A guide to different versions of Pro Tools

A guide to different versions of Pro Tools

Knowing your HD system from your M-Powered can easily baffle the Pro Tools newcomer. So we’ve put together the following guide to make things a little clearer, and help you decide on the right Pro Tools for you.

When Avid released Pro Tools 9 back in November, it marked a major departure from their traditional product model. Pro Tools 9 removed the requirement for Avid-branded audio hardware, meaning that users can now choose Pro Tools HD hardware, a Pro Tools LE interface such as an Mbox or 002/003 unit, or pretty much any other audio interface by a third-party manufacturer, depending on what your needs are.

The majority of users have welcomed this, calling it a huge step forward in terms of opening the platform up to a wider user base. But it has also created a fair bit of confusion – especially with the arrival of the HD Native platform. Before we get stuck into Pro Tools 9 though, let’s just briefly recap the Pro Tools 8 lineup so we can see how the new versions have improved.

Pro Tools 8 HD. This version required HD hardware to run an HD core system, such as HD1, HD2 or higher. Pro Tools 8 HD supported the TDM plug-in system that utilised DSP on the cards to run plug-ins and the mixer, as well as the native RTAS plug-in format. It also supported up to 192 audio tracks and 192 channels of I/O, all in full surround.

Pro Tools 8 LE. An all-native version of Pro Tools that required an LE interface to run. It shipped with Mbox, 002 and 003 products, offered up to 32 tracks of audio and native host-based mixing (just RTAS plug-ins, no TDM). With no surround mixing, Pro Tools 8 LE lacked the timecode ruler and full synchronisation options of Pro Tools HD.

Pro Tools M-Powered. With pretty much identical software specs to Pro Tools LE, this version worked with M-Audio interfaces instead.

On top of these, there are also three expansion options:

Music Production Toolkit. An expansion pack for Pro Tools LE and M-Powered which increased the track count to 48 tracks, added half a dozen plug-ins and unlocked the MP3 export option and Multitrack version of Beat Detective, previously only available in the HD version.

DV Toolkit. For Pro Tools LE only, this expansion unlocks a collection of post-production facilities otherwise only available in Pro Tools HD, including the timecode ruler, Digibase Pro and Digitranslator OMF import/export, plus increased 64-track count.

Complete Production Toolkit. A combination of Music Production Toolkit and DV Toolkit, this essentially unlocks every feature of Pro Tools HD that is otherwise missing from LE (with the exception of any processing that requires the DSP cards), track count is increased to 128, full synchronisation features are available as well as surround mixing up to 7.1.

Improvements in PT 9

Hopefully that’s gone some way in clarifying what each former Pro Tools product is and does. Now we’ll look at the latest additions to the family, Pro Tools 9 and Pro Tools HD 9

There are now just two Pro Tools flavours – Pro Tools 9 and Pro Tools HD 9. There is no LE 9, and no M-Powered 9. Pro Tools HD 9 is the version that you need if you are using an HD Core system or an HD Native Card, and Pro Tools 9 is the version for everyone else. Pro Tools 9 will work with any core audio or ASIO-compliant sound card, including Avid’s own 003 and Mbox ranges. It does not ship free with any interface, but is rather a full priced boxed copy of the software.

A fully standalone piece of software that can work with any audio interface, PT 9 is also a lot more fully featured than Pro Tools LE, essentially including all the features from DV Toolkit and Music Production Toolkit. It can handle up to 96 simultaneous tracks, sample rates up to 192kHz, 32 channels of I/O and 256 internal buses. The Digitranslator OMF import/export option and MP3 option are included, as is Beat Detective – the full multitrack option which now has full automatic delay compensation. Also new is native support for EuCon control surfaces.


As previously mentioned, all the features of the Music Production Toolkit and DV Toolkits are included in Pro Tools 9 and therefore no longer available to buy. However, the Complete Production Toolkit (now in its second version) is still available and, as before, unlocks the remaining HD-only features, so you get 7.1 surround mixing, VCA mixing, up to 192 tracks and advanced editing tools. Complete Production Toolkit 2 also includes support for Avid’s ICON control surfaces.

