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CONTENTS89
40
THE PHANTOM IN THE OPERA THEATRE Sydney Opera House uses wave-field synthesis & 97 speakers to recreate a phantom live orchestra
Jack White talks the making of Blunderbuss on 8-track, why he uses tape, and staying creative.
FEATURES 50
YAMAHA CL CONSOLE
34 44 56 58
Dale Sherlock Unlocks Amps Vance Powell Engineers Jack White’s Blunderbuss Jim Moginie’s Colour Tone Theory Mars Volta’s Juan Alderete On Effects
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What’s On PC Audio, Ivy Bridge Last Word, Cubby Colby Stav’s Word, Guitar Heroics
REVIEWS 82 Quest HPI Speakers 86 d&b V Series PA 88 Eve Audio SC205 Nearfield Monitors 90 Tascam Portastudio DP-24 94 Fractal Audio Axe-FXII Guitar Processor + Roundups: Wireless Guitar Gear, Reampers & Loop Pedals
SEE PAGE 97 AT 5
FEATURE
DALE SHERLOCK UNLOCKS AMPS One of Australia’s best boutique guitar tube amp builders opens up on an amp that does it all, what makes up an amplifier’s tone, a love of tubes, and his general bias for the right bias. Story: Mark Davie
“Fender wanted to electrify instruments because musicians needed their instrument to be louder. Using the only technology available at the time, electron tubes, he put everything into making them sound great. Fender created not only great amplifiers, but an art form which still exists today. This is what occurs when you have someone passionate about something. They produce something great.” That was Aussie tube amplifier designer and builder Dale Sherlock reflecting on the origins of tube amplifiers and Leo Fender’s contribution to the industry. Like Fender, Sherlock is passionate about the art of amplifier building. He’s not another copycat or cloner. His main interest lies in creating things that are truly unique. Armed with a knowledge base of over 30 years of tube amp experience coupled with the tight tolerances of an aviation engineering background, and decades of finding and fixing faults in just about every tube amp there is, he’s built the ultimate tube amplifier — the Fat Head. His other amp creations include the Buff Tone, Buddy, Angry Ant plus countless amplifiers customised for musicians throughout Australia. The Fat Head in particular is an amplifier head with four different circuits in one box that takes you on a tone journey anywhere from belllike cleans to super hi-gain and everywhere in between. He even has a patent for one of his inventions, the Tube Management System (T.M.S.) included in the Fat Head. The great thing about boutique amplifier builders is that you’re going back to an age of passion. While larger more widely known amplifier manufacturers are increasingly cutting corners to maintain profit levels as their costs rise, builders like Sherlock will scour the globe for quality components and have transformers made to exact specifications. It’s all in the name of allowing the art form to develop great amplifiers, rather AT 6
than being controlled by the dollar. Sherlock’s amplifiers are built to the standards of the era when tone was king. The flexibility of the Fat Head is outstanding, to a level that goes much deeper than twiddling a few tone controls. The Tube Management System (T.M.S) gives the lay-guitarist the ability to bias output tubes themselves. It even allows different tube types to be used in unison, the user to be able to set the bias from Class AB to somewhere closer to Class A, and even run Simul-class (two tubes in Class AB, and the other two in Class A). Truly giving the guitarist total control over their tone. NOT AS EASY AS IT SEEMS
To create a unique circuit is not a simple matter. Some amp builders will use someone else’s design — because they know it will work and will give a particular result — without necessarily knowing why or how the circuit actually works. Sherlock, however, shows there’s still room for ingenuity. “There are the common tube amp circuits around” said Sherlock. “When you buy an IC (Integrated Circuit) or transistor for example there’s a recommended circuit and tech data on how to operate or set a circuit up for it supplied by the manufacturer. The RCA engineers for example, a tube manufacturer at the time of Leo Fender, gave the technical people that bought their tubes information on how to build circuits to operate them. “What Fender and other builders did was use the basic circuits supplied by the engineers. He modified them to suit his requirements and wired them all together to create amplifiers. It was the same with Marshall and all other builders, even today. What all amp builders do is connect these common circuits in ways that are unique. This is achieved by changing component values like resistors and capacitors and by adding other circuits like tone stacks and other sound shaping filters. This is what primarily gives each amplifier its unique and distinctive tone.
“A tube amplifier is something that increases its volume or signal level from something small to large hence the name amplifier. People think the tube itself makes it sound like a certain amplifier, and to a certain extent it does, but it’s mostly the amps circuitry that makes it sound like a certain amplifier. If you put a Marshall tone stack in between two amplifying stages of an amplifier its overall sound would be more like a Marshall than if you put a Fender tone stack in between the same two stages of amplification. It’s the circuitry that shapes the sound of the amplifier — the tube just basically amplifies. However this is not completely true in that all tubes do have their own particular tonal characteristics and contribute to a lesser extent to the overall sound of the amplifier.” THE 12AX7 FAMILY
The 12AX7s are the most common amplifying tubes (or valves as they are known in some countries) found in tube amps. The “12AX7 tube is the highest gain of the 12AX7 family, 12AT7 is lower in gain, the 12AY7 is lower again and the 12AU7 is the lowest,” said Sherlock. “These tubes basically sound similar but have different gain structures and are designed for different circuit applications. Putting a 12AT7 in an amplifier circuit won’t result in a Marshall sound. Likewise, putting in a 12AX7 wouldn’t result in a Fender sound. A single stage of amplification, for example a 12AX7 triode, will evenly amplify the full range of frequencies put through it. However if you add tone circuitry, it shapes that range of frequencies to give a certain contour. A Marshall contour is lots of mids and tops, where as a Fender is more of a scooped sound with the bass and the treble up, and the mids down.” STACKS OF TONE
What Sherlock is referring to as one of the things that shapes the sound of a guitar amplifier is the tone stack. Basically a stack of tone controls comprising of treble, mid and bass controls, which is usually a capacitor and pot for each frequency range. There are five main tone stacks used with variations of these. The most familiar are the three-knob versions seen on Fender and Marshall Amps. Though they’re not identical, they both carry one problem. That is being of a passive nature, the controls are not fully independent of each other, of which the mid control is the worst, and actually quite ineffective. If you’ve ever wondered why you tend to get broadband boost whenever you whack up the mid control on your Twin or JTM, it’s because the mid control acts more like a volume control, affecting most frequencies. Though it’s also the only one of the three knobs that affects the mids too. Then there’s the Baxandall, which you’ll typically find in hi-fi equipment, but also makes an appearance in the Ampeg B15 bass amp, among others. In use, the Baxandall has only two tone controls, one for bass, and another for treble. The result looks like two shelving filters usually centred around 1kHz. With this configuration, to get a mid boost or cut, you just have to either
turn both the treble and bass controls up to get a mid cut, or vice versa for a mid boost. Obviously it also requires more gain to actually boost the mids. The Baxandall can also leave your signal flat with the controls in the centre position, while the Fender and Marshall tone stacks will typically exert some kind of shape over your sound whilst being in the flat position. There are plenty of other variations as in when VOX added a top boost to their AC30 model; it was essentially half a Baxandall. While the normal input has a larger .047uF series pass capacitor in place, which lets pretty much everything above 150Hz through at full gain, the bright channel has a smaller 500pF bypass capacitor in place that acts as a high pass filter that can add 6dB gain above 1kHz. And there are also Butterworth filters that constitute a high or low pass filter, while leaving the pass band flat. “All those different tone stacks will give you certain tones,” said Sherlock. “The tone stacks used in Fender/Marshall are generally passive in that they are not in feedback loops and operate like frequency dependent volume controls. The Baxandall tone stack however operates differently in that it’s usually (but not always) incorporated into the feedback loop of solid state hi-fi amplifiers and therefore gives greater control over the adjustment of the frequencies. The passive tone stacks are fine for most guitar amps because you generally don’t want massive amounts of bass boost for guitar anyway, which are more of a mids and top end instrument. So passive tone stack works well for guitars, and is why everyone uses them. “The components are selected to give the tones that everyone is used to. In fact these passive tone stacks have helped define the sound of the electric guitar in contemporary rock.” FOR THE LOVE OF IT
Sherlock loves tube amps. He makes a living from them, but “you have to do it for the love”, he says. To explain he grabs a small black plastic-covered device the size of a baby incisor in one hand, and with his other hand he fishes out a pre-amp tube roughly the size of an adult pinkie. “This is a transistor,” he says, holding up the 3-legged black chip. “It was supposed to supersede or render the tube obsolete, which it has been largely successful at doing but not completely in the case of high end audio. This transistor requires a fraction of the voltage to operate, it doesn’t require any heater current to heat it up, and the size is a fraction of the size of the valve. Now, with IC technologies we can fit thousands of those small transistors into one chip. So why would you want to use this old tube technology these days when this technology is available?” The answer is simple. “Because tubes sound great.” “Although some computer software applications and tube simulator amps come close at mimicking the sound of tubes, there is still nothing to this day that beats the mighty electron tube. That’s the AT 7
FEATURE
Jack White’s debut solo album Blunderbuss is his first to reach No 1. It’s also the first album recorded to 8-track tape to make it to the top in decades. Jack White uncovers his preferred working method; why he finally built his own studio; and how Bob Ludwig bypassed the Loudness Wars. Story: Paul Tingen
