Apple release Logic Pro X

Apple release Logic Pro X

For the first time since Logic Pro 9 four years ago, Apple have released a major update to their professional music recording and production software for Mac, with Logic Pro X. The numeral-savvy among you may have noticed that Apple have gone with an X rather than ’10’, mirroring video-editing app Final Cut Pro’s name change to FCPX.

As with Final Cut Pro X, it looks like this time round Apple have revamped the user interface of Logic Pro X, making it a bit more intuitive and user-friendly (think a more advanced version of GarageBand). It’s not just a cosmetic change though – new tools and a broader collection of virtual instruments and effects make Logic Pro X the most advanced and feature-packed version of Apple’s DAW to date. Here are a few of the standout features we’ll be most looking forward to…

Drummer. Logic Pro X now comes with its own virtual session drum player that will play along with you in a range of styles and techniques. Apple say, of the 15 drummers from the rock, alternative, songwriter and R&B genres, “each drummer has a custom kit, delivers his or her own signature sound, and can perform over a million unique groove and fill combinations”. There’s also Drum Kit Designer for creating your own deeply sampled, realistic-sounding drum kits.

MIDI Plug-ins. Logic Pro X features a number of MIDI plug-ins that can be applied to software instrument tracks, including a chord trigger, arpeggiator, transposer, randomiser and modulator. The MIDI plug-in engine is also scriptable which opens up a whole wealth of possibilities for the ardent programmer.

Track Stacks. Helping you manage complex sessions, Track Stacks allow you to consolidate tracks together either as a simple folder or summed to an Aux bus for sub mixing

Arrangement Track. Now you can use markers in the Arrangement Track to define sections such as verse, bridge and chorus, and then easily move or delete sections from your arrangement.

Flex Pitch. The Flex Pitch plug-in means you can now fine-tune pitch when you’re tuning instruments or vocals, and manipulate melodic content in pitch and time domains. Roll over each note so the parameters show up, then you can adjust the gain of individual notes without excessive compression or time-consuming automation editing.

Logic Remote. Now you can control your Logic Pro workflow wirelessly from anywhere using iPad, and even play instruments directly on iPad using the multitouch interface.

One of the caveats of Logic Pro X is that it is now exclusively 64-bit, and compatibility with third party plug-ins is limited to those which have 64-bit versions. While Logic Pro X is backwardly compatible with sessions made using 32-bit versions, and plug-ins that are 32-bit only won’t run, tread carefully if you have a lot of older plug-ins. This might be a short term bugbear, but hopefully it will prompt developers to update their plug-ins to 64-bit sooner rather than later.

We’re going to be taking a more in-depth look at how Apple’s Logic Pro X really performs soon, but in the meantime don’t hesitate to post any comments or questions in the box below, or get in touch with one of the team…

Want to know more? Give us a call on 03332 409 306 or email You can also follow @Jigsaw24Audio on Twitter or ‘Like’ us on Facebook

Who wants to see a Magma ExpressBox 3T Thunderbolt Chassis in action?

Who wants to see a Magma ExpressBox 3T Thunderbolt Chassis in action?

Avid created quite a buzz at NAB when they demonstrated Pro Tools HDX running in a Magma Thunderbolt expansion chassis. What was on show was very much a pre-release version of the chassis and a beta version of Pro Tools, but it garnered a lot of interest from those looking to either get a full-power Pro Tools HDX or HD Native system in a portable format, or to opt for an iMac or Mac mini rather than a tower-based system.

In the intervening months we’ve been fielding plenty of questions from excited Pro Tools users. The only information we’ve been able to pass on up to now was that we’d seen the demo version running, and that Avid had confirmed that they will be qualifying the Magma 3T chassis as their approved Thunderbolt solution. (This was always a contingency in the event of Apple’s Mac Pro ceasing to be available, and the relationship between the two companies goes way back to the qualification of the original PCI CardBus chassis.)

Magma 3T expansion chassis

Magma 3T expansion chassis

Finally, however, the Magma 3T is shipping…

Stock is expected imminently and we’ve had a chance to test one of the first units. The Magma ExpressBox 3T, as its name suggests, is a 3-slot expansion chassis for PCIe cards, with two of those slots being x8 and the other x4. The unit is powered via a standard IEC cable and there are internal power cables for cards such as the HDX cards, which require more power.