Pro Tools HD

HD-wise, there’s not so many new features in Pro when compared to the changes in the native version. The surround panner has been improved, and there is variable stereo panner depth available. MP3 and Digitranslator options are included (as they are with Pro Tools 9). Perhaps the most interesting new feature is that Pro Tools HD 9 will run in full HD mode if HD hardware is present, but if not then it will run in standard Pro Tools 9 mode. The Pro Tools HD 9 licence also includes a licence for the Complete Production Toolkit which is great for HD owners who also want a portable rig, as you can move sessions to the portable system and not lose surround capabilities, VCA faders or advanced automation modes. Lovely.

Getting the right system

HD Native systems require Pro Tools HD 9. Even though there’s no onboard DSP for mixing and plug-ins, HD Native is still an HD product and comes with the same full feature set. At the moment, there are still Mboxes and 003s available that have Pro Tools LE 8 in the box. When these have all gone, the interfaces will ship with just a driver disk. There will also be versions that are bundled with Pro Tools 9, but these will be more expensive.

If you own a Pro Tools HD system, then you just need a Pro Tools HD 9 update. If you own Pro Tools LE, you can buy a Pro Tools 9 crossgrade. You will also need an iLok if you don’t already own one. If you own M-Powered, there is an M-Powered crossgrade available.

For more information, there’s a very useful table over at the Avid website where you can compare all the products against each other. You can also call our audio team for more Pro Tools 9 or Pro Tools HD 9 advice on 03332 409 306, email audio@Jigsaw24.com or leave us a comment below and we’ll get back to you as soon as possible.

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 broadcast@Jigsaw24.com.

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 broadcast@Jigsaw24.com. Or have a look at the complete range of Symphony I/O products here.

Nuendo 5: A real alternative to Pro Tools for audio post-production

Nuendo 5: A real alternative to Pro Tools for audio post-production

I’ll confess it has been a while since I took a long hard look at Steinberg’s Nuendo. When the audio post-production package was first released in 2000, Steinberg were aiming squarely at the Pro Tools’ market share. They promised a real alternative for both professional recording facilities and post-production, and it was announced alongside Nuendo-branded interfaces from RME and Apogee.

Sadly, at the time, it failed to make the impact Steinberg had hoped, partly due to the perception that this was just Cubase with extra features, and also due to a lack of sync and controller options that could measure up to Avid’s offerings. Over the next few years (somewhat confusingly) Cubase and Nuendo continued to leapfrog each other with features. While the post-production community didn’t really jump on the Nuendo train, it gained a lot of uptake among sound design departments in computer games companies.

Fast forward ten years: the landscape has changed markedly and Steinberg have released Nuendo 5. Apple’s Final Cut Pro and the revitalised Adobe Premiere have carved a huge chunk out of Avid’s user-base, allowing smaller film production companies to compete successfully, without massive outlays on editing stations. Many of the larger TV production companies have folded, with their experienced engineers setting up shop for themselves.

Avid’s Pro Tools HD may still be the post-production platform of choice for the main studios, but Nuendo 5 may have just quietly come of age to offer a professional standard of audio post for the rest of us. Here’s an overview of why every video production company should take a good look at Nuendo for its audio post needs.

What is Nuendo? In short, Nuendo is an audio production environment which allows a near infinite number of audio tracks with full automated and clip-based mixing. It offers native surround support for formats from 5.1 right up to 10.2 and fully synchronised in-session video playback with a vast array of online and offline sound processing plug-ins for realtime mixing. Nuendo supports media exchange, audio up to 192kHz and full hands-on control from a variety of control surfaces.

Native and cross-platform. Like all high performance digital audio workstations, it requires a pretty hefty machine to run at its best, but you can run it on a Mac or Windows machine. In other words, you’re not bound to an audio production platform based on your preferred OS. No hardware restrictions mean you have your choice of audio interfaces and converters based on your needs.

Full interchange support. Nuendo can handle media exchange formats such as MXF, OMF, XML and also reads CMX 3600 EDL lists provided by the video editor as well as exported and imported CSV formatted spotting and ADR lists.

Full resolution video playout. Nuendo uses the QuickTime codec for video playback and can play out via FireWire or through Blackmagic or AJA cards.