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In the feature film It Might Get Loud (2008), which features guitarists Jack White, Jimmy Page and The Edge, White makes some striking statements about music technology. He declared that “technology is a big destroyer of emotion and truth,” which leads to “a disease you have to fight in any creative field: ease of use.” White has also stated that he feels that he can’t be proud of something unless he “overcame some kind of struggle,” and in the movie demonstrated his point with his famous red JB Hutto Montgomery Airline, a kitschy, low-quality plastic guitar which was sold via mail order during 1958-68, and which is regarded as difficult if not impossible to play. “You have to pick a fight with it, and win,” clarified White. During another scene in It Might Get Loud, blood is seen streaming from White’s picking hand, and there’s a shot of one of his electric guitars covered in congealed blood. White’s deliberately contradictory attitude to music making in general and technology in particular informs almost everything he does. Perhaps it originated from his early days in his native Detroit when he was an upholstering apprentice, which involved a lot of picking fights with matter, and the occasional stains of blood. It showed in The White Stripes, in which he made his life difficult not only by playing the JB Hutto guitar, but also by playing with only one other musician, a drummer who was a beginner when the band started, and whose style has been called ‘primal’ and ‘simplistic’. It also showed in White’s often stated refusal to become a studio gear head, as well as his preference for recording on analogue, a medium that involves a lot more of a struggle than working with a DAW (barring the odd computer crash, of course). However, in 2009, White decided to build his own studio, Third Man, which meant throwing at least one of his principles overboard as it involved him having to engage with studio gear on a much deeper level than ever before. The creation of Third Man Studios was related to White establishing a physical location for Third Man Records, the independent label on which all six White Stripes studio albums have appeared, as well as the output of the other two bands White is involved in: The Raconteurs and The Dead Weather. However, in recent years the output of Third Man Records has multiplied dramatically, with White producing singles and/or albums by the likes of Wanda Jackson, The Black Belles, Karen Elson, Loretta Lynn, Seasick Steve, Jerry Lee Lewis, Stephen Colbert, Alabama Shakes, and many others, plus White’s first solo album, Blunderbuss. Whether the studio informed the many releases, or the other way round, is probably a chicken and egg situation, but it appears that Third Man Studios has turned into Jack White’s Mad Scientist’s Lab. On the phone from Gulf Shores on the Alabama coast, where White was about to play a concert on a festival on the beach, he elaborated on the why of his own studio and how it changed his relationship with technology. “In the Third Man Recording headquarters I have what is probably the only live venue in the world where you can record to tape. It’s an actual live venue with a recording booth attached to the stage. But my recording studio is in a different location, and I’ve done all my recordings there since I built it three years ago. For the longest time I did not want to have my own studio gear, mostly because with the White Stripes
I wanted to have the constriction of going into a studio and having a set time of 10 days or two weeks to finish an album, and using whatever gear they happen to have there. I recorded the second White Stripes record De Stijl, in my living room, and we would record a song, and a couple of days later another song or two, and then a week would go by and we would record again. When having a studio at home it is too tempting to get distracted. The phone rings, somebody knocks on the door. So I said to myself: I can’t do this shit anymore, and I decided to record in commercial studios from then on. But after 10 to 15 years of recording like that I felt that it was finally time for me to have my own place to produce music, and have exactly what I want in there: the exact tape machines, the exact microphones, the exact amplifiers that I like, and so on.” Paul Tingen: Your studio is built around a 1073-based Neve console and two Studer A800 2-inch 8-track tape recorders. Why analogue and why only eight tracks for most of your recordings? Jack White: I love analogue because of what it makes you do. Digital recording gives you all this freedom, all these options to change the sounds that you are putting down, and those are for the most part not good choices to have for an artist. For example, if you record a vocal onto tape and there is one word wrong, it is too dangerous to edit it out and fix it with a razor blade. So a lot of times you will leave things in that inherently are a good idea to leave in. But if you’re working in Pro Tools, you are going to Auto-Tune that note or use a bunch of other plug-ins to get that note to sound the way you want it. But these attempts to make things sound perfectly in time and perfectly in tune are all huge mistakes, because you are taking away all the inherent soulful qualities of what is going on. Plus the plug-ins you may be using are all emulations of real things in real life, like reverb, or tape delay, and so on. All those plug-ins have adjustments on them, and none of them are mechanical. I think that when you get out of the world of mechanics, you start to lose the inherent beauty of the sounds. I can’t work in this scenario, but I’m not saying that other people should work like I do. If they think they can make something beautiful in the digital realm, more power to them. Regarding my preference for eight-track recording, it helps me make decisions. If there are only two tracks left, what am I going to do with them? I can do a guitar solo and a vocal harmony and that’s it. But if that were on Pro Tools, I would have 200 more choices. Let’s put another tambourine track on, let’s put four more guitar tracks on, and double that snare drum, and all of a sudden you’re putting all this shit on that you should never have put on to begin with. Opportunity has killed it. If you don’t have all these choices, your mind will focus on better things. PT: Wasn’t the last White Stripes album Icky Thump (2007) recorded by Joe Chiccarelli at Blackbird Studios on two 16-track recorders? JW: The Icky Thump situation was a debate between Joe and I. He wanted to do drums on six tracks, and I only on one track, but I did not feel like fighting him on it. It was a matter of trying to find the biggest headspace without going to a 24-track headspace, so I decided that 16 tracks would be a good compromise. I had done many records on eight tracks by that stage, so I allowed myself some room with albums
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FEATURE
MAKING OF BLUNDERBUSS
VANCE POWELL Vance Powell has been White’s engineer on over 100 records in a three-year period. He talks about the making of Blunderbuss and how analogue recording doesn’t hold him back, but helps them make records faster. Story: Paul Tingen
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Jack White may be old school, in the best sense, but on his first solo album, Blunderbuss, he appears to have taken his principles to new levels. The album has benefitted from being recorded with little noticeable compression, and mastered without any compression or limiting, which allows the music room to breathe. The album is not only a refutation of loudness orthodoxy, but in having been recorded on 8-track analogue and mixed to analogue, it also throws down a challenge to the entire 21st century record-making approach, that increasingly relies on a DAW’s endless tracks, endless editing and endless mixing capabilities. If these facts don’t give pause for thought, a listen to the album will. Regardless of whether the music is your cup of tea, there’s no denying the album’s sonic qualities, somehow combining the velvety lushness of bygone days with a 21st century presence and in-your-faceness. The mix is striking too: drums are centre-panned mono, and
the wide sonic panorama has been achieved by very discrete panning of mono instruments. At the controls during the recording and mixing of Blunderbuss, and therefore ideally placed to relate the full inside story of its making, was Vance Powell. The Joplin, Missouri, native had a background in computer programming, and live sound as well as studio engineering, when he was asked in 2001 to help set up Blackbird Studios in Nashville. During his tenure as chief engineer at Blackbird the studio expanded to one of the most important recording facilities in the US, with a strong presence of esoteric and analogue gear. In 2006 Powell joined Mitch Dane and set up his own Nashville recording facility, Sputnik Sound. Powell resigned from his post at Blackbird in 2010, because he “wanted to make records again,” rather than run a studio and train assistant engineers. Powell’s first contact with White dates from the same time he set up Sputnik when he worked on Danger Mouse and Daniel Luppi’s
Rome album, on which White guested (and which was finally released in 2011). Following this White asked Powell to mix the White Stripes’ Spanish-language version of the song Conquest (2007). Powell and White clearly hit it off, and the former now divides most of his time between Sputnik and White’s Third Man Studio, which is also located in Nashville. ANALOGUE ALL ROUND
Powell’s room at Sputnik has a Neotek Élan desk, and Studer A800s, with 16-track and 24-track headstacks, a pair of Studer A80s (½-inch and ¼-inch), a ¼-inch Studer A820, an Ampex 1-inch 8-track, and a very impressive 6-foot high outboard rack that he calls his ‘Wall of Rock’. Powell has recorded dozens of records with White at Third Man Studio, where the gear consists of a 16-channel, 1073-based Neve console originally from SABC in Johannesburg with labelling still in Afrikaans, plus two Studer A800 2-inch 8-track tape recorders, an ATR102 1-inch, lots of outboard and a choice selection of mics, many of them ribbons. So what motivates Powell, and presumably also White, to make their lives difficult and expensive with all those analogue reel-to-reels that are surely obsolete by now? There’s a whole posse of former analogueloving engineers out there that will tell anyone who’s willing to listen that hi-resolution digital and the latest generation of A/D and D/A converters have become so good that tape has become an irrelevant relic from the past. Powell, however, is having none of it… “Analogue still sounds better. Absolutely. I also have a ProTools rig here at Sputnik, because I’m pragmatic. I’m not some sort of crazy, throwback, retro analogue freak. In today’s markets you have to have an exceptional working knowledge of ProTools. And I like to have an exceptional working knowledge of analogue as well, because it’s my preferred way of working. People often come here with their projects on ProTools, or they need to take their project elsewhere, and so I have a very simple process of doing transfers between tape and ProTools. I CAN ALSO SWITCH BETWEEN PLAYBACK FROM TAPE AND PLAYBACK FROM PROTOOLS AT THE PRESS OF A BUTTON, AND I CAN TELL YOU, THERE’S A MARKED DIFFERENCE. But for me it’s
