Tascam’s DR-60D offers DSLR filmmakers affordable, professional audio

Tascam’s DR-60D offers DSLR filmmakers affordable, professional audio

Anyone wanting to shoot video on a DSLR has just been given a bit of a boost, with the arrival of Tascam’s new DR-60D 4-channel audio recorder and mixer. Whereas before you would have had to use an external box with preamps or a location recorder to capture audio, the Tascam DR-60D  gives you great recording quality and monitoring at an affordable price.

The DR-60D attaches directly to your DSLR, so you can screw it on to any tripod and use it unnoticed. You can then simply plug in all your mics, lavaliers, booms etc and record up to four separate channels simultaneously, while it mixes down the audio in realtime and sends a stereo feed to the camera. That means you get your four channel recording, and a separate copy which you can use as a ‘rough guide’.

Good, professional sound

The main points to remember though are quality and price. Tascam have a reputation of manufacturing very good sounding professional hardware, and the DR-60D sounds miles better than recording directly to your camera. Able to record up to 24-bit 96k audio directly to SD/SDHC cards, the DR-60D also acts as a mixer, passing through a stereo mix down to the camera. It features two XLR inputs with phantom power and an additional stereo line input, plus headphone monitoring directly off the unit.

Very affordable option

Then we get to price – at only £265.83 ex VAT, the Tascam DR-60D costs about the same as traditional XLR adapters (such as the Beachtek DXA range) but offers a lot more functionality, failsafe audio  and higher quality sound, making it a very attractive and affordable option for serious DSLR shooters. It’s already proving popular in the US, especially with those who have an eye on getting a new Canon 7D or Blackmagic Design Pocket Cinema Camera, and we’re going to have some of the first shipment in the UK in May, so get your pre-orders of the Tascam DR-60D 4 channel audio recorder and mixer in now!

Want to know more about the Tascam DR-60DGive us a call on 03332 409 306 or email audio@Jigsaw24.com. You can also follow @Jigsaw24Audio on Twitter or ‘Like’ us on Facebook

NAB 2013 news: Genelec add Intelligent Signal Sensing to speakers

No massive hardware updates from Genelec at NAB 2013, but they have announced that a new eco-friendly power saving mode will now be rolled out across the whole of their 8000 series speakers – their Intelligent Signal Sensing (ISS) system.

The new Genelec speaker models are exactly the same as the old ones, performance wise, it’s just that they now incorporate a standby mode which is engaged when no audio has been passed through the speaker for an hour, causing them to go into a low power mode (1/2 a watt) to ensure we all stay green, and in pocket too!

“The typical idle state power consumption of an active loudspeaker, or any other audio amplifier system, is between 10 and 20 watts,” said Lars-Olof Janflod, Marketing and Public Relations Director at Genelec. “If left in standby, this level of consumption is maintained by the amplifier system and released as heat. Over time this can total a significant amount which results in both negative fiscal and environmental costs.”

Genelec aren’t actually unveiling the Intelligent Signal Sensing feature until the ISE show later this month, but if you’re in Vegas, you could probably go and bug them about it at NAB 2013, stand C1633.

Want to know more about our full range of Genelec speakers and studio monitors? Give us a call on 03332 409 306 or email broadcast@Jigsaw24.com. For all the latest NAB 2013 news, follow @Jigsaw24Video on Twitter‘Like’ us on Facebook or check out our roundup post

NAB 2013 news: TC Electronic’s DB6 Loudness Processors and Adobe hook-up

Loudness monitoring is now an imperative part of all broadcast, and no-one does it better than TC Electronic. Their latest DB6 solutions are made specifically to fit in your rack between the output of whatever you’re transmitting and input of whatever you’re transmitting to. They’re going to be premiering at NAB 2013 in Las Vegas and, more importantly, are now available to pre-order from Jigsaw24…

The TC Electronic DB6 Loudness Processors come in two varieties – single channel and multi-channel – and are designed to make sure all your audio is in check, by letting you monitor and adjust your output on the fly. Both models are fully compliant with all major broadcast standards, including ITU BS.1770-3, ATSC A/85, EBU R128, ARIB TR-B32, OP-59 and more.

Powering both is TC’s Loudness Wizard software which takes care of all the correction and format conversion (stereo, 5.1 etc), as well as TC’s LM6 Loudness Radar Meter, the output and logging of which can be monitored via a connected PC or Mac.