Video follows edits. In edit mode, Nuendo will scrub the video with the edit, so you can trim from the beginning or end of a region. When performing moves or copies, the video will follow the cursor.

Centralised storage of sound libraries. Nuendo has an impressive feature called Media Bay – a fully searchable dynamic database that can be used to catalogue audio files. Media Bay can work with local storage and networked volumes meaning that sound effects databases can be stored centrally and accessed by multiple users. Its speed allows the user to search for sound files, store multiple searches, preview files and set regions for import.

Multiple marker tracks and cycle markers. Nuendo allows the creation of multiple marker and cycle regions so you can work easily on dialogue, foley or SFX. Uniquely, marker information can be imported via .csv files, meaning the video editor can use a spreadsheet program to provide all the notes with timing points. The sound editor can then import these directly into Nuendo where they will appear as markers complete with all the notes. Great for EDL and post conforms too.

Single-pass batch export of all stems. Once a project is finished, Nuendo allows you to export or bounce your project faster than realtime, using the full processor power of the computer. Not only can it bounce the full mix, it can simultaneously produce bounces of all your stems, so you can have a finished mix, your dialogue stem, audio stem, SFX stem etc. This is a huge advantage that Pro Tools users are longing for – no more bouncing in realtime. One bounce for every stem means you can output all of your audio files for a half hour TV show while you have a cup of tea, rather than it being a two or three hour job.

Clip-based editing. The most requested feature from users of older Avid audio systems is the ability to edit volumes of clips without using a realtime mixer. The person doing the mix for a show will typically not be the person editing the audio, so they don’t want to inherit a session chock-full of automation. Nuendo offers clip-based volume control so you can adjust levels without ever touching the mixer.

Comprehensive control room monitoring. Nuendo’s monitor matrix lets you choose to listen to the full mix or individual stems, busses or outputs at the click of a button.

Clip Packages. Nuendo allows you to group sound clips together and save them as clip packages so that complex sound effects made up of multiple sound files can be reused without having to be rendered first.

Jog and shuttle control for transport and editing. Deep integration with Euphonix controllers means that these can be used not just for mixing but also for editing via the jog wheel.

SystemLink and SyncStation. SystemLink allows multiple Nuendo stations to be harnessed for really large projects, with sample accuracy. Steinberg’s SyncStation allows for control of multiple devices including decks and consoles.

Although far from exhaustive, the list of features above shows that Steinberg are really paying close attention to the needs of the post-production community. But will Nuendo replace Pro Tools? It’s hard to say, with Pro Tools being so firmly entrenched in so many existing facilities. With the features on offer, those setting up new ventures will be hard pushed to justify the expense of an HD system over the cost of a Nuendo system – particularly with Nuendo’s ability to exchange files with Pro Tools.

If you would like more information or to arrange a demo, please call 03332 400 222 or email our team at broadcast@Jigsaw24.com.

Time to take a fresh look at Digital Mixers

Time to take a fresh look at Digital Mixers

The technology trends of the last few years would indicate that, for those of us that run digital recording studios, our love affair with digital mixers has hit a decidedly rocky patch.

At the dawn of the age where the computer began to reign supreme in the studio, the most popular setup by far centred around a Yamaha 01v or 02r digital mixer, connected via a Korg 1212 ADAT card to the computer. At the time it was pretty much the only viable non-Pro Tools computer-based setup. It provided 8 channels of bidirectional I/O, the analogue-to-digital conversion happened outside the computer to avoid noise, you could monitor directly from the desk (monitoring off a soundcard at that time resulted in about half a second of latency) and the desks themselves offered the automation, effects and processing that was either not available in software, or would result in your computer grinding to a halt under the weight of the algorithms.

‘In the box’

As computers gained faster processors, the dream of being able to do everything ‘in the box’ grew more tangible. Extensive mixing effects and automation became available within recording software packages, such as Logic and Cubase, which replaced much of the workload of a desk. Audio interfaces began to offer more simultaneous inputs and sprouted mic preamps too, replacing the desk’s I/O.  And when Mackie introduced the HUI, a way to keep those lovely moving faders for more-than-one-thing-at-a-time operation, we waved goodbye to the consoles.