not only about the sonic texture of analogue. I think analogue is far superior for other reasons as well. Most of all, it has a far superior workflow. People sit around looking at waveforms, lining up kick drum and bass notes visually. But music was never meant to be looked at. “Someone in my studio the other day said the greatest quote I ever heard: ‘HAVING LOTS OF TRACKS ON A SONG IS LIKE PUTTING A LOT OF STICKERS ON YOUR CAR TO GIVE IT MORE HORSEPOWER’. If I have a
mantra, it is, ‘Just because you can do something, doesn’t mean that you should’. Just because you can put 20 channels of loops together, or can play Wurlitzer, Rhodes, piano, B3, and cheesy ’80s synths all at the same time, doesn’t mean you should. Just because you can make 50 play
lists of the same solo, doesn’t mean you should. That is not good record-making. It is poor record-making. Your brain can’t hold all that information. By contrast, analogue forces the producer’s and the artist’s hand. It forces them to make decisions. It forces the engineer to put a lot of sound into a small space and make it sound large. You have to record things to tape that sound good, because there’s no endless editing or messing with the sound afterwards. What’s there is what’s there. “The other thing is that you have a limited amount of tracks, and that forces the producer and the engineer to think ahead. If you have two tracks left and you still need to record background vocals and tambourine and a few other things, how are you going to do that? Maybe the background singers have to play the tambourine while singing. These decisions become part of the tapestry of the song. It’s not a limitation to not have endless tracks, it’s an opportunity. And you don’t really run out of tracks, because you can always copy to another machine. The Beatles had to make decisions. Somebody asked the other day what I thought The Beatles would sound like if they’d had 700 tracks. My thought was that they would probably have sounded terrible, although George Martin would hopefully have stepped in and forced them to make decisions.” POT LUCK RECORDING
Moving on to the subject of working with White, Powell explains that Blunderbuss is one of many albums he has recorded with the singer/ guitarist/producer. Other projects he worked on are all Third Man Records releases, and include mixing The Raconteurs’ Consolers Of The Lonely (2008, it won White, Powell and Joe Chiccarelli a Grammy for best-engineered non-classical album), the James Bond-movie song Another Way To Die (2008, also featuring Alicia Keys), The Dead Weather’s Sea of Cowards (2010), Wanda Jackson’s Party Ain’t Over (2011), and many others, totalling, says Powell, a whopping “100 records” in the three years of the studio’s existence. The beginning of Blunderbuss apparently was the outcome of a happy accident. Sometime in the spring of 2011, White had assembled a group of musicians for a session with RZA (Wu-Tang Clan), who cancelled the morning of. Rather than waste everyone’s time, White decided to use the occasion to run through a few of his songs. Powell: “I had no idea at the time what we were doing. This often happens though. Jack will call me and say, ‘can you be here on Wednesday at 10’ and I’ll get there for nine, make sure everything is set up and get the tape machine aligned and then at 10 I find out what we’re doing. But I don’t need to know what we are doing. Whatever it is, we will make it happen. For the Wanda Jackson session, I knew that there’d be a rhythm section of guitar, bass, drums and piano, but I didn’t know that there would also be a steel guitar,
“
The electric guitar distortion on the solo in Take Me With You…I realised that I was blowing the 1073 up!
”
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FEATURE
THE PHANTOM IN THE OPERA THEATRE Wave-field Synthesis reaches new heights in recreating an orchestra that isn’t really there. Story: Robert Clark
As head of recording and broadcast at one of Australia’s most diverse and iconic performance venues, Tony David Cray is used to a challenge. The Sydney Opera House can have a rock band in one venue, a ballet in another, and an intimate cabaret act in yet another space — all at the same time. When the Grammy Awardwinning engineer/producer was invited to take on the role of sound designer for Opera Australia’s production of Die Tote Stadt, however, he was faced with a task he’d never encountered before: to create the impression of an orchestra emanating from the Opera Theatre’s orchestra pit, despite the actual orchestra being in a completely separate venue. Welcome to opera in the twenty-first century!
was just 23 years old, and looking to show off — he scored it for a huge orchestra. In addition to traditional strings, woodwinds, brass and percussion, he also threw in two harps, a mandolin, celeste, organ, harmonium, piano, two off-stage ensembles and three choirs. The music itself is highly cinematic, which is no coincidence given that Korngold would go on to become one of Hollywood’s most influential film composers. His score, then, presented challenges but also opportunities. It needed a big space to fit everyone, but it also lent itself to a more adventurous treatment of the sound, so Opera Australia leapt at the chance to utilise the famed skills of the Opera House’s sound department and do something radically different.
THE CHALLENGE
After much consideration, it was decided the orchestra and choirs would be housed in The Studio — a spacious, multi-purpose venue on
When Austrian composer Erich Wolfgang Korngold wrote this masterpiece in 1920 he AT 12
the western side of the House. The audio would then be piped live into the Opera Theatre for each performance. Cray was certainly startled when the idea was first put to him: “I was gobsmacked when I heard the decision. In terms of purists and in classical music being performed live, for the opera company to say let’s put the orchestra in another theatre... I could only see predominantly train-crashes on this unless we found something miraculous.” As if by fate, shortly after the bombshell had dropped Tony happened to bump into Ralf Zuleeg, the Head of Education and Application Support at speaker manufacturing company d&b Audiotechnik; who was being given a tour of the Opera Theatre. “I was describing what we were doing with a couple of projects at the time,” said Cray of their meeting, “and I could just see that Ralf had a certain twinkle in the eye, so I thought I’d lay on him this crazy idea of the opera production with no orchestra in the pit. As soon as I mentioned it I could see in his body language that it was exciting for him, and it was from that point that d&b leapt onto it. Then Ralf suggested that I check out wave-field synthesis. We went out to dinner that night, and before we met up I researched some white papers on it, and I just thought: this is perfect, it could be amazing.” WAVE-FIELD SYNTHESIS EXPLAINED
As a mathematical formula, Wave-field Synthesis has been around for a few hundred years, but it has been evolving into a complex algorithm for increasingly practical use since the late-eighties. Basically it’s all about recreating wave-fields to suggest imaginary sound sources, which allows you to replicate spatial circumstances that don’t actually exist. One company that specialises in the advancement and application of wavefield synthesis is IOSONO, a small R&D firm established by the Fraunhofer Institute for Digital Media Technology around 10 years ago. These days it concentrates on streamlining the usability and efficacy of the theory in everyday audio situations via customised software and hardware. Stephan Mauer is the Manager of Application and Product Planning at the German-based company, and he has an elegant analogy to explain what wave-field synthesis achieves. “Just imagine a smooth water surface and you throw in a stone, then you have the circular waves around it, and imagine at the shore of the beach you have a certain pattern of the waves as they approach the shore. And just imagine that at the shore there are loudspeakers and everything you see on the water is purely virtual. And what we do with the loudspeakers is just compute what every loudspeaker should do to maintain this wavefront. So, there is no sound behind the speaker but we think of a virtual sound source and ask what the speakers would have to do in their position in order to recreate the sound field that would have been created by this virtual source. In other words, I don’t have a violin there, but I want to perceive it in that position, so I
need to re-create the wave front that the violin itself would create; that’s what the algorithm knows how to do.” As it turns out, Stephan Mauer and Ralf Zuleeg had already been collaborating on projects using IOSONO’s technology in live venues around Europe. Zuleeg: “This collaboration between IOSONO and d&b started at the end of last year, when we began to discuss whether the wave-field synthesis algorithm could really work in the livesound world. I put one machine in a club where we have nearly everything — rock bands, DJs, whatever — to just get a feeling for the machine and what it can do. The weekend before I flew out to Sydney I also mixed a classical show in Backnang using the unit, and after that I was really looking forward to getting to the Opera House.” With such promising results, Ralf suggested to Tony that they bring Stephan and his IOSONO machine to the Opera House for this project. He was convinced that by using wave-field synthesis, each musician could be virtually positioned in the pit; creating the illusion of a full orchestra playing underneath the stage as per the traditional format. Moreover, by utilising a large speaker array, they had the potential to place any sound source at any point within the Opera Theatre in three dimensions — giving them plenty of scope for immersive effects. It sounded like exactly what Tony and his team were looking for. FROM THE SOURCE TO THE SPACE
The audio team ultimately ended up with a total of 56 mics positioned in and around the orchestra to send to the Opera Theatre via a long, sweat-inducing signal-chain. From mobile racks of preamps situated behind the orchestra, the signal is fed via multimode fibre to the Opera House’s Central Control Room (CCR), which then patches the audio through to a Euphonix System5 console in the House’s recording studio. Gain, dynamics and EQ are then applied before the audio is sent to the IOSONO unit via MADI coaxial cables to work its magic; determining each source’s final destination in the theatre space with pinpoint accuracy via a graphic interface. From the IOSONO box the audio heads out through the CCR to two fader buckets in the Opera Theatre via a conversion to Cat5 fibre, which enables balance engineer Tod Deeley to make adjustments to the levels in real-time in the space — a crucial part of the process. These adjustments are then fed back into the IOSONO and out again into the theatre’s MADI to analogue AES before ending up at the d&b amps and a PA system comprising 97 individual speakers. As if that isn’t complicated enough, each of the singers on stage wear a microphone that is fed back to the orchestra and conductor, but their voices are not amplified into the audience because it was strongly felt that would compromise the acoustic qualities of the music. The ‘bottleneck’ of the system, as Stephan Mauer describes it, is really the speakers: “We can
The Battle Plan: where to put which mic (top), Conductor Christian Badea’s sonic mud-map (bottom).
