT Sennheiser: (02) 9910 6700 or sales@sennheiser.com.au
ON EARS
I spent time listening to my favourite playlists through the HD25 Light and found them an easy listening headphone that left my desire for clarity satisfied. Sennheiser states it has the same driver as the original HD25 however a close look at the spec sheet reveals slight differences in impedance (60Ω rather than 70Ω), and frequency response (16Hz22kHz compared to 16Hz-22kHz) – perhaps due to a variation in circuitry. While I didn’t have a ‘normal’ HD25 to A/B with the Light, there’s a definite shyness in the high end that I’d otherwise have put down to the closed back design. Rather than being open and spacious, the soundstage feels more enveloping and intimate. At times the blossoming bass presence was a touch excessive, especially in the low mids, but if you’re a DJ this is the last thing you’d complain about.
PROS • Impressive tone with gutsy lows • Lightweight
CONS • Headband feels a little delicate
Comfort is average over long listening periods with a slightly tight fit on my head. If you care about comfort you can swap out the stock synthetic leather earpads for Sennheiser’s softer velour option. The skinny plastic headband feels flimsy when bent so I’d be wary of tossing the HD25s in a bag of cables or hardware, where they could get damaged. Excellent isolation from surrounding noise combined with minimal audio bleed make the HD25 Light great for the tracking room or a drummer on a live stage — or even as a casual listening headphone on public transport. It’s a no frills, no fuss pair of cans. They might be pitched as DJ headphones but don’t let that turn you off. The Sennheiser HD25 Light is equally indispensable to an audio engineer.
SUMMARY Sennheiser’s HD25 Light won’t disappoint as a trusty ‘desert island’ monitoring tool for audio engineers or recording musicians. Professional enough for critical listening yet portable enough for field recording or live performance, the HD 25 Light is an affordable all-rounder that sounds bigger than its size.
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REGULARS
PC Audio Sometimes PC plug-ins can have almost subliminal effects, but in combination these can become sublime. Column: Martin Walker
Many PC effect plug-ins now arrive with serious audio performance that would have been unheard of a few years ago and, when modelled from hardware gear, can sound uncannily close to their original inspiration. Nevertheless, where hardware and software versions of the same item of gear are compared in detail by reviewers, the software version invariably gets close, but will never get the cigar. Now you may think there’s an element of politics involved in such statements, but I firmly believe that there are also more subtle forces at play. Despite both hardware and software plug-in alternatives seemingly offering identical specifications, in hardware the result is a cumulative one, resulting from the many smaller audio changes that happen at each electronic stage. So, for example a typical low-end frequency response of perhaps -3dB at 10Hz in a hardware unit is achieved via a combination of various interstage coupling capacitors (each providing its own cutoff characteristic), along with the low-end rolloff of input and output transformers and the like. Each of these components and others will alter such things as frequency and phase response, harmonic distortion and so on in subtle ways, and during the hardware design phase, final component value tweaks across all the stages are only made after lots of detailed listening tests. On the other hand, while some plug-in designs (particularly the more expensive ones!) do claim to model the hardware on a component by component basis, many plug-in equivalents are likely to simply roll off the low end with a suitable algorithm – the result may still be a rolloff of -3dB at 10Hz, but a lot of the audio hardware subtlety may be lost in the process. GO FORTH & MULTIPLY
Hardware or software gear can also be cascaded in series, such that each device adds a little to the overall effect. I’ve heard this technique called ‘sonic varnishing’ — adding layer upon layer of more subtle effects to hopefully achieve audio nirvana. This can, for instance, work well with compression, by letting each device in the chain tame a part of the sound; or with reverb, where you can end up with more complex and individual acoustic environments by combining different aspects from each. Additionally, the software plug-in can be used in a way that’s less feasible with hardware, and that’s to keep adding multiple instances to each and AT 46
every one of your playback channels. For most of us this would be prohibitively expensive to do with hardware, but it’s the perfect opportunity to let software plug-ins shine. The most obvious examples are the proliferation of ‘console’ plug-ins that add some ‘flavour’ to your mixes. These will typically roll off the frequency response at low and high extremes, add a dash of benign harmonic distortion for added richness (particularly at the low end), and while the difference may be extremely subtle on a single channel, once added to every channel, the sonic difference is more pronounced. There are lots of such plugins on offer, many of which recreate the entire control set of famous console designs, complete with sections for EQ, compression and so on, while some also offer other analogue subtleties: such as small random variations in frequency response from one instance of a console channel EQ to another, to mimic the inevitable spread of electronic component values that you get in real-world circuitry. This randomness can ‘flesh out’ electronic sounds in particular, and such variations between the left and right channels of a stereo EQ will convert a straight down the middle mono centre signal into a more fleshed-out central mono mass. Nevertheless, you don’t necessarily need all this functionality to explore the essential analogue console ‘sound’. If you want to dip your toe in the console waters without spending any money, why not download Airwindows’ Channel7 (www. airwindows.com/channel7), which offers a choice of three basic console types along with a drive control that morphs from perfectly clean, through very pure saturation, to more obvious fatness. I’ve mentioned Chris Johnson of Airwindows before in this column, because his plug-ins often achieve subtle yet beneficial results due to novel algorithms of his own design. He deliberately avoids the slavish modelling of real-world audio artefacts in favour of novel combinations of his own mathematical operations, and, as always, the truth of the algorithm is in the listening. THE NEXT STAGE
Another fascinating example from the Airwindows stable is the recent Interstage plug-in (www. airwindows.com/interstage), which aims to sound
like running your audio through an ‘optimal analogue stage’. It offers no adjustable parameters at all, but performs its magic by reshaping the low end, similar to that of a capacitor-coupled circuit, while the highs runs into ‘active component electronic limits’, by ‘restricting treble slews based on the general amount of energy in the circuit’. Essentially you’ll hear no change if you don’t need what it’s doing, but Interstage will kick in when it detects excessively digital bass and treble in your audio, especially once audio levels are approaching 0dBFS. Once again, the results are subtle, but nevertheless fascinating, and when switching Interstage in and out blind, I’ve often found myself preferring it in circuit due to its goal of converting any overtly digital sounds into more analogue equivalents. WINDING THINGS UP
Another aspect of audio components that can add cumulative mojo is the sound of transformers, and in the plug-in arena I’ve been impressed by Kazrog’s True Iron plug-in (kazrog.com), which models the euphonic effects of audio transformers. Unless you push the parameters quite hard for special saturated effects, the results are again very subtle, yet once again if you add a single instance of True Iron on each of your playback channels the cumulative results slowly build into a warm roundness that fits beautifully with many musical genres. True Iron can also knock the perfect edges off clean softsynth sounds for a little more grit and attitude. One important aspect with all cumulative plugin use is to be careful about gain-staging. All the euphonic effects described in this column will vary considerably with input level, so it’s important to spend some time discovering the ‘sweet spots’. After all, we’re only looking for subtle changes for each instance. I don’t necessarily add exactly the same treatment to every channel either, as some (bass instruments and drums for instance) may benefit from more transformer warmth, while others blossom with console tweaks. It can be a good idea, once you’ve got everything as you want, to render the song, then temporarily deactivate all the mojo plug-ins and render it again, so you can hear the difference properly and get your ears more finely attuned to these ‘analogue improvements’.
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