The typical modern digital recording studio follows this desk-free paradigm, with a computer at its heart, an audio interface providing all the I/O, mic preamps and DSP-driven monitor mixing and a control surface to keep the whole thing hands-on. DSP units, such as Focusrite’s LiquidMix or TC’s Powercore, even offer some extra muscle for heavy duty effects processing if we want it.

Tascam mixers

If you think about it, what we have really done is break our digital mixer down into components, (faders, I/O and effects/processing) and then purchased them all again as individual units – and quite expensive components at that. And that’s where TASCAM have been clever.

Their DM3200 and DM4800 units are digital mixing desks, but TASCAM have thought to offer an optional FireWire card, which allows them to stream audio directly to and from the computer, 16 channels in either direction. The faders of the desk can be toggled between controlling the desk and functioning as a 16 fader control surface (24 on the DM4800) for your software by emulating Mackie Control and HUI protocols). And all of the channels of audio going to and from the computer can be processed using the EQ, dynamics and TC-derived effects within the desk.

Attractive package

So on one hand, the TASCAM is a digital desk, but you could also view it as a comprehensive audio interface with 16 very high quality mic preamps, built in 16- or 24-fader control surface and additional DSP processor. Considering you could spend more than the price of the DM3200 just buying a 16 fader control surface, for anyone aspiring to a digital recording studio, these digital consoles form a very attractive package.

Written by Rob Holsman in association with Ade Leader, Jigsaw24’s copywriter.

Want to find out more about the TASCAM digital mixers? Get in touch with us on 03332 400 222 or email broadcast@Jigsaw24.com.

Zero latency monitoring for Digidesign 002 and 003

Zero latency monitoring for Digidesign 002 and 003

Reading through the various Pro Tools forums, there are two commonly recurring themes.

Firstly, Pro Tools users love Pro Tools software for its reliability, flexibility, ease of use, precision editing and being the only audio platform that maintains sample accurate sync across all tracks in a session. Secondly, we’re not so enamoured of the LE hardware. The 002 and 003 ranges have been singled out for having no zero latency monitoring, audio converters below par when compared with other manufacturers and low-gain, noisy preamps.

Users are addressing the audio quality issues by either adding external converters from the likes of RME and Apogee to the digital I/O of their Digidesign interfaces or having companies such as Black Lion Audio modify their internal workings. However neither of these solutions address the latency issue  although we’ve found one add-on that does –  using an RME Fireface audio interface as an external AD/DA converter and monitor mixer.

Imagine you are a singer who is recording though a Digidesign 003. You have the mic in front of you and the backing track in your headphones. When you sing, you will hear your voice in your headphones, but only after it has gone down the firewire cable, been processed by Pro Tools, and sent back to the 003’s headphone sockets. All of this causes a noticeable delay, and it makes it really hard to deliver a performance.

The length of the delay depends upon the buffer size set in Pro Tools, the speed of the computer and whether or not there are any plug-ins being used. You can reduce the buffer size to minimum, but that delay will still be noticeable.

Other manufacturers produce audio interfaces which feature on-board routing that sends the mic signal directly to the headphones independently of it being sent to the recording software. However, Pro Tools LE can’t use anything other than a Digidesign interface, and only the small MBox products have this feature.

RME offer two models of Fireface – the 400 and the 800.  They differ in the number of channels and firewire connectivity, but  share a key feature – although they are audio interfaces they can function as standalone AD-DA converters. Because they are audio interfaces, they have headphone monitoring sockets, and because they are rather good audio interfaces, they offer true zero-latency monitoring.  RME’s secret weapon for this is their TotalMix software which not only allows you to create a custom headphone mix from all the available inputs via an onboard mixer, it also features a routing matrix which can send the signal from any input to any output(s). (Although, if we’re getting technical, TotalMix allows you to create custom mixes for each pair of stereo outputs, so multiple independent monitor mixes can be created.)