START WITH THE SOURCE One of the first people to give a strong opinion on how the system should work in the space was the conductor, Christian Badea. From the outset, the Romanian maestro was very keen to accommodate the technology and maximise its effectiveness. In fact, he hand-drew a layout of the theatre with detailed instructions as to where he wanted various sounds to be placed in the theatre. With this information, Cray began to consider how to ensure the IOSONO machine received clear, precise signals in order to have maximum flexibility within the space: “I knew that the algorithm itself would work better with certain sources. If you want to tell the IOSONO box there’s a point source here or there, you don’t want the instruments around it to be bleeding into that. So you want to get a lot closer than normal, and to try and make up for the fact that we’re so close and the sound can get quite harsh, we choose a mic that’s cardioid and that will have a profile that’s a bit more gentle in the higher frequencies. It’s kind of like we’re zooming in but it’s softer.” In order to maximise isolation even further, mixer Jason Blackwell mic’d up the woodwinds using an overhead figure-eight pattern. Tony explains: “a figure-eight pattern is when you have two cardioids above each other, and that means around them, almost like Saturn’s rings, there’s a null-point. This will cancel out the brass and percussion, which is usually something that you’re always fighting. It’s common that you find you don’t have enough woodwinds but you’ve got too much brass because they’re just cutting straight into it.” AT 13
FEATURE
THE DOWN LOW The Mars Volta bassist Juan Alderete de la Peña talks about how far the bass instrument can stretch and what gear effects his sound. Interview: Mark Davie
While a lot of modern records seem to be asking how low can a bass go… in the mix, Juan Alderete de la Peña has different ideas for his favourite instrument. As well as being one of the most solid, and at the same time, experimental bass players around, Alderete has a thing for effects. He plays with The Mars Volta, and members of it in the Omar Rodriguez Lopez Group and his experimental outfit, Vato Negro. He, drummer Deantoni Parks, and ringleader/ guitarist Omar Rodriguez-Lopez have to be one of the most expressive and talented group of musicians going around — not to mention prolific. Big Sir, on the other hand is Alderete’s duo with vocalist/instrumentalist Lisa Papineau, the sculpted, fretless antidote to Vato Negro’s often ballsy, eccentric improvisation. People following Alderete’s career would have heard his bass produce sounds anywhere from beautifully, lyrical vibrato modulations on his fretless, to subby helicopter whooshes that push his Ampeg SVT to its limits. He’s a master of expression, and creativity on an instrument that has typically been given the supporting role. To spread a bit of his wild experimentation with the world, Alderete recently started pedalsandeffects.com. Sick of the stodgy, straight up pedal demos on YouTube, Alderete has decided to populate the online video site with his own off-kilter renditions. Mark Davie: What inspired you to start Pedals & Effects? Juan Alderete de la Peña: You can pretty much learn anything on YouTube. I have a little nephew who wanted to learn how to do origami and he went to YouTube, next thing he’s making origami. I was frustrated with a lot of the music-related instructional videos, especially when it comes to pedals. You always find it’s guitar-dominated people, who are studio guys looking for creamy, subtle nuances from their guitar pedals. Whereas I don’t really believe that it makes that big a difference when you’re in a rock band. Everything in a rock band has to be big, really heavy on the effect so you can hear it through the PA at a rock venue or festival. Pedals & Effects lets you hear how insane a pedal can go. I’m going to play guitar and keyboard ideas but my instrument is bass so I’m starting with that. But if you’re looking for an over the top fuzz, I’m going to cover some of them. If you’re looking for a really grindy distortion, I’m going to cover a lot of them. If you’re looking for how to make your bass sound like a synthesizer, I’m going to cover that. I’m also going to talk about pick-ups, strings, instruments, and all the tools for the musician and then have a forum where people can have a conversation about it. And if it’s all in one area the community becomes a lot stronger. MD: So it’s not going to be how to achieve Mars Volta or Vato Negro sounds? JA: No. For a recent installment I thought it would be interesting to record a great bass player friend of mine and myself trying to get sounds on a new pedal for the first time. It’s the person, the artist, the individual that has an impact on it, and we ended
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up getting radically different results out of that pedal when we hadn’t heard each other use it. LENDING A SOUND
MD: So how many pedals would you say that you have? JA: In storage or in my studio? Pedals are like books. You loan them out all the time figuring that there’s a good chance you may never see them again. I’d have to say over the last 20 years I’ve gone through 500 or 600 pedals. I probably have about 300 in my studio, another 100 in storage, and probably another 100 out and about with friends. And I probably have 100 that I’ll never see again. MD: Are you a bit of a snob with pedals? Do you chase particular germanium fuzzes or some specific edition of a tank-built Sovtek fuzz? JA: I am and I’m not. I really believe that any musical person can take the shittiest or the best pedal and have it work for them for something. You’ll hear something that you can use. Though I’m not one for subtle pedals. There’s this guy out here in California that makes Full Tone pedals. I’ve never been a fan of them. They’re for guitar players who are going for a ZZ Top or Tom Petty subtle sound. I have no interest in that. I go for weirder builders that are just in their garage turning out over-the-top pedals.
my Boss OC-2 Octave pedal, which goes to my Dwarfcraft Devices Baby Thundaa fuzz, and that goes up to my Digitech PDS 20/20 Sampler/Delay and then that goes into the Boss TR-2 Tremolo, with the LFO on it, and then that goes into the Line 6 DL4, and that goes into the Earthquake Devices Rainbow Machine and then that goes to my DI. Underneath the pedal board is an old DOD MOSFET preamp that just gives me a low end boost. And I forgot one of the most destructive pedals. I knew I was missing one, the DOD Meatbox. That’s the one that used to be $40-$60 on the internet and then I started doing a lot of interviews where I was saying the DOD Meatbox blows up P.A. systems and the price went up to $300-$400. MD: You kind of screwed yourself there. JA: Yeah, my bad. I REMEMBER IN PARTICULAR THE 2010 BIG DAY OUT WHERE THE GUYS IN MUSE WERE
LIKE, ‘WHAT THE HELL IS THAT PEDAL? EVERYBODY IN THE AUDIENCE GRABBED THEIR CHEST WHEN YOU HIT IT.’
That’s the DOD Meatbox! That thing is fierce. MD: Are you ever concerned about the transition from studio to stage? If you’re being experimental in the studio and set up a string of five pedals and then go to another section and you might need to turn three off and another two on with only two feet?
But also I have my favourite series of the Sovtek. In Russia they were making them out of whatever components they had access to. So you’ll find from one to another in the same series that they’ll have different parts, capacitors, and sound totally different. So if I buy the first series green I’m not guaranteed to get that sound. I’ve found several of them and they don’t sound the same. I’ve owned ‘Civil War’ versions that are very expensive on the internet, and I hated mine. You’ve literally got to hear them and spend time with them. Modern pedals are more consistent.
JA: Never. There’s always a way. Especially with a great pedal like the Line6 M9 pedal. If I really start getting that crazy, playing all the songs from the old sets, I would take that pedal and just program everything. It’s not going to sound exactly the same, but my pedals in the studio aren’t going to sound the same in a venue that sucks or through a PA that sucks. You’ve got to give in a little to the odds and say, ‘I hope this all goes well and I get an 80% across this entire tour, and run into more good venues than bad venues’.
I buy backups even three to four deep of almost every pedal I feel is indispensable, just because I know they’re going to break. I have two Boss VB-2 Vibrato pedals, that’s about $800 of pedals right there!
MD: Do you mod your pedals?
SLIMMING DOWN THE RIG
MD: You can’t take 300 pedals on tour, so what do you take? JA: In the old days I used to surround myself with four pedal boards that were four feet long. We would ship our gear to Australia and I’d ship two Ampeg SVT fridges, three SVT-VR heads, six basses, four giant pedal boards and then a bunch of spare gear. Then we’d come home and never make any money because we’d spent $70,000 on freight. These days my 12 pedals fit in a Pelican case, it’s different times. MD: What are those 12 pedals at the moment? JA: One’s a TC Electronic tuner, and then in this order: Boss CS-2 compressor/sustainer, into the Boss VB-2 Vibrato which goes over to the ElectroHarmonix Ravish Sitar pedal which goes into the Earthquake Devices Organizer, then down into
MOD FACTOR
JA: I have some. Like the Baby Thundaa that Dwarfcraft Devices made me. He usually has more knobs and switches, which was just too difficult, especially on festivals, my tech’s not going to have enough time to flip all the switches. I took a photo of the setting I was using and I said, ‘Can you just make it do that?’ And it came back with just two knobs and a switch. Matt from Wren & Cuff makes me custom fuzzes all the time. Same with William at WMD, he makes the Geiger Counter. He was trying to make me a sub pedal that I really love.