Getting the unit up and running has been simplicity itself

Simply fit the Pro Tools HDX or HD Native card(s) in the slots as you would in a Mac Pro, connect a Thunderbolt cable between your computer and the chassis, and turn it on. The chassis itself fires up as soon as you turn the computer on, and the latest version of Pro Tools HD (10.2) just sees the cards straight away. There is absolutely no difference in the user experience between this setup and a traditional Mac Pro / Windows workstation.

You do need to have Pro Tools HD 10.2 software in order for this to work, and I need to draw attention to the caveat on Avid’s website – namely the one that states that Thunderbolt chassis are not supported in this release. Avid have confirmed this is because official testing has not yet been completed and there may be some permutation still to be tested before they finally make this a supported solution. In use (and I have been using this setup for a week now and demonstrating it to customers in a variety of scenarios) I haven’t found anything that trips it up, it has been completely solid in performance. As well as running it with an HDX2 system, ours also has a Decklink Intensity card for video playout (click on the gallery link below to take a closer look).

We’re happy to arrange a demonstration for anyone who is interested in seeing the Magma ExpressBox 3T chassis running a full Pro Tools HDX / HD Native system. If you would like a demonstration, please give us a call on 03332 409 306 or email

Why you need TC’s LM2 Radar Loudness Meter

Why you need TC’s LM2 Radar Loudness Meter

TC Electronic know the importance of loudness in broadcast audio, being renowned for their top TM7 and TM9 metering hardware, and LM6 metering plug-in. Now they’ve made it much more accessible, with an affordable LM2 Radar Loudness Meter plug-in that works with all major DAWs.

What sets the LM2 apart from the competition is the Radar view. Instead of the usual (pretty dull) numeric or histogram readout, the meter is displayed as a funky radar – with the main view showing you loudness history, while the outer ring displays momentary short-term loudness. It’s much easier to get an idea of loudness at a glance, but also displays an accurate readout too.

The LM2 plug-in looks set to be a good addition for anyone working in broadcast, as it will work in pretty much any DAW (in stereo), and most digital video editors on Mac OS X or PC. It supports all major plug-in formats, including AAX, RTAS and Audio Suite for Pro Tools, as well as AU and VST. At the price point ($99 until the 1st July), there aren’t really any competitors on the market which can match the LM2. Apart from TC’s LM6 native loudness meter, which you’ll need if you’re working in surround rather than just stereo.

Visit our store to find out more about TC Electronic’s range of loudness metering hardware and software options. You can also call us on 03332 409 306, email or keep up with the latest audio news and offers on our Twitter (@Jigsaw24Audio) and Facebook page.

What is MADI? A guide to the audio format

What is MADI? A guide to the audio format

If you work with broadcast audio and frequently face the problem of how to move a lot of channels of audio over a long distance, such as outside broadcast or around a large production environment, then you’re probably already familiar with the MADI protocol. If not, then it could be time to embrace the technology and take a break from hugely heavy multicores, complex switching systems and limiting audio-over-Ethernet solutions.

What is MADI?

Put simply, MADI (Multi-channel Audio Digital Interface) allows you to send up to 64 discrete channels of audio via a single fibre optic cable or coaxial cable and over distances of up to 2km. It supports audio formats up to 24-bit/192kHz and doesn’t use lossy compression. For this reason MADI has become an obvious choice for many of the largest outside broadcast operations, and for creating an audio network around a building where bulky multicores would be undesirable and prohibitively expensive.

The MADI format has been adopted my most of the leading pro audio manufacturers, including AVID, RME, Euphonix, SADiE, SSL, Yamaha and Soundcraft, and products that use the technology range from I/O formats on consoles, audio capture cards for computers and remotely controllable mic preamps (the MADI protocol also supports control data).


 In addition to a huge channel count and range, the other key advantage of MADI is flexibility. The individual audio channels within a MADI stream can be routed, split and recombined independently, enabling a source to be sent to multiple destinations or a single recorder to capture from multiple sources. RME, for example, manufacture an 8×8 MADI router capable of accepting up to 512 channels of audio and routing any of them to any combination the 512 available outputs.