TC Electronic DB6 Single Broadcast Audio Loudness Proccessor. This single channel version of the DB6 lets you process a single stream of SD/HD/3G and gives you two 5.1 capable processing engines per SDI stream, each able of delivering to any platform and any codec; be it HDTV, Mobile TV, Digital Radio or Podcast using AAC, Dolby, Ogg Vorbis, Lossless or Linear. Housed in a single rack space, this makes DB6 one of the most compact and powerful solutions on the market.

Pre-order the TC Electronic DB6 Single now.

TC Electronic DB6 Multi Broadcast Audio Loudness Proccessor. As the name suggests, this version of the DB6 lets you monitor multiple channels. It’s expandable to cope with up to three streams of audio simultaneously, letting you monitor values of each stream completely independently. Because of this, it’s ideal for live broadcast, and especially OB trucks, where you need to handle multiple channels of audio at once.

Pre-order the TC Electronic DB6 Multi now.

LoudnessRadar now integrated into Adobe

If that wasn’t enough metering goodness from TC Electronic for you, you’ll be happy to know that TC have now hooked up with Adobe to offer their LM6 loudness monitoring software in Premiere Pro and Audition. LoudnesRadar gives an easy-to-read overview of loudness over time, displaying Loudness History on a revolving radar, Momentary Loudness on an outer ring, True-Peak clips, Program Loudness and Loudness Range in a single view. All very handy for keeping on top of your levels all within your Adobe apps.

TC Electronic will also be playing host to a number of audio loudness seminars at NAB 2013. If you’re lucky enough to be in Vegas, head over to Central Hall, Meeting Room C204 for what will surely be some interesting advice on loudness.

Want to know more about the TC Electronic DB6 Broadcast Audio Loudness Proccessors? Give us a call on 03332 409 306 or email broadcast@Jigsaw24.com. For all the latest NAB 2013 news, follow @Jigsaw24Video on Twitter‘Like’ us on Facebook or check out our roundup post

Steinberg Cubase reaches 7th heaven

Steinberg Cubase reaches 7th heaven

Steinberg have released the magnificient seventh edition of their Cubase music production software to a chorus of trumpets, synths and a whole host of other virtual instruments. It’s certainly one of the most popular DAWs out there, but is Cubase 7 the most feature-packed version yet? Let’s take a look…

What’s new in Cubase 7?

Perhaps the most salient points to pick out in the new Steinberg Cubase 7 music production software are an all-new mixer, on-board monitoring and better collaboration features. To start, themixer – which supposedly sounds better than ever – also has customisable views to speed up mixing by keeping your most important controls on show. There’s also a new channel strip with EQ and dynamics.

Cubase 7 also has its own built-in EBU 128 loudness monitoring, which is a first for any DAW. In an industry increasingly governed by acceptable loudness standards, having monitoring tools right there in your DAW without having to fork out for expensive hardware or plug-ins is a real boon – especially for smaller broadcast studios.

Other useful features to pick out include the ability to collaborate with people in other studios in realtime via VST Connect SE. In terms of music-making, the new VariAudio system and Chord track interact to let you create perfect harmonies easily, and there’s simple drag and drop plug-in support, which makes adding effects and instruments a breeze.

When and where can I get Cubase 7?

Steinberg’s elves have been working away to get this out in time for Christmas, and you’ll be able to get your copy or upgrade in the first week of December. You can pre-order Cubase 7 from Jigsaw24 now. If you’ve only just bought Cubase 6.5, don’t worry, Steinberg are offering a free upgrade grace period to v7 for anyone who’s purchased after October 26th (just sign in to your e-Licenser account after the first week of December, and v7 will be waiting for you).

– Pre-order your copy of Steinberg Cubase 7 music production software now!

To find out more about Cubase 7, email audio@Jigsaw24.com, call 03332 409 306 or keep up with the latest audio news and offers on our Twitter (@Jigsaw24Audio) and Facebook page.

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 audio@Jigsaw24.com.

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.

Why do I need loudness metering?

“As TV broadcasters complete their transition to an all-digital delivery, content providers may soon have to embrace a new means for monitoring audio output in order to conform to acceptable loudness standards. Just as what the move from the analogue of tape to the digital of CD did for music, the move to digital TV broadcast has brought about a delivery system capable of much greater dynamic range. While that…” read more

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 audio@Jigsaw24.com or keep up with the latest audio news and offers on our Twitter (@Jigsaw24Audio) and Facebook page.