Here’s how to set it all up:

1.   Install the drivers for the Fireface and daisy chain it via firewire to the spare port on the 003.

2.   Connect the ADAT In of the 003 to ADAT Out of the Fireface via lightpipe and vice versa.

3.   Open the control panel for the Fireface, switch to the matrix view and check the appropriate boxes so that input 1 is routed to ADAT out 1, input 2 to ADAT output 2 and so on.

4.   Route the returning signal the same way, so ADAT input 1 goes to analogue out 1, ADAT input 2 goes to analogue out 2 etc Doing this will allow the fireface to function as an AD and DA converter.

5.   Check the appropriate boxes so that analogue inputs 1-8 are also sent to both Phones L and Phones R. Now you’ve got a headphone mix of incoming signals.

6.   Monitor speakers can now be connected to the RME’s analogue outputs 1 and 2

7.   Change the I/O setup of Pro Tools so that your main output path is now ADAT 1-2. Your mix will be sent out of ADAT 1-2 and the Fireface routing matrix will send it to the speakers.

8.   Now, whatever you plug into any input of the RME will be recorded via the corresponding ADAT input of the 003. For example you can plug a mic into mic preamp 2 of the RME and you’ll be recording it in Pro Tools via ADAT In 2. If it helps, you can always rename the inputs in the Pro Tools I/O setup page to make it clear and even disable the 003 analogue inputs if you’re not going to be using them.

9.   Go to Pro Tools Preferences > Operation and uncheck the “Link Record and Play Faders”. When you’re recording from an input on the RME, you’ll hear your source twice – once in realtime through the RME and as a delayed signal through Pro Tools. This will get really disconcerting unless you mute the volume of the track you’re recording onto. Unlinking Record and Play Faders means you can set the faders of all tracks to zero when they are record armed and they’ll return to normal when you’re not recording. Pro Tools will remember this state too, so whenever you go into record they’ll re-mute. Neat, huh?

So to summarise, what you gain is:

–   Better quality AD and DA conversion. If you want an example, just import some audio into Pro Tools and compare the difference between playing it back through the 003s own analogue outputs and via the RME outputs via firewire. You will hear a wider more detailed stereo image, and a greater frequency range too.

–   Zero Latency monitoring. At least from the RME’s inputs.

–   Better mic preamps. The RME’s digitally controlled mic preamps offer more gain, lower noise floor and more headroom than those in the unit.

What you are not doing is replacing the Digidesign 002 or 003, or using the Fireface as an interface that works with Pro Tools. Despite being an audio interface, we are using the Fireface as a standalone AD/DA converter to augment the inputs.

Happy tracking!

Written by Rob Holsman in association with Ade Leader, Jigsaw’s copywriter.

For more information call our audio team on 03332 400 222 or email broadcast@Jigsaw24.com.

Akai APC40 vs. Novation Launchpad – Review

Akai APC40 vs. Novation Launchpad – Review

Maybe it’s a sign of my impending decline into senility, but I still think of Ableton Live as a newcomer to the audio sequencer application party. Despite it celebrating its ten-year anniversary, I have managed to completely ignore it until a few months ago when I had to sequence some pre-recorded parts at a live gig – I instantly fell in love with the software. So, it has been with some fervour that I have been investigating the two controllers that are currently on the market for Live: Akai’s APC40 (released back in the early summer) and Novation’s Launchpad (which arrived in October).

It would be unfair of me to draw a direct comparison between these two, as it’s clear that they appeal to different markets. There’s also quite a difference in price; the Akai typically costs around £379 incVAT, and the Launchpad comes in at a more streamlined £149. Instead, I’ll look at what each of these offers the Live user.

The Akai APC40
The Akai APC40 is an extremely rugged unit with a 430mm x 335mm metal chassis and a generous collection of controls. The clip launch grid is where most of the action happens; it represents tracks 1-8 horizontally and clips 1-5 for eachtrack vertically (making 40 clip controllers). If a clip is playing, its associated pad is green. Red means it is recording, while amber means a clip is present but not playing, and an unlit pad shows an empty slot. There are buttons to trigger an entire row of clips too. If you have more than eight tracks or more than five rows of clips, the SHIFT button allows you to bank around either across or down to access all your clips. Record Arm and Solo controls for each track act as you’d expect, and the Activation buttons for each track show which ones are not muted – rather like a mute button on a mixing console, but in reverse.