Remotely controllable and taking up just 1U of rack space, that’s some serious flexibility! In fact, even RME’s MADI PCIe capture cards offer this sort of flexibility, as the onboard TotalMix allows for flexible routing of all input and output streams independently of recording.


 A single MADI stream will give you:

  • 64 channels of input + 64 channels of output at 24-bit 48kHz
  • 32 channels of input + 32 channels of output at 24-bit 96kHz
  • 16 channels of input + 16 channels of output at 24-bit 192kHz
  • Range: 2000m over fibre (SFP) or 100m over coaxial copper (75Ω, bnc)

The opportunities of MADI for broadcasters are huge. It offers a level of flexibility that is almost impossible to achieve with standard analogue cabling. With fibre cable costs (and weight) being a fraction of that of copper multicore, it offers a lighter, cheaper, more flexible, more robust and faster deployment solution for creating a fixed or portable audio network than any other system currently available.

 If you want to know more about audio formats and systems, give us a call on 03332 409 306, email or leave us a comment below. You can also keep up with more news, reviews and offers by following us on Twitter (@Jigsaw24Audio).

Review: The M-Audio Fast Track C600 interface

Review: The M-Audio Fast Track C600 interface

Over the festive period, I had quite a bit of work to do completing an album project, so decided to tie that in with testing one of our new batch of M-Audio C600 audio interfaces. Features like the scriptable buttons and especially the new ‘Multi’ button turned out to be a real time saver…

Reviewing audio interfaces is a tricky task, so I wanted to make sure I gave M-Audio’s Fast Track C600 interface a thorough trial. Sure, you can blast through the features, plug in some sources, have a listen, and it’s easy enough to spot the good points and the glaring omissions. But if you want more than a cursory overview, you have to spend some quality time with it, and that means getting stuck into a project to see how the various components perform when it matters.

What the C600 brings to the table

The C600 and its smaller sibling, the C400, are a departure for M-Audio. There’s a growing ethos that the audio interface need not be a dumb box stuck in a rack somewhere, but something that can sit on your desktop and offer additional functionality and productivity. The C600 is certainly one such unit. A 24-bit/96kHz USB 2.0 interface with four mic/line inputs (two of which have instrument inputs), stereo S/PDIF, twin headphone outputs and six analogue outputs, it also brings some added bonuses to the the table. It has monitor control of up to three pairs of speakers, transport control, onboard monitor mixing (complete with delay and reverb) and the unique ‘Multi’ button, which allows for scriptable actions (more on this later).

M-Audio C600 in action

During the time I had the C600, I needed to add some backing vocals, track some electric and acoustic guitars and of course I had a load of mixes to do, so pretty much every aspect of the interface would have its work cut out. The first thing I noticed was that the sound quality of the C600 is very good. Avid make a point of saying they have “leveraged technology” from the HD Omni interface for these units and you can hear that in the quality of the mic preamps and converters. Mic signals have plenty of clean headroom and very low noise, and output has a huge frequency range with exceptional stereo imaging. The instrument inputs handled electric guitars perfectly and produced predictably good results through IK’s AmpliTube.

The control section

For an interface that’s designed for project studio use, the C600’s control section is a real strong point. You can connect up to three sets of monitor speakers and switch between them via dedicated buttons on the interface, with the control software allowing you to level match. Being able to do A/B comparisons is hugely useful and usually requires some sort of monitor controllers; unfortunately such devices inevitably colour the signal so being able to do it without intervention is a real boon.

M-Audio C600 in action

The C600 usefully features a set of transport control buttons, but what’s even more useful is that each button can be re-allocated, so they can be mapped to any control function in your DAW or even be assigned as shortcut buttons. I’ve never needed to use a rewind or fast-forward button on a non-linear editor, so being able to map one or the other to a function such as ‘save’ is incredibly helpful.