“
You can sit there and work all day on your heavy guitar tones, but if you throw a bass through distortion you’re going to kill people
”
MD: People always seem to mod those TR-2s as well for a volume drop. JA: Whatever I use I hit it so hard that there’s no way I’d notice the volume drop. I hit it with a fuzz right before, and my compressor’s on. I want to have a lot of attack at the top end so it sounds choppy. Still to this day no tremolo will sound as good as that one. And I own a ton of them. MD: Can you talk a bit about your rig and basses?
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REVIEW
AMPED ABOUT
REAMPING
Michael Carpenter looks beyond his amp simulators, and with four little reamp devices discovers a depth of tones and gear he never realised he was missing. Review: Michael Carpenter
I have a confession to make, and it’s probably not going to help my credibility as a guitarist or a ‘traditional rock ‘n’ roll/country’ record producer. We have lots of great guitar amps, big and small, and lots of guitar pedals, normal and strange, and I hardly ever use these tools to make records. There. I’ve said it. Sorry. I love guitars and have a healthy arsenal of weird and wonderful guitars. And I generally make guitar-driven records. But over the last few years, I’ve willingly moved over to the world of virtual amplifiers. I have a couple of external floorboards and a whole bunch of awesome amp simulators in my DAW. Not only that, I’ve carefully built a killer library of reliable starting point ‘presets’ that help make sessions, and therefore ‘creativity’ flow. And – it’s not just that it’s convenient. I pull really great guitar sounds out of my chosen tools. Meanwhile, my trusty and reliable guitar amps and pedals are demoted to ‘live use only’. It’s not always like this, but I’d say they only get used 10% of the time. I’m going to ask a quick question. How many of you recording dudes, either in big fancy studios or in your smaller bedroom ones, have a process for reamping your DI’d guitar signal? Or even worse, went to the effort of buying a reamp device but rarely use it? Yeah... that’s what I thought. When presented with the idea for writing this article, I actually had a brief moment of ‘shame’. But, I also saw this as an opportunity. Can I change my ways and get my beloved AC30 onto records? THAT CHEAP AND NASTY DELAY THAT HAS A SOUND LIKE NOTHING ELSE — CAN IT BUMP OFF THE PLETHORA OF PLUG-IN DELAYS? I WAS KEEN TO FIND OUT.
104 WAYS TO REAMP A STRAT
Now, I have tried to reamp things in the past. I’ve made a few records where the guitar player has only wanted to record a single guitar track per song. They brought in a whole menagerie of AT 16
amps though, and wanted, once they’d recorded the guitar take with a ‘working’ sound, to very carefully layer guitar tones throughout every song using different amp/pedal combinations. Of course, I didn’t have a reamp box, so I did the horribly crude thing of taking a line from the low impedance output of the DAW, turned it down to a level that the amp would accept and used that. The tone was what I’d call ‘acceptable’, but it was certainly good enough for our purposes. This is of course a silly way of doing it, and I subsequently learned that I was tonally shooting myself in the foot. There are very basic adaptors that you can get from your local electronics store that do the job, but I’ve always felt they change the tone too much. So we’re going to have a brief look at four purpose built products that do the job for you. They all have slightly different features, and will appeal to different people based on these features. So, I’ll try to outline how I used them and what their strengths and weaknesses, if any, were. BROADCAST PRO AUDIO PASSIVE LINK
The first cab off the rank is the Broadcast Pro Audio Passive Link, built by Australia’s own Rob Squire. This is a simple, effective, easy to understand box, in a distinctive green colour. The first thing I loved was that it has both male and female XLR inputs wired in parallel, making for a simple interface to your recorder outputs or patch bay. Sometimes it’s the little things, and I loved this about it — it meant we were up and running with it really quickly. It’s a passive unit, so no power is needed, which is really handy too. On the top of the box you have a volume and tone control. The 12 o’clock position is unity, so you effectively have a volume and tone cut and boost. These sound great, and are effective in tailoring the direct sound out of your recorder back into your amp, making it easier to find the sweet spot of your guitar/amp tone. On the bottom
side of the box, you have two reamp outputs to run multiple amps (very cool) and the essential ground lift to take care of hums, which it did well.
added a subtly robust midrange, which I liked for the part I was recording.
My first task for the day was to reamp a bass guitar. The DI track sounded good, and I would normally run an amp simulation plug in parallel to introduce the right amount of ‘amp tone’. But for something different, I decided to run the bass into a toy, battery powered Marshall amp. My aim was to get a track of maximum ‘punky grit’ to blend in with the DI. I set up a pre fader send to an output from the DAW, set the volume of the send in the DAW to 0.0, and plugged the Passive Link into this monster rig. I got an effective sound quickly, recorded a bit to a new track, and then, to test the integrity of the signal, quickly plugged the bass into the little amp, and recorded a bit of this sound, to compare. The sound was really close to being identical when flicking between them. I was impressed — a great, simple start.
Moving on to the active units, the X-Amp Studio Reamper, gives the appearance of being a more substantial, versatile box. The first thing you notice is the isolated second output with polarity switch, designed to run multiple amps cleanly. I was very keen to put this into action, so the next guitar track to reamp would have the AC30 running, as well as our little ‘Mini Fender Tweed’. The tones were obviously extremely different, but I had an idea that the combination would create something unique. Again, setup was simple. There’s a screwdriver operated output level control (though you can move it with your fingers). I’ve no idea why there isn’t a knob on this like on the JCR Studio Reamper, but, while not a big issue, it did reduce the tactile appeal of a volume knob. However, there are no tone controls, which I missed. As with the Broadcast ‘Passive Link’, the multiple outs create more possibilities, especially with the isolated second channel, helping to clean up the tones between outputs, as well as the phase switch allowing you to get the best out of the tones of multiple amps.
RADIAL JCR STUDIO REAMPER
The first thing you notice when pulling the Radial JCR Reamper out of its packaging is that it ‘feels’ very solid in hand. The thick metal casing, firm switches and knobs, its weight, and the full length rubber base, to avoid slipping and scratching the surface its placed on, give this a trustworthy, reliable aesthetic. It too is a passive unit, which again makes for easy setup. It has a male XLR input though, which was only a problem because I’d grabbed a female XLR input when testing the Passive Link. It also has a ¼-inch TRS jack input, which does give you options when connecting to your patchbays. The first feature I loved was that the unit has a ‘mute’ button. As I’d moved away from tiny amps and was now plugging into the louder Vox AC30, the ability to mute the signal being sent to the amp was extremely useful. Rather than having a tone knob, the unit had a switch that selected either a high-pass filter, low-pass filter or a flat EQ. This proved to be handy when cleaning up the gruntier tone I’d recorded on this next track. On the Jazzmaster track that followed, the lowpass filter beautifully tailored the edgy bite of the signal. So the tone switches, while not technically as versatile as a tone knob, have been designed to do a particular job, and they do it well. Falling into the ‘little things make a difference department’, I noticed the output level knob on the bottom side of the unit feels great to handle. The resistance when turning it was excellent, and felt very deliberate when driving into the amp to find the sweet spot. My only criticism of the JCR Studio Reamper is that the ground and phase switches are sunken into the box, so they can only be switched using a pen or something similar. I looked in the manual to see if there was a reason for it, but there was no mention of it. It just felt a little silly. Regardless, this box did the job well, and once again, when comparing the reamped and re-recorded signal, the tones were very similar. If anything, the JCR
RADIAL X-AMP STUDIO REAMPER
In use though, I found this one a tiny bit thinner in tone to the others. It was a little harder work getting the tones I needed. I thought it may have been the DI’d guitar tone, so I switched to one of the other guitar tracks and recorded a section. The tone was subtly not as beefy. A slight tweak of the output volume did help this, but I couldn’t help but wonder whether the tone control that was missing was playing a more significant part in the reamping process than I’d expected. Regardless, this unit made the process of cleanly setting up multiple amps a breeze, and would be a great unit if you know this is something you would do a lot.