The rotary controls have illuminated outer rings to show current values, and one bank of eight is available for adjusting sends or pans for the eight tracks currently present on the clip pads – these shift with the pads when banking. The other eight rotaries allow you to directly access parameters on the currently selected device, and there are eight banks of possible controls, meaning you can access up to 64 parameters per device. In addition to the rotaries, there are eight buttons dedicated to the following Ableton functions: CLIP/TRACK view toggle; DEVICE On/Off; Previous and Next device selection buttons; DETAIL VIEW On/Off; REC QUANTIZATION On/Off; MIDI OVERDUB On/Off; and METRONOME On/Off.

Lastly, the crossfader works as you would expect, fading between whatever has been defined as crossfader assignments A and B in the software. There’s also a Cue Level control, which deals with the volume sent to the Cue Output, eight 45mm faders and a tap tempo button.

The Novation Launchpad
Novation’s Launchpad is a compact and lightweight unit, measuring just 240mm square and made of moulded plastic. It features an 8×8 grid of touch sensitive illuminating pads, which function and illuminate in exactly the same way as the Akai’s (showing clips as ready, recording, playing or empty). There are also scene launch buttons to trigger collections of clips together.

Although there are no faders or rotaries, Novation have equipped the Launchpad with a mixer mode that allows the pads to illustrate or control, and pan and send levels. The pads light up to give a bar graph representation of the mixer values and can be touched to change levels and values. Selecting different modes is quick and easy, and happens via the various scene launch buttons; multiple launchpads can be used together to expand controllability.

As I said earlier…
The two units are clearly aimed at different markets, so a head-to-head comparison is unfair. The Novation Launchpad offers easy access to the basic functionality of Live in a small footprint; it’s ideal for someone building up tracks, who doesn’t mind using the mouse and keyboard. It does solve the main issue of being able to cue up and launch multiple clips at once, which is the biggest challenge facing Live users and, although you can assign clips to keys of any MIDI keyboard, the illuminating buttons of either unit provide essential feedback. But, to anyone using Live as a performance instrument or their main software, it feels like Novation have left out too many features to be a serious contender.

The APC40 has been designed for the Ableton user who wants maximum interaction with the controller, and minimum reliance on a mouse. Both units use the same illuminating pad topology, but the APC’s rotary controls and faders give a precise level of control – the Launchpad gives a choice of only eight values when using the pads for pan or level control. For anyone wanting to play in realtime with the values of Ableton’s devices, such as tweaking filter resonance and cut-off (who wouldn’t!), the second bank of rotaries on the APC is great and means no mode switching if you also want to play with pan controls. There is simply no way of achieving this with the Launchpad, although it can work with other products in Novation’s range (such as the Nocturn) to deliver the crossfader functionality.

The strangest omission from the Launchpad is that there is no tap tempo button. You can, however, easily get Live to learn the function from any button; I used USER 1, which worked perfectly. But the ability to jiggle tempos is such a fundamental feature of the Ableton Live software, it seems almost incredible that any dedicated Live controller doesn’t have a button for tap tempo. The APC40 sports not only a dedicated tap button but also buttons to nudge the tempo up and down, which is perfect if you’re beat matching records.

Ableton Live has been crying out for a dedicated controller since it first arrived. The use of a controller leverages far more functionality out of the software than you can achieve with a mouse. Novation and Akai have each produced very able controllers that will appeal to different types of users – based largely on how much you intend to rely on a sole controller to do all your functions or whether you are happy to use a mix of additional controls and the occasional mouse interaction. But which ever you use, adding a controller will give your Ableton experience a new lease of life.

To find out more, get in touch with the Broadcast team on 03332 400 222 or email broadcast@jigsaw24.com.


Sound for Picture V: Surround Sound

Sound for Picture V: Surround Sound

In previous location recording articles, we looked at why transparent audio is perhaps the most important aspect of any video production. This culminates during the post-production stage with the addition of realistic ambient sound that captivates the audience and reduces, or at best removes, awareness of their real-world surroundings.