Taking this idea one step further is the ‘Multi’ button. This button allows you to perform any series of actions that you can do with key commands, at up to eight steps. You can define the key command for each step using the control panel, and pressing the Multi button takes you through the sequence one step at a time – a real time saver if you find yourself performing the same sequence of key commands repeatedly. Different setups for the Multi button can be saved too, so you’re able to have different functionality for different tasks.

The control panel software is clear and easy to use, and the mixer especially is extremely functional and lets you balance incoming signals and software returns for accurate tracking. It’s a DSP-driven system which gives near-zero latency and provides reverb and delay for comfort monitoring when tracking. It’s also designed to offer independent headphone mixes to each of the headphone outputs (both of which, I should mention, offer loud, clear and very high quality output). One additional thing that is often overlooked – both the control panel and drivers are very stable. Not once did I experience any unusual behaviour or unexpected quitting. Sadly this is not the norm for even considerably more expensive high-end units.

M-Audio C600 in action

There are the inevitable niggles but that’s because, like any user, I’d like the moon on a stick. I would love a version of this interface with more inputs such as an ADAT so I could accommodate recording a drum kit for instance (are you listening Avid?). The monitor control section doesn’t offer a dim control or mono switch which you’d normally find on a dedicated controller, but to be honest I can’t recall the last time I used either and it’s certainly something I would do without if the alternative is to colour the sound with another unit between the output and my ears. The only frustrating omission in my view is the lack of a talkback mic, which meant some wild gesticulating and shouting to attract the singer’s attention.

The verdict

This interface is a real winner. If you’re in the market for a project studio interface with some real time-saving factions, the M-Audio Fast Track C600 is worth investigating. The scriptable buttons are a huge gift to the musician, as they eliminate a lot of breaking of musical flow as you switch from ‘playing’ mode to ‘computer operator’ mode; something that happens every time you pick up the mouse. The fact the buttons are simply performing keystrokes (rather than being tied to DAW functions) and are fully scriptable means that video editors (who perform far more repetitive keystroke-oriented tasks than musicians, and always need monitor control) could find this really helps speed up their workflow.

For more information on the M-Audio Fast Track C600 interface, call 03332 409 306, email or leave us a comment below. You can also keep up with more news, reviews and offers on our Twitter (@Jigsaw24Audio) and Facebook page.


Why listening to Fleetwood Mac’s ‘Rumours’ on vinyl made me want to burn my CDs

Why listening to Fleetwood Mac’s ‘Rumours’ on vinyl made me want to burn my CDs

Every so often I feel compelled to spend an evening pulling out my record collection and rediscovering a time when I actively enjoyed the process of listening to music. This happens with almost alarming certainty when I have either a) had a little too much to drink or b) split up with my girlfriend (sometimes an unhelpful combination of both). And in almost all cases I seem to arrive at the same conclusion – that for some reason vinyl sounds better.

After much ruminating I have arrived at the conclusion that this has nothing to do with me being some closet analogue purist. I don’t think there is anything intrinsically wrong with my speakers being wobbled by a stream of 1s and 0s as opposed to a stylus jiggling in a groove on a vinyl disc. It has nothing to do with the hiss and crackle of vinyl imparting the pseudo-comforting sound of nature or acting as the sonic glue that imparts a sense of life into an otherwise sterile performance. In fact it is not about how vinyl sounds when compared to CD at all, it is about how the music on a CDcompares with its vinyl equivalent, a result of the process I have come to call ‘masterdisation.’

A common mistake made by advocates of vinyl is that a CD has less dynamic range. The CD format is capable of a dynamic range of 96dB as opposed to around 65 – 70dB for a vinyl record. However, the process of mastering for vinyl favoured using as much dynamic range as was possible, with the caveat that the quietest part should never fall below the agreed noise-floor for the background sounds inherent in a device which basically drags a needle across a plastic surface.

Mastering engineers were still encouraged to try and make the loudest records possible, but there was a limit because above a certain level the needle would literally jump out of the groove. With CDs, the opposite is true. Record company executives looked at the loudness wars in the ’80s, when radio stations competed to get more listeners by being the loudest on the air, and decided they were prepared to sacrifice dynamics if they could have a record that seemed louder than every other.