“
It allowed me to access a box full of effects and use them as ‘free plug-ins’
”
RADIAL EXTC EFFECTS REAMPER
When I first looked at this unit, I wasn’t sure what it was. Reading through the manual, I started to understand that it was designed to make it effortless to plug in your high impedance guitar effect pedals directly into your recording setup. Once I understood this concept, I became really excited and curious of the possibilities. Boxes of pedals discarded from my colourful history of pedal boards could have new life! Thinking about it, essentially it allowed me to access a box full of effects and use them as ‘free plug-ins’. For someone interested in the more rustic idea of using effects that are grittier than your standard plug in, this was an exciting prospect. The EXTC Effects Reamper is, once you got your head around what the inputs and outputs mean, quite easy to use and set up. It allows you to run two separate effects loops, dial in the send and receive level, and, most importantly, adjust the blend control between the effect and the dry send. Essentially, it lets your external pedal effects act like a dedicated outboard unit. The
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REVIEW
NO STRINGS ATTACHED A look at the latest in wireless guitar systems that will get you attached to being detached. Review: Mark Woods
Freedom is the big one, that’s what everyone wants and that’s what the manufacturers of wireless systems for guitars are selling. Freedom from leads that tie guitarists to their pedals and amps. Freedom to run around the stage without creating, or tripping over, a tangled mess. Freedom to break the barrier between stage and audience and leap into the crowd shredding like crazy. Even the freedom to appear at the FOH mixing position playing out of time and making comments on how loud the guitar should be in the mix….although we don’t encourage that. If all this freedom sounds appealing you’ll find it’s a good time to be looking into wireless guitar systems; prices are lower than ever before, competition has delivered more comprehensive functionality combined with simpler operation, and the latest digital models can offer quality indistinguishable from wired devices. Before we start exploring some of the latest models it’s worth mentioning that there is some uncertainty about changes to government policy regarding available transmission frequencies and the switch to digital TV. Wireless audio equipment currently shares spectrum with TV across the 520-820MHz range but after the change to digital TV is complete the 694-820MHz range will be unavailable. It is expected that wireless audio devices and TV will still share spectrum but in the smaller range of 520-694MHz. There may be problems for older equipment if they are operating in the 694-820MHz range but the latest wireless devices have been designed to operate in either the 520-694MHz range, or above 900MHz, and be sufficiently ‘frequency-agile’ to find clear channels in a crowded spectrum, so hopefully there won’t be any unplanned obsolescence surprises (check out Cutting The Cord On Wireless Audio in Issue 88 for more info). AT 18
FREQUENCY OF SELECTION
Finding and selecting frequencies has always been the most difficult and frustrating part of using wireless devices. I had example of this recently when the prickly singer of a once well-known band I used to mix came to perform at the Theatre Royal in Castlemaine. Despite him performing solo with an acoustic guitar he was using in-ears and at soundcheck I could see I was sending signal to his transmitter but he wasn’t getting any sound. We had a bit of the old ‘I’m not getting it’… ‘well I’m sending it’…back and forth until his helper matched his belt-pack receiver’s frequency to his transmitter. It worked then but we both could have done without the aggravation. The latest wireless devices simplify this by using a scan process that automatically finds the clearest available frequency at the touch of a button. Once that frequency has been identified the transmitter is held close to the receiver and the ‘sync’ button is pushed. A clever infrared (or Wi-Fi) system allows the transmitter to sync to the receiver and away you go. If multiple wireless units are required the first receiver is set up, synced to the transmitter and left on. The next receiver is activated and finds the next available frequency, sync to transmitter and its running. And so on. The number of devices that can be operated simultaneously varies but even the entry-level models will deal with up to around 10 devices and that’s enough for most events. I’ve been listening to a range of newly released wireless guitar systems and have generally been impressed by their sound quality, features, ease-of-use and price.
SAMSON AIRLINE SYNTH AR300 RECEIVER/AG300 TRANSMITTER Price: $649
The Samson AirLine Synth is an analogue system available at a lowish price that offers much the same features as higher priced models, and employs a unique approach to the transmitter design. The AR300 UHF wireless receiver is a sturdy all-metal half-rack unit with removable protective strips wrapping around the top, sides and bottom for safe placement on or around the stage. A kit is supplied to enable rack mounting after the protective strips have been removed. On the front there’s a fairly hard-to-read 50 x 20mm backlit LCD screen between pots for Squelch, Level and Edit. A good feature is the way the two removable antennas can be attached to the front or the rear of the unit. If the unit is rack mounted it’s probably better to have them on the front to get them in clean air and for easy adjustment but if the receiver is sitting on an amp or hanging around the stage then they might be more out-ofthe-way mounted on the rear. Another handy option is the AR4300 chassis that enables four units to be housed in a single rack space. It includes antenna distribution and a headphone monitor system that can listen to any one of the four receivers or all of them. The Samson AirLine receiver is a frequency-agile, auto channel selecting design that provides for over 300 channels. Infrared scanning syncs the transmitter to the receiver and up to 11 systems can be used simultaneously. Signal output is via balanced XLR or unbalanced ¼-inch sockets located on the rear panel. The AG300 transmitter is very different to the usual belt-pack design in that it plugs directly into the output socket of the guitar. Powered by a single AA battery the transmitter has a dual-mount jack plug that works on either flat-mounted sockets (Gibson style) or, by unscrewing the jack plug and placing it in the other receptacle, edge-mounted sockets (Fender style). It’s a clever idea that eliminates the need to find somewhere comfortable, and out of sight, for the belt-pack and lead from the guitar output. Controls on the transmitter include power on (with simple battery indicator) and a mute
switch for killing the sound or hot-swapping the transmitter between guitars. There’s a small backlit LCD screen at one end that’s even harder to see than the one on the receiver but it only displays the channel number and it’s not essential to see it easily. Under the battery cover there’s a 15dB pad switch and a rotary gain control that doesn’t usually need be changed unless, for example, the guitar has active pickups. RF output power is quoted at 10mW ERP. Battery life is quoted at 10 hours and that’s very good from a single AA battery. The Samson system is easy enough to use once you’ve got the frequencies synced. I couldn’t do it at first; I kept reading the instructions and I was doing everything correctly but no sync. Then it worked but it wasn’t obvious what I had done to make it happen. After some experimenting it seemed the transmitter had to be quite close to the receiver and in a certain position…and has to be switched on at the right time. On repeated attempts it became easier so I guess it’s just a quirk you get used to. Once I had sound I began to notice how well the transmitter works. It’s so light it doesn’t change the feel of the guitar and it doesn’t get in the way when playing. The feeling of freedom you get going wireless is enhanced by not having to attach a belt-pack or deal with the connecting lead. Swapping guitars is especially easy, the mute switch falls readily to hand and its smoother changing the transmitter directly to another guitar compared to finding and inserting a connecting lead. Of course changing from some guitars, a Fender to a Gibson for example, means you have to unscrew and move the input plug but it still only takes a couple of seconds. The sound quality is good, the system seemed to introduce a light compression to the signal and the very high frequencies were slightly attenuated compared to a direct connection but overall the quality is professional and perfectly acceptable. The pre-amp in the transmitter has been designed for guitar pickups with an input impedance of 290kΩ. The operating range seems unaffected by having only one AA battery (not supplied) for power and with the receiver inside the studio I was able to go outside and about 40m away before it lost strength…and that was around several corners, so the claimed 100m line-ofsight transmission might be right. While its range
is powerful for a bug system, the limited 4MHz bandwidth (642-646MHz) may not be wide enough in some areas.