As many of us have become accustomed to over the last decade, most feature films and indeed some television programming comes with more than just left and right audio channels. Surround sound mixing makes use of multiple audio channels to envelop the audience, making them feel as though they’re in the middle of the action. By placing the audience in the middle of the soundscape and enabling them to hear sounds coming from all around them, film makers are able to maximise ‘suspended disbelief’ – the ultimate goal of any sound track.

A basic surround sound system will comprise at least six speakers that literally surround the audience: centre (C), left (L), right (R), surround left (SL), surround right (SR) and low frequency effect (LFE). This simple setup is better known as a 5.1 system – 5 speakers and 1 bass unit. The LFE channel (the .1) came to be known as such because it typically handles 1/10 of the frequency range of the other speakers.

5.1 Surround Sound speaker placement

Further speakers, specifically rear centre fill, can be added to create a 6.1 system:

6.1 Surround Sound speaker placement

And two additional speakers can be added to a 5.1 system, either centre left (CL) and centre right (CR) or left surround (LS) and right surround (RS) to create a 7.1 system:


7.1 Surround Sound speaker placement (widescreen format on left)

Whilst there are virtual surround algorithms (such as the Sound Retrieval System, or ‘SRS’) that make use of two speakers and psycho-acoustic phase effects to emulate surround sound, we’ll focus in this article on the true surround sound formats.

Dolby Digital™

Dolby Digital, formally known as AC-3 (short for audio coding 3) is the

standard surround format for our home cinema systems. As you will undoubtedly have noticed, Dolby Digital is the standard format for DVD video. In fact, the scope of this format has gone beyond DVD alone and is now also part of the High Definition TV (HDTV) standard.

The Dolby Digital format is capable of providing up to 5 discrete channels of full frequency effects (from 20Hz up to 20kHz) plus an additional sixth channel dedicated to low frequencies (20Hz to 120Hz). It’s important to be aware that not all Dolby Digital systems will have 5.1 channels of audio, and those that do will be designated as such – e.g. ‘Dolby Digital 5.1′. In fact, Dolby Digital can have as few as one channel of audio (mono) designated as Dolby Digital 1.0. Dolby Digital is encoded on the film release print and must be licensed from Dolby Labs for a fee.

DTS Digital Surround™

DTS Digital Surround (or simply ‘DTS’) is a competing alternative to Dolby Digital. DTS, like its competitor, is a 5.1 surround sound format that is available in cinemas and as an optional soundtrack on some DVD movies for home viewing. Unlike Dolby Digital, DTS is not a standard soundtrack format for either commercial DVD releases or the HDTV format used in digital television broadcasting.

As you would expect from a competing format, there are a few pros and cons to consider when opting for DTS over Dolby Digital. Firstly, DTS uses higher data rates than Dolby Digital and therefore some argue that there is an audible improvement in sound quality. This higher data rate, however, requires more storage and, as such, more space is needed on the DVD to accommodate DTS than Dolby Digital.

Dolby Surround Pro-Logic™

Dolby Surround Pro-Logic first made an appearance in home cinema systems in the early 1990s and is still the standard for analogue television broadcasts since the Dolby Surround Pro-Logic signal can be encoded in a stereo analogue signal. As all DVD players down-mix the Dolby Digital information to the Dolby Surround Pro-Logic format and output the signal as a stereo pair, you can still watch DVD movies on an older Dolby Surround Pro-Logic receiver.

Dolby Digital EX™, THX Surround EX™ and DTS Extended Surround (DTS-ES)™


Dolby Digital EX isan extension of Dolby Digital 5.1 to include a surround back channel, whosecorresponding speaker is placed directly behind the audience. This allows audioevents to occur behind the audience, further enhancing the 360˚ surround sound and enveloping the audience to a greater extent. The THX Surround EX format is a joint venture by Lucasfilm THX and Dolby Laboratories and is the home cinema version of Dolby Digital EX. While Lucasfilm THX licenses the THX Surround EX format for use on receivers and pre-amplifiers, Dolby Laboratories currently license THX Surround EX under its own name (Dolby Digital EX) for consumer home cinema systems. The method by which Dolby Digital EX and DTS-ES encode the surround back channel is known as matrix encoding since the back channel is encoded and later decoded from the surround left and surround right channels. The surround back channel information is encoded into the surround left and surround right channels and, for this reason, is sometimes referred to as Dolby Digital 5.1 EX or DTS 5.1 ES. As this surround back channel is not a discrete channel, the system is still technically a 5.1 system.