With dynamics no longer a concern, CD mastering engineers found themselves armed with the same tools as the radios had used. Multiband compressors and limiters let them compress most of those 1s and 0s into straight 1s and despite having a much larger dynamic range than vinyl, it is common for a modern pop CD to be mastered with less than 10dB difference between the loudest and quietest parts. And this is, I think, the key to why so many people claim a preference for vinyl.

Firstly, dynamics are a key dimension in audio. It holds listeners’ interest and we start to actively listen. With no dynamics, listeners get fatigued and lose interest. Secondly, overly loud mastering introduces digital distortions, as CD player converters run out of headroom to recreate the soundwave. In his book ‘Perfecting Sound Forever: The Story of Recorded Music’ (so exhaustively researched it frankly has no business being as enjoyable or entertaining as it is), author Greg Milner cites The Red Hot Chilli Peppers’ ‘Californication’ as being a watershed album for ‘overloud’ mastering. Almost devoid of dynamics (a total dynamic range of less than 6dB across the whole album), the sound of digital clipping produced throughout is recognised by our brains as being painfully loud regardless of how loud the disc is actually being played, and actually becomes unpleasant to listen to.

Finally, all of this compression started to fundamentally change how we perceived the sounds of instruments. Sounds were robbed of transients and others had subtleties boosted. CDs started to sound less like music played on vinyl and more like music heard on the radio. We no longer needed to listen to records, because they were practically screaming at us, the musical equivalent of over-hyped orange-lacquered reality TV celebrities shrieking into our headphones.

The irony of this is that, as CDs used loudness to attract our attention, the effect made the listener less interested. It’s a shame the CD format was standardised before the loudness wars started. In the digital TV age, broadcasters now have access to loudness metadata which allows them to match perceived loudness of different pieces of programme material. If CDs could somehow incorporate the same loudness metadata, a CD wouldn’t have to compete on volume – playback systems would be able to compensate in balancing volumes between different albums based on how loudly the listener will perceive them. Overloud mastering would become undesirable due to the artifacts and limited dynamic range.

While some artists are beginning to see that overloud mastering is detrimental to the enjoyment of the music, the mastering decisions rarely rest with the artist. It may well be that, in future, radio will incorporate loudness monitoring that will help in the fight to reclaim the music from the sound of the CD. For anyone looking to master their own music, I’d advocate paying close attention to how your music sounds, not just how loud it is. Squashing all the transients out of your music may end up reducing a lot more than just the peaks.

Want to know more about mastering? Get in touch with our audio team on 03332 409 306 or email

Direct USB recording with RME’s Fireface UFX update

Direct USB recording with RME’s Fireface UFX update

An oddity that didn’t escape the attention of those who saw RME’s flagship Fireface UFX audio interface at launch is that it had a USB slot on the front panel. There was very little mention of what it was for, but the rumours were that you would be able to record directly to a mass storage device at some point in the future.

Well, this functionality has now entered public ‘alpha test’ phase, so last weekend I downloaded and installed it to give it a try. Turns out, it’s rather good…

In order to activate the recording features, a new version of the UFX firmware needs to be uploaded to the interface, which is done via a PC or Mac over USB. Once updated, the Meters button on the front of the unit allows you to toggle the recording controls. Setting this up is simple; use the Channel button to scroll through all the channels and activate the ‘Record’ check box for the channels you want to record from. Your inputs will be recorded directly to whatever storage device is connected to the front USB port. There are just a handful of caveats before you start:

– The recorded file will be a single multichannel WAV, not individual files.

– Drives must be formatted to FAT32, otherwise you’ll see a File System Error message.

– Your recording will be dry inputs, so effects added within TotalMix will not be captured.Some drives don’t seem to work, but most do.

– As a point of reference, I had no problem recording ten minutes of 20 tracks on a 4GB Kingston memory stick.

Uses for direct recording…

Although this is not an official release – and RME are using this period to iron out any flaws and incompatibilities – I can’t praise this update highly enough. There are clear uses for this technology, from having a safety recording running in the event of a DAW crash to being able to record live gigs where using a computer might be ill-advised. (Excessive bass vibrations, for example, can play havoc with internal drives in computers, and Apple MacBooks have a safety feature that ‘parks’ the hard drive in the event of it being dropped to prevent head damage. The problem is, it can’t distinguish between bass vibrations and a nasty fall, resulting in some untimely shutdowns when recording!)