Loudness monitoring: PPMs are not enough

Loudness monitoring: PPMs are not enough

As TV broadcasters complete their transition to an all-digital delivery, content providers may soon have to embrace a new means for monitoring audio output in order to conform to acceptable loudness standards.

Just as what the move from the analogue of tape to the digital of CD did for music, the move to digital TV broadcast has brought about a delivery system capable of much greater dynamic range. While that should make for a better all-round experience, it presents a whole new problem for broadcasters who need to maintain consistent volumes between programmes and advertisements, to prevent annoying jumps in volume when a programme breaks to adverts.

The current UK guide suggests a maximum peak level during a programme of 6dB, and 4.5dB for adverts, and while there isn’t a disagreement about that ratio, there is an issue with the way it is measured. Peak metering only looks at the loudest part of the programme material, which may not be representative of the content as a whole. If the very loudest sounds occur right at the start, that will set the level for the entire programme, and the more dynamic range present in the audio, the greater the effect.


Adverts on the other hand, don’t have this problem. With a run time of just a matter of seconds, advert makers try to pack in as much visual and sonic impact. Dynamic range isn’t going to vary all that much in that time, so it’s already likely to sound proportionally louder. And with all the adverts jostling for attention, advert-makers will employ every trick available to make their product appear louder than their competitors. By utilising heavy compression and limiting (ideal, since dynamic range isn’t a concern), it is possible to massively increase the perceived volume of the sound in an advert while remaining within the decibel limit. This is exactly the technique employed in the music industry (arguably to its detriment) as both record labels and radio stations fight to produce louder-sounding music than their competitors. In the TV paradigm, while still remaining broadcast-legal and within the allotted limits, it’s possible to produce an advert which is subjectively much louder than the programme during which it appears. All of which makes for very annoying viewing.

TV broadcasters are aware of this, and they’re also aware of the risk of viewers changing channels just to avoid the adverts. The trouble is that it is an exploitation of the way the broadcast standard peak level metering works, which is not at all how human beings perceive volume. We don’t hear something as being loud based on just the loudest points, we form a perception of overall level based on the sonic content as a whole over time. We call this the ‘loudness’ and it’s easiest to think of it in terms of sonic density – by compressing a signal to a smaller dynamic range, we perceive an increase in density and thus it seems louder, even if the peak volume remains the same.


Loudness has largely been a subjective matter for years, but standards now exist to measure it, which means there are ways to monitor broadcast output based on how the viewer will hear it. The ITU BS.1770/1 standard gives broadcasters and content-makers a means to measure the loudness of their product in Loudness Units, a unit entirely separate from the traditional dB scale. As it looks (effectively) at ‘sonic density’ over a duration of time, a programme can provide metadata for its loudness value, which makes it possible for the broadcaster to standardise output. Tests on music have shown that by matching the loudness coefficient of older music against modern releases (which have a tendency to be mastered much louder), the listener perceives them to be the same volume, sometimes even reversing the trend and producing a better response to the music which has retained the greater dynamic range.

Although there are still competing standards, it’s inevitable that before long it will become a requirement for loudness to be taken into account. To that end, broadcast content providers will need to invest in new methods of monitoring that include loudness. Software currently exists from TC Electronic and Dolby to achieve this within the popular music production platforms such as Pro Tools and Nuendo, and TC Electronic have also released two models of standalone hardware loudness monitors with their TM7 and TM9 products, with the same software available for their System 6000 mainframe processor already widely used in broadcast mastering.

So while PPM meters may still have a role in calibration, it could well be that their time as the de facto standard of level monitoring is at an end. Broadcasters looking to invest in PPM meters definitely need to be aware of these changes if they want to stay competitive, so if this sounds like you, then please get in touch with one of Jigsaw’s broadcast consultants to chat about your options.

 You can call us on 03332 409 306 or email broadcast@Jigsaw24.com. You can also view our entire broadcast range on Jigsaw24.

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 audio@Jigsaw24.com 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 audio@Jigsaw24.com or leave us a comment below. You can also keep up with more news, reviews and offers on our Twitter (@Jigsaw24Audio) and Facebook page.


Are you listening to your music, or just your recording equipment?

Are you listening to your music, or just your recording equipment?

Over the weekend I received a message from a friend’s band that their new single had been released on iTunes. Ever the good supporter of the local scene, I of course checked it out immediately, and was left somewhat confused.

The best I could come up with by way of description was that it sounded expensive. I couldn’t tell you what the song was like musically because there was an enormous production in the way. The main thing I noticed was that this sounded a lot like a big label modern rock release but unfortunately very little like the band that I know really well, having seen – and even gigged with – them more than a dozen times.