SENNHEISER XS WIRELESS RECEIVER Price: $529
Sennheiser has a long history making highquality wireless systems and their EW100 G3 range is regularly found performing well in demanding professional applications. The new XS Wireless system is around half the price but it’s still a quality unit that’s well made and easy to use. The main differences are less RF power, less bandwidth and a slightly worse signal to noise ratio. The analogue EM10 receiver is metal rather than plastic and the clean front panel features a fairly easy-to-read backlit LCD screen, up and down nudge buttons, a ‘set’ and a ‘sync’ button and a volume control. The chunky antennas attach at the rear as are the expected balanced XLR/unbalanced ¼-inch jack and the squelch pot. No rack mounting kit is supplied but there are screw holes in the top and on both sides to accommodate the range of optional mounting accessories. The unit ships with a universal power supply and international adaptors. Frequency selection is simple with either manual tuning, auto scan or presets. There are five frequency ranges covering frequencies between 548MHz and 865MHz. The unit I tried was set to 614-638MHz. There are 960 possible frequencies per range and up to 12 units can be used in each frequency range. The 2.4GHz Wi-Fi sync feature works by holding down the sync button on the transmitter and pressing the sync button on the receiver. It worked first time every time. The SK20 belt-pack is a traditional belt-pack that’s familiar and logically laid out. It’s made of plastic but feels strong. The antenna is fixed in place and there is a small backlit LED screen on the front. The left side has a recessed power switch and sync button while on the right side there’s a 4-position input sensitivity switch. On the top there’s a large and easy to use mute button and indicator lights for mute and power. On the back are a strong belt clip and the battery compartment that takes two AA batteries for approximately 10 hours use. RF output power is AT 19
REVIEW
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QUICK CONTROL SURFACE TOUR
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YAMAHA CL 5
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Channel Faders — 24 input faders, name, on, cue, input meter, gain/pan/assign knob
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Selected Channel section — just like an input channel from an analogue console
3
Touch Screen Display — the heart of the machine with many layers of parameters
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Centralogic section — eight faders like the inputs but selectable to be any input or output or master
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Scene Memory & Monitoring section
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User Defined Keys — customisable, mute groups, tap delay, etc
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Master Output Faders
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Output Meters
User Defined Knobs — Customisable, effects parameters, etc
USB Connector — for recording and playback from USB key or recall/save show files iPad bay for wireless remote control
DIGITAL LIVE SOUND MIXING SYSTEM Yamaha’s CL series sounds better than any digital Yamaha before it, and is let off the leash by Australia’s own Dante networking. Power user Trevor Cronin gets into the nitty gritty details that make the CL a huge step forward. Review: Trevor Cronin
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Over 25 years ago Yamaha became one of the pioneers of digital audio, with the DMP7 being its first digital mixer. It has since offered the pro audio world a vast array of popular digital mixing consoles. The Yamaha PM5D large format mixing console has been an industry standard for many years along with the smaller M7CL, LS9 and 01V live mixers and the O2R and DM series studio consoles. The all new CL series fits in somewhere between the PM5D and the M7CL, and depending on what configuration is purchased, it can far exceed the capability and functionality of both machines. THE PACKAGE DEAL
This is a complete audio interface and mixing system that allows for the full range of audiorelated input, control, manipulation, storage, output and distribution of audio signals required for current live sound systems. It’s a very powerful package, combining technologies from many of the leading players in the professional audio world with Yamaha’s own wealth of leading technologies and manufacturing. New for Yamaha, the system uses the Australiandeveloped Dante digital audio network as standard for passing audio in and out of the system, with remote stage boxes and other devices connected via Cat5 cables. It’s a big move to discard their commitment to using their own ‘Ethersound’ system on this new product. Obviously they figure that Dante is the best system currently available. The new console sticks with the existing Centralogic user interface with a touch screen not dissimilar to the M7CL model, featuring a bigger and better viewing area and much faster operation. The fader name display is now large and fully customisable with colours (a little like a Midas), and feature an encoder for gain, pan, etc. Continuing with the existing Yamaha standard, three mini YGDAI card bays are included for connection of a large range of interfaces. Yamaha and Lab.gruppen have teamed up to allow Lake Processing (another Australian developed technology) on the new MY8-LAKE expansion card. The card provides 8-in/8-out Mesa system EQ, 4-in/12-out Contour speaker crossover, and 4-in/4-out and 2-in/6-out Mesa and Contour combination modes. The Mesa EQ can also be used on input sources. The Lake Controller application running on a computer allows full access and management of Lake, and compatibility with Smaart Live analysing and speaker tuning software. There is also a new Dugan Designs automatic microphone mixer card (MY-16). Check out the Yamaha website for available cards. www.yamahaproaudio.com ON STAGE WITH RIO
The CL relies mainly on Rio external stage boxes for its connections. These are all new items with newly designed (and better sounding) digitally controlled microphone preamplifiers that have a handy new gain sharing feature that allows two or more consoles to adjust the head amp gain without affecting each other’s settings — so no more arguments between front of house, monitors and broadcast engineers.
The smaller format CL3 with optional meter bridge
Two options are available: the 5U size Rio3224D provides 32 ins, 16 outs, and four AES/EBU digital outputs, and the 3U size Rio1608D has 16 ins and 8 outs but no AES. Up to eight units can be connected to the console. Via console control you can flash LEDs on the boxes to show which box or channel needs to be patched. No doubt there will be some new products to supplement these two models in the future with different input and output options. Being Dante you can use the Rio boxes with any other Dante system, or Ethernet enabled computer. Add Dante cards to your PM5D and use the Rio units, or hook up a computer to a Rio3224D for a very small and practical 32-in/16-out recording system. CENTRALOGICAL IN USE
At this stage there are three consoles available — CL1, CL3 and CL5. I managed to get my hands on the largest model, the CL5, at a trade show in dusty Dubai for many hours over two days, then more recently at the CL console seminar. Although I didn’t mix a real live show I did get to know the machine by mixing a variety of prerecorded live shows via the Nuendo multitrack input and by testing out some microphones. The CL5 package can have a maximum of 72 mono inputs and 40 outputs, and features 34 faders. For monitor mixing duties you can configure 32 mixes (24 mix and 8 matrix busses) all with output EQ. The console interface will feel comfortable to users of other Yamaha mixers due to the continued use of the Centralogic interface, which is also where the console gets its name — CL = Central Logic. For users new to Yamaha, it’s laid out in a way that should offer little difficulty getting your head around how it all works. There are enough knobs, buttons, faders and meters to use before needing to touch the screen. Build quality is superb, using a strong onepiece aluminium extrusion, which is quite an impressive piece of engineering. Internally the system uses the Yamaha operating system and features all new DSP along with
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Overall it is a big step forward in digital live sound mixing
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new faster and higher capacity electronics. This makes the CL more powerful and energy efficient compared to previous models. There are plenty of great new features to make the life of live sound engineers just that bit easier. LIVE RECORDING
Record and playback from a USB key is a great Yamaha feature that has again been included in this unit. It is only in ‘lo-fi’ MP3 (or WMA/ ACC) format, so fine for the intro music and a quick recording for reference. For higher quality recordings a PC or Mac computer with Dante network drivers can run the included Steinberg Nuendo ‘Live’ multi-track hard disk system for show recording and virtual sound check/general playback. No extra interface is required as Dante uses the computer’s high-speed network port for bidirectional transfer. More than one computer can be used at the same time — such a great feature for critical live recordings or playback. The Integration of Steinberg’s Nuendo using Dante is a fantastic feature, it allows for state-of-the-art recordings for very little extra cost (making mobile recording companies redundant?). Steinberg is now owned by Yamaha and the console features a slightly stripped down version of Nuendo that can handle 64 inputs and 64 outputs. Tracks are automatically named from the console inputs and there is a 60-second AT 21
REVIEW
HPI 110 Physically the HPI 110 is a good-looking speaker; it’s tall and tower-like with distinctive ports on the bottom corners. The 620x380x346mm cabinet is made from reinforced birch ply and I like the Quest logo on the front, it’s simple, clean and easy to recognise.
TOUGH WHEN GRILLED For drive it uses a 10-inch neodymium woofer and a 1.4-inch neodymium compression driver mounted on a rotatable asymmetric flare. A strong perforated steel grille covers the front then wraps around the sides of the cabinet and combines with foam under the grille to offer excellent protection from unwanted solid or liquid attacks.
QUEST ENGINEERING HPI 110 & HPI 8i SPEAKERS Aussie brand Quest Engineering has a new range of passive speakers designed and assembled in suburban Melbourne that are serious for their size. Review: Mark Woods AT 22
HPI 8i
Quest Engineering is an Australian pro speaker and amplifier company that has traditionally been more of an export success than an Aussie mainstay. Until recently, Quest has sent its designs overseas for manufacturing, but Quest’s latest HPI series is designed, built, and tested in suburban Melbourne. Quest has embraced the homegrown ethos to the extent of using Lorantz drivers in its HPI subs that are manufactured from Eucalyptus pulp. These days, Quest is bringing ‘overseas’ here. French electro-acoustic engineer ‘G’ has come onboard for the new and upcoming ranges, spending his time in and out of the anechoic cubbyhole at the back of the Group Technologies warehouse (Quest distributor). Quest has invested over half a million dollars in R&D, and a pair of 2-way passive speakers, the HPI 110 and HPI 8i are among the first trickles of what is shaping up to be a wave of all-new product. They are part of Quest’s new series of compact, full-range speakers targeted at portable live sound or installations where high sound quality is required.
THE ULTIMATE MONITORING HEADPHONE
ASYMETRICAL HORN
The HPI 110 can be used as stand-alone FOH or playback speakers, or multiple units can be used in a vertical array. They look like no other speakers I know and when they’re mounted on a stand and viewed from the three-quarter rear position they remind me of the creatures from the Alien movies. There’s a polemount hole on the bottom and at 16.5kg they’re fairly easy to get on a stand, but they are back heavy and it wouldn’t take much of a knock to make them topple over backwards…better to fall back than forward I suppose. Recommended amplifier power is 480–720W for a maximum of 130dB SPL and the frequency response is quoted at 70Hz–18kHz ±3dB. It’s always a good sign when speaker manufacturers provide a frequency response graph. Quest has, and it confirms the commendably flat response, an achievement more noteworthy in an unprocessed speaker. Designed for mid-throw application with an emphasis on keeping the sound focused on the audience, the horn coverage ranges from 15 degrees up and 45 degrees down in the vertical plane. In the horizontal plane the asymmetrical flare design throws out at 30 degrees from the top of the flare and 45 degrees from the bottom. This effectively distributes the sound to a wider area close to the front of the cabinet while squeezing the sound out of a narrower opening on the top of the flare to give a longer throw. QUALITY VOICE
In use, the HPI 110 speakers are straightforward and capable. They don’t require a processor and they will work well with any amp of suitable power. The voicing is very good with the mids forward, but an even frequency response without the hollowed-out quality of most DSP controlled, artificially voiced powered speakers. Speaking into the HPI 110 with an SM58 reveals a fast response, in-your-face sound with presence and plenty of mid-high detail. Up close and loud they’re very resistant to feedback but will eventually get glassy around 2–4kHz to let you know they will squeal if pushed further. It’s once you start to move away from the speakers that you begin to notice the throw and how strong they sound at what seems like a long way for a smallish speaker. Indoors the low frequency response is good down to around 100Hz but drops off below that. Speaking into it right on the microphone lets the proximity effect fill out the low end nicely without boominess and straight out of the box these speakers would be ideal for any application that required loud, articulate vocals. For music playback they benefit from some added low-frequency EQ, if they’re not used with a sub, but this only works at low-medium levels. If the application requires medium or high levels of music playback then a sub will be needed. For live music FOH duties they would also normally be used with subs and there are three models available in the HPI range. The HPI 12S uses a single 12-inch woofer and is recommended for small venues, next up is the HPI 212S that has two 12-inch woofers and finally there’s the HPI 18LP with a single 18-inch speaker that will reach down to 38Hz. Interestingly, the woofers in the sub range are produced by Lorantz in Melbourne and as previously mentioned, the cones are made from specially selected paper pulp sourced from Eucalyptus trees.