Sometimes there are just situations where a standalone recorder is what you want to use, and that’s exactly what this firmware update turns the UFX into – a standalone hard disk recorder. The decision to record a single multichannel audio file is a good one too, as it makes it much easier to write high data volumes to slower devices (such as memory sticks) than trying to simultaneously write multiple files. It also ensures that all files remain synchronous when importing into an editing program like Pro Tools or Cubase which both handle multichannel files natively, automatically showing each channel as a separate region.

A couple of (minor) downsides…

Unfortunately for users of Apple’s Logic Pro and Logic Express, Logic doesn’t handle multichannel audio at all (other than surround formats) but you can use a freeware application such as Audacity to export the individual tracks from the multitrack WAV, after which it is business as usual. It’s also unfortunate that the file format is FAT32, as this imposes a 2GB size limit on recording files. If you’re recording from all 28 available mono inputs then you’ve got just under 15 minutes of run time before you have to drop out of record and start a new file but, unless you’re recording a prog rock opera, that should just be a case of waiting for a gap between songs.

My overall verdict…

The RME Fireface UFX was already one of the best professional audio interfaces available based on stability features and sheer audio performance, but once this update leaves preview and becomes official, it’s going to stand out from the competition, pushing the UFX into an exciting class of its own and making it a simple choice for people looking to record critical, non-repeatable performances.

Head over to the RME forums to download the pre-release firmware (including an updated version of TotalMix). You can also visit our website to get your hands on RME’s Fireface UFX interface.

Call us for more information on 03332 409 306, email or leave your thoughts in the comments box below and we’ll be in touch.

Novation’s UltraNova Synth – A review

Novation’s UltraNova Synth – A review

On first impressions, Novation’s UltraNova hardware synth is a beautiful thing, with its pitchbend and modulation wheels glowing a cool blue, and the 3-octave keyboard set into a matching blue housing with little red buttons.

Not just a pretty face though, the UltraNova also includes a gooseneck microphone that slots into an XLR socket on the top panel for vocoding and can even double as an audio interface by hooking it up to your computer via the USB socket and I/O jacks on the back. Handy!

So, it looks great, but what does it actually do?

First, a bit of history. Novation have long been known for producing quality hardware controllers for musicians working on computers, and also have a background in hardware synthesisers. In 1998, they designed the highly regarded SuperNova synth rack, capable of producing immense pads and atmospheric textures. The UltraNova is the latest in a line of synths based on that original rack, and Novation have been improving and innovating along the way.


As a performance keyboard, the UltraNova’s both responsive and fun to play. Working through the presets on offer (there are four banks of 127 each, some of which are blank patches), it becomes obvious fairly quickly that Novation have a wide user group in mind. Nasty dubstep bass sounds sit side-by-side with Eno-esque washes and Jean-Michel Jarre arpeggios.

The synth engine in the UltraNova is extremely powerful. Three oscillators, a noise generator and two ring modulators provide the sound sources, with each oscillator drawing on a bank of 14 analogue waveform simulations, 20 digital ones and 36 wavetables. The sources are mixed, then pass through two separate filters on their way to the enveloped amplifier and effects units. No less than 14 different filter types can be used, and the two filters can be used in different types of parallel and series arrangements, independently or with their cutoff and/or resonance linked. There are also filter distortion modes, with esoteric names like ‘Valve’ and ‘Diode’ which crunch things up rather nicely.


Oscillators are the key to a synth’s character, and these don’t disappoint. The waveforms are extremely useable in themselves, and there are some little tricks available to make them even more interesting. For a start, each oscillator has a ‘density’ control which seems to add multiple instances of the same wave, and turning the control produces the sound of several oscillators in unison. There’s a detune control for this, so even a single oscillator can sound like massed synths. Not only that, but each oscillator can be put into hard sync with itself, and the harmonic series adjusted by detuning the sync source. This is a classic hard, cutting sound greatly loved in techno music, and it normally needs all of a synth’s power to produce it. But, in the UltraNova, I can build sounds with three of these at once if I really want to.