I’m not so naive or idealistic to imply that all music needs to be an accurate representation of real instruments in real spaces. Creating the impossible is as much a joy of music production as it is with film making – a chance to create something captivating that could maybe never exist in the real world. I do wonder, though, if we have become so used to processing every signal that the notion of accurately capturing the sound of an instrument is in danger of becoming a lost art. It seems odd that at one end of the production chain most people are aware of the need to be able to monitor accurately. If your speakers flatten the sound, you won’t hear mistakes; if they add a ton of bass, you’ll be producing bass-light mixes. But at the other end of the chain, it doesn’t seem to be important. We get bombarded by products that actively colour the sound – mics that add ‘shimmer’ to vocals, preamps that add ‘warmth’, plug-ins to simulate the effects of tape or desk circuitry on the signal path.

The essential recording ethos

When I first started producing music, an engineer friend took me under his wing. Highly opinionated on many aspects of recording, he had one essential piece of equipment that I have adopted. It was a plastic chair, and whenever he had a session, the very first thing he would do was spend five minutes sat out in the live room with the band, listening to what they sounded like in the room. His recording ethos was always: no matter what the final result you’re trying to achieve, a good engineer should always be able to capture on record what they hear with their own ears. You can tweak to your heart’s content in order to achieve the desired result after that, but if you can achieve this starting point then you’re on to a good thing.

Furniture aside, I have my own favourite piece of recording kit – the AKG 414 mic, and for the same reasons. The first time I used one with a vocalist, I was astonished to hear the same voice I heard in the live room coming from my speakers. It dawned on me that I had become so used to the sound of a voice as it sounds through a microphone coming out of the speakers that I wasn’t listening for accuracy any longer. Not only that, I was even processing without listening, applying compression and EQ because that’s an accepted signal chain on a vocal, without ever considering whether it was necessary.

 Virtual processing plug-ins

One of the biggest advantages of modern DAW software is an almost limitless supply of virtual processing equipment in plug-in form. A traditional analogue studio might be able to afford a few choice pieces of outboard and would have to use them sparingly, but now we can strap 1176 limiters and Pultec EQs on as many channels as our computers can handle. The downside is that we now wield these tools almost indiscriminately, using a compressor when we could adjust a level, simply because it’s no longer a limited resource. And like a lot of users, I was reaching for presets rather than listening to the effects of the controls. The sound of processing was overshadowing the sound of my music.

I don’t doubt there will be people reading this and thinking “rookie mistake”. But I wonder – how much time do we spend investigating the other aspects of the signal chain? How many people compare multiple audio interfaces for conversion accuracy before making a purchase? On both input and output? When I switched from an ageing Digidesign 001 to an RME Fireface, I was amazed at how much I hadn’t been hearing on playback, let alone how much had been missing on capture. There are probably a few people starting to wonder now, I’m guessing. So let’s go a step further. How many engineers have listened to multiple DAW systems to see which software sounds the best? Does something recorded in Logic sound different to something recorded in Cubase? And, if so, which is more accurate?

At every step in the chain there is potential for the signal to be changed, whether it is by the sound of the converters or the algorithms used in recording software. And each step away further removes you from being able to recreate the original sound. In other words, even without processing the signal in any way, what goes in is no longer what comes out. Consider any piece of software which claims to be able to automatically time stretch or pitch shift your audio. In order for this to happen automatically, the software must be analysing for transients and computing a stretchable map. In other words, you’ll always be listening to processed audio and it is impossible to process digital audio and leave it unchanged. The same is true if you use a valve preamp to warm up a bright microphone. And what about any monitor controller you may have – are they altering the sound before it even reaches the speakers?

I would encourage anyone who is serious about recording to critically evaluate every link in the recording chain. Prism established the credibility of their converters, like the Orpheus interface, with world-class engineers comparing the sound they knew best of all – the full bandwidth sound of tape hiss! They considered their converters ready once no discernable difference could be detected by the best ears in the business.

For the rest of us, I’d suggest playing a CD you know really well through your speakers. Then play it through the converters of your audio interface and see if it sounds any different. Then record it into your DAW and play it back again. What about now? Are you getting out what you put in? If you’re hearing a difference somewhere along the chain, then maybe it’s time to consider an upgrade…

If you’re looking for the best in audio accuracy, give our audio specialists a call on 03332 409 306, or email audio@Jigsaw24.com. You can also visit our website to view our full product range.