MONITORING MADE EASY The HD380 Pro headphones are ideal for professional monitoring. They have a closed capsule design and feature Sennheiser’s “Eargonomic Acoustic Refinement” (E.A.R) technology. E.A.R. channels the audio signal directly into your ear while the closed capsule design provides excellent passive noise reduction. The HD380 Pro also offers an extended frequency response, with an increased sound pressure level for accurate sound reproduction in any situation. The lightweight and secure-fit design offers a comfortable listening experience over long listening sessions.
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LITTLE BROTHER
The HPI 8i is like a little brother to the 110 and shares its good looks. It features an 8-inch neodymium woofer and a 1-inch exit ferrite compression driver, again on a rotatable asymmetrical flare that throws both short and AT 23
REVIEW
CHANNEL CONTROLS Like most digital live mixers, the DP-24 has one set of dedicated EQ, pan, and send controls that function on the selected channels. However, there’s no dedicated button for phase, to insert an effect, or to access digital trim. These are all onscreen via the jog wheel.
FADER PLAY The 19 short-throw faders, including a single stereo master fader, have a decently resistant feel, are solid, and not difficult to play with.
TASCAM PORTASTUDIO DP-24 Tascam is still hanging on to the Portastudio concept amidst increasing pressure from DAWs, so does it still have a place? Review: Mark Davie
The Portastudio feels like it should be extinct by now. A standalone, all-in-one recorder/mixer seems unnecessarily limiting in an age of laptops and eight-channel 1U interfaces. Especially amongst the more recent control surface/interface combinations that have come to market. The Portastudio’s unwillingness to relent is interesting, so we put the latest Tascam DP-24 through its paces to see where it fits in the current landscape. Like JoeCo’s Blackbox for live recording, and location recorders from Nagra and others, there are plenty of digital recording devices out there that don’t need to speak to a computer to work, and are better for it. The difference is that the Portastudio is trying to cut ties with computers altogether. It wants to capture your song, take you through the mix process, master it and burn it to CD. I’m surprised there’s not a label maker installed! It even refuses to look like a computer, with no way to hook up an external monitor, and the USB connection won’t let the unit act as a soundcard, it’s only function is to turn the DP-24 into a rather cumbersome card reader.
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STUDIO ON THE GO
Out of the box, the unit is set to 16-bit/44.1k, presumably to play nicely with CD masters. Though you can set up 24-bit/48k when you begin a new song, recording to SD or SDHC cards. There’s 42dB of trim available from nominal input gains of -14dBu (mic), with 16dB of headroom in the ADC. It handled everything well, with no need for pads on electric guitars amps or close miked drums. If you need any more gain, a digital trim feature can give +6dB boost to the track. Inputs are XLR/jack combo mic/line connectors with the last of the eight being able to switch between guitar or line for the jack input. The mic preamps are decent, and clean right up to the limit of their range where they start to introduce a little noise. And they worked well on most everything I put through them. Phantom is switchable over banks of four channels. The trim pots have a bit of play in them and no genuine level markings, which makes them feel a bit light on. While all the other knobs are stepped and feel sturdy.
Navigating the waters of the Portastudio is not too difficult. By the end of an hour, you can pretty much get the hang of it. To the right of the screen (which at first seems a little dinky, but you soon realise it covers everything you need) the buttons for Assign, Mixer, Dynamics and Effect let you navigate through the pages on the screen, and the Home screen can toggle between a mixer and timeline view. The timeline view comprises straight coloured bars for tracks, but by selecting a track, hitting Jog Play, and going into the waveform view, you can do some more detailed waveform editing. The record arm button doubles as a mute or solo selector, with the LED above it lighting up instead of the translucent button, when solo or mute mode is activated. It’s a simple system that works well. Timeline navigation is simple too. You can set up to 99 markers in a song, which double as navigation points and locators for setting the In and Out points for any editing functions. The transport buttons are inset, clear, round, and importantly, the record and play buttons light up when in use. In fact, the whole system is clearly thought out, each element has a different tactile feel, and there are plenty of signifiers to keep the DAW user (used to watching the tracks recorded in realtime) happy that recording is happening as intended. I first put the Tascam through its paces at a band rehearsal, recording to all eight tracks simultaneously for two hours straight. The unit didn’t falter once, and has been rock solid through mixdown with more tracks added. It did present one conundrum though. While I could set markers on the fly, I assumed that you would be able to separate these sections, and make them into their own song or ‘session’. But while there is the ability to do just about any editing, cutting, pasting, inside your track, there’s no easy way to copy specific portions of a longer recording post the fact and create different songs, you can only copy the entire song again and edit it down. Worse still, you can only create one master per song, so I couldn’t even work methodically through each song and export it. I thought the DP-24 would be perfect for those sort of set and forget recordings, so that was a bit of a shame. Best to stop and start between each track. LIGHT ON THE EFFECTS
Coming from DAW land, where with a basic laptop and any one of a number of software packages you can have access to unlimited processing and effects till your computer runs out of legs, the Tascam felt quite limiting. You can set compression on your inputs, but not add it to tracks in mixdown. Likewise, the guitar processor insert effect is only available on one channel. Obviously with only one guitar DI input, it makes sense to only have one guitar insert to get the sound before recording. The way to get around that is to bounce guitar tracks to another channel with the effects printed. You’re also limited to two dynamic inserts when you’re tracking with the guitar processor enabled. Despite the post processing limitations, you do get good control over dynamics on the way in.
Each of the eight inputs can have a compressor, noise suppressor, de-esser or exciter added to it, though not more than one. It works in a ‘last used’ manner, you can scroll through all four processors, and whatever processor you last use is the one that is assigned to that channel. The compressor had variable attack and release, threshold, ratio and makeup gain, but has no metering for gain reduction, the exciter did its job, but introduced a bit of high frequency noise, while I couldn’t understand what the de-esser was doing at all. It had a frequency control that ranged 1-4Hz (not anywhere near the sibilance range as far as I can tell!) and a ‘depth control’. The manual provided no information on how to use it either, and is quite light on in regard to effect parameters. The tests that I did with some vocal sibilance definitely removed a bit of the essing without any great side effect, so it does work, it’s just how that’s a mystery.
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You also get a single onboard effects processor that can be set up for Reverb, Delay or Chorus. You can use Send 1 on each channel to hook it up. The second send is for external effects, which are available over unbalanced jacks. THE PORTASTUDIO WAY
Taking a song from woah-to-go with the Portastudio is a step-by-step process. Once you’ve assigned your inputs, set up a bit of compression and tracked your song, you push the mixdown/ mastering button once to enter Mixdown mode, which locks out any more recording. Once you’ve got all your effects and EQ dialled in, you can set In and Out points and record a master file, setting levels with faders on the fly. Once you enter the Mastering mode, you can add multiband compression separated into low, medium and high bands, with crossovers at 125Hz–1kHz for the low point, and between 1-8kHz for the high crossover. You can also set one of six knee settings, and use auto or manual makeup gain. No limiting or loudness maximisers, just a normaliser, and dither. The EQ is the same as the track EQ, which has three broad ranging filters, a low and high shelf, and a mid control. The low shelf extends from 32Hz-1.6kHz, nearly meeting the high shelf starting point from 1.7-18kHz. The mid control can boost or cut frequencies with a variable Q that centres at any point across the entire EQ range (32Hz-18k). It’s a simple EQ that is useful for broad sculpting of tracks with 12dB of boost or cut, but not particularly surgical.
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Once you’re happy with your master, you can burn it to CD. Which is valuable for checking out the track or passing it around, but for some reason seems a bit arcane. Especially with the absence of any MP3 encoder — while obviously not encouraging quality, it’s still a handy way of sending around demo ideas. There are plenty of DAW-like features that have made their way into the DP-24. Virtual Track lets you store up to eight takes for each channel — great for auditioning solos or stacking vocal takes to edit together. The editing is quite complete, you can Copy/Paste to paste a selection over a track, or Copy/Insert to insert a selection and AT 25
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