Finally the sound escapes via five effects slots, stackable and splittable just in case you want to experiment with compressed reverb layered with distorted echo, for example. Pretty much every sound on the UltraNova can be modulated by pretty much anything else (with 20 sources and 66 destinations), and some of the presets make impressive use of the possibilities, sounding hugely complex and full of motion.

And yes, the vocoder sounds pretty good too. It’s only a 12-band device, but very useable. If you want to, you can process any analogue input using the synth section, so even guitarists and drummers can get something out of this little synth.


With a bit of clever use of its ten knobs, programming the UltraNova is relatively simple and never tedious. One large knob always selects patches and, in performance, the other large knob normally alters filter cutoff. The other eight smaller knobs above the 144-character display edit whatever parameter is directly under them. Press the Filter button, for instance, and you get all eight parameters for Filter 1 on the eight controls. Press the Select Down key and all the parameters for Filter 2 appear. Press Next Page, and the shared parameters for the filters appear. Easy.

The eight knobs respond to touch too, so simply tapping one puts that parameter onto the large Filter knob. This can then be ‘locked’ so the large knob permanently edits that parameter, even if you switch to a different page – really handy if you want to balance, say, filter and effects distortion without toggling pages. Even better, you can choose your favourite eight parameters for each individual patch and assign them to the eight controls using the ‘Tweak’ page so, during performance, you have exactly the parameters you want to play with all on one page.


It’s been a pleasure exploring the Novation UltraNova, I must say. There’s a lot to like here, and very little to criticise. If anything, it’s a little too diverse, and perhaps anyone who spends ten minutes trying out the patches will come away thinking that only 10% of them are useful. The point is: it’s a synth with something for everyone and it’s possible to make sounds with it that are personal and, above all, different. On reflection, the UltraNova is well worth the investment in time to explore properly.

Check it out in action in the video below.

For more information on the Novation UltraNova hardware synth (with free stand and headphones!), give us a call on 03332 409 306 or email We’d also love to hear your thoughts on the UltraNova, so feel free to leave a comment and we’ll be in touch.

Audient ASP008 Review

Audient ASP008 Review

For anyone looking to add a number of microphone preamps to a digital recording setup, a quick trawl of the web will show that 8 channel mic preamps are in plentiful supply. With so many manufacturers moving production to China to compete on price,  it would seem that Audient have their work cut out for them if they are to try and gain a foothold in such a competitive market.

But Audient aren’t here to compete on price. There are a lot of multi-channel preamps in the sub-£500 price bracket, such as Focusrite’s Octopre and the Presonus Digimax, but then precious little until you get to units such as the Focusrite ISA 828 at over £1500. With the ASP008, Audient have filled that gap – it’s an 8 channel preamp with digital outs, yes, but it eschews the cheaper IC and op-amp based circuitry of mass manufactured units in favour of an all-analogue, transformer-based Discrete Class A design, and adds variable impedance on all inputs to the mix. Oh, and they are all assembled in England if you are interested.

Audient are best known for their analogue consoles and the ASP008’s analogue heritage is apparent the minute you unpack it – it’s heavy. And heavy is good, because heavy means a big power transformer to deliver constant voltage across the components, and real transformers handling the signal, rather than PCBs. My geek tendencies compelled me to open the lid and I can definitely confirm that!


The ASP008 offers eight mic inputs on the rear panel via female XLR sockets. Each channel has individual ‘soft start’ phantom power, a switch to trim to line level, a phase switch and a -12dB/octave high-pass filter which is variable from 25Hz to 250Hz. Each channel also has a 3-position impedance switch, offering 200Ω, 1.5kΩ and 5kΩ load values. Channels 1 and 2 also feature front panel instrument inputs and -20dB pad switches.

The rear of the unit has a DB25 connector for all eight line level inputs, another for the analogue outputs and, if you have the digital output board (which, lets face it, is the only sensible way to buy the unit) you also have ADAT out sockets supporting SMUX up to 96KHz, eight channels of AES/EBU (also switchable to SPDIF) via a 9-pin D-connector and a wordclock input. Digitally, the ASP008 can run up to 96KHz and a rear button selects between internal and external clocking.