One misconception we need to banish about THX is that it’s a surround sound format. THX is essentially a measure of quality that can be found in conjunction with different surround sound formats such as Dolby Digital and DTS. THX standards are a set of criteria that dubbing stages and movie theatres adhere to in order to guarantee that what the audience hears during playback is as close to what the mixing engineer heard during mixdown as possible. In theory, any production mastered to the THX standard should sound the same in any THX qualified cinema in the world. As such, THX isn’t really a format of surround sound, it’s a measure of quality that brings a standardisation to the film-making community.

Stay tuned to Jigsaw Broadcast for my forthcoming feature on audio post-production systems. If you have any queries about anything you have read, get in touch with the Broadcast team on 03332 400 222 or take a look at our full broadcast range.

Sound for Picture III: Location Recording

Sound for Picture III: Location Recording

For video producers who have limited experience in professional audio recording, choosing the right microphone in a given situation can often seem like a minefield. Not only do microphones come in a variety of shapes and sizes, they will also be grouped according to their physical build characteristics and pickup pattern.

Luckily, there are tried and tested methods when it comes to location recording that can make the decision-making process a lot easier. For most location recording applications, your decision will be based around different types of shotgun and lavalier microphones, each of which exhibit different recording characteristics.

Shotgun Microphones

Shotgun microphones, so named because their long narrow tube resembles the barrel of a shotgun, come in two common types: “long shotgun” and “short shotgun”. As a rule, a long shotgun will have a narrower angle of acceptance than a short shotgun – it will reject more off-axis sound from the side of the microphone. Short shotguns have a wider angle of acceptance and, as such, will pick up more off-axis sound and will not isolate the talent or sound source as well as a long shotgun will.

It is important to note that while shotgun mics are highly directional, they have small lobes of sensitivity to the left, right and rear and will therefore pick up sound from behind the microphone barrel. For this reason, shotgun mics are almost exclusively mounted on boom poles so that they can be directed at the talent from above, depending on framing, and this yields superior isolation of individual sources and reasonable separation from unwanted background noise.

Perhaps the greatest misunderstanding with regard to shotgun mics is that they are able to pick up sound from a greater distance than other types of microphone. While shotgun mics are extremely directional, the further they are from the source of the sound, the more likely they are to pick up unwanted ambient noise. It is the distance between the source and the mic that will have the most dramatic effect on recording quality regardless of the type, cost or quality of the mic.

Jigsaw Recommends: Sennheiser MKH-416, RODE NTG-2, RODE VideoMic, Neumann KMR 82i.

Microphone Mounts and Booms

There comes a point in time when mounting a microphone on a camera becomes impractical and doesn’t allow the audio to be captured to its full potential. Moreover, any directional microphone will only capture predominantly in the direction that the camera is pointing, which inevitably makes a camera-mounted microphone a supplementary feature, for example, to capture guide audio.

Presuming that you are working as part of a production team as opposed to single-person shoots, it would seem nonsensical to spend money on a professional level microphone such as the Sennheiser MKH-416 or Neumann KMR 82i and then not spend the little extra on boom poles and suitable mounts for boom/handheld use. The most effective microphone mounting units utilise suspension systems that acoustically separate the microphone from hardware that is susceptible to transmission of sound waves or generating unwanted noise through movement. Any sound arriving at the microphone capsule will also arrive at any solid object such as a mount or camera and this sound wave will propagate through the medium causing unwanted vibration of that object. By using a suspension mount, these extraneous vibrations are minimised and the recorded sound is less likely to be influenced by external factors. It’s also true that when microphones are handheld, the opportunity for additional noise is at premium – even the slightest movement or accidental touch of the boom pole, especially when the mics are mobile, can transmit to the microphone and thus the recorded audio. In these situations, suspension mounts are invaluable in preserving the transparency and accuracy of your recorded audio.