So the Audient ASP008 is an extremely well-specified unit as far as connectivity goes, but the important functions of any mic preamp is how good it sounds and in particular how well it responds to the mic. And this is where the ASP008 really excels. Audient claim that distortion is less than 0.001% with 20dB gain, and it’s certainly apparent that the unit has a huge amount of headroom available. It’s not a crystal clear transparent unit, but rather added a wonderful analogue warmth to pretty much any signal that I fed through it. Lows were rich and detailed, mids were clear and well defined and high frequencies never seemed to inherit an air of brittleness that plagues many cheaper units (especially at higher gain settings) and the noise floor is incredibly low.

But the real trump card for the ASP008 is the variable impedance settings for each mic preamp. Changing the load that a microphone ‘sees’ can have anything from a subtle to drastic effect on the sound of a microphone across frequency response, dynamic range and transient response. Modern transformer-less condensers exhibit less of an effect but older, transformer-coupled mics, dynamics and ribbons definitely change character as the impedance is changed, giving you a whole new palette of sounds to work with.


The Audient ASP008 is not aimed at the user who just wants to add some mic inputs to their digital recording setup. Instead, it’s aimed at users who want some of that analogue magic to infiltrate their pristine digital world and experience a bit more depth from their mics. Pro Tools HD users in particular will love the fact that the unit has AES/EBU out, so they won’t be limited to ADAT-only digital connections. At its price point, the Audient’s only real competition is the RME Octamic II, which is no less wonderful but entirely different in character – being an example in transparency. But if it’s warmth and character you’re looking for, I’d recommend the Audient ASP008 all the way.

If you want to try the Audient ASP008 we have loan units available to try in your own studio. For more information, call our audio team on 03332 400 300 or email

Apogee new Symphony I/O interface and converter offers Pro Tools HD and Logic connectivity

Apogee new Symphony I/O interface and converter offers Pro Tools HD and Logic connectivity

Apogee have announced a brand new flagship audio converter and interface aimed squarely at the professional market. The Symphony I/O is capable of up to 32 channels of I/O and is a fully modular unit, giving the user a choice of five I/O modules to fit into the chassis.

The chassis itself features a pair of built-in Pro Tools HD interface connection ports , as well as USB 2.0, Ethernet, wordclock and loop sync connections. The ports can also be used for connection to Apogee’s own Symphony and Symphony Mobile PCIe cards for users of native DAWs such as Logic, DP and Nuendo.

The standard Symphony I/O includes one module pre-installed which offers eight channels of analogue and eight channels of ADAT I/O. This configuration can be added to or swapped with four other available modules. These feature:

Eight channels of analogue + eight channels of AES/EBU I/O.

– 16 channels of analogue input and 16 channels of ADAT output.

– 16 channels of  ADAT input and 16 channels of analogue output.

– 8-channel mic preamp module with four instrument inputs and eight insert points. (This module works in conjunction with the standard 8-channel analogue input module to add a digitally controlled, 85dB microphone preamp to each input).

We’ve seen no shortage of high-end analogue converters emerge in the last few years, but this release from Apogee puts them very much at the head of the pack again. It’s a huge step forward for those looking for a real professional solution.

Apogee clearly know their market and the inclusion of the sockets that allow connection directly to Pro Tools core cards (previously an option for the Rosetta series of converters) is very welcome. Particularly in this case, as the interface can be switched from Pro Tools mode to Symphony mode for those using Apogee’s own card.

I must admit, for me this came as no surprise given that both formats use the same ‘proprietary’ connection format, but it’s still good news nonetheless. And with such a wealth of I/O options available, it’s going to be a tough customer that can’t find a way to configure it to meet his or her needs.

Apogee also claim the redesigned circuitry now uses fewer components and higher quality ones to minimise the signal path even further. This results in a flatter frequency response and improved phase error performance, which should mean that Symphony I/O sounds even better than their previous converters.

For more information, get in touch with our team on 03332 400 222 or email Or have a look at the complete range of Symphony I/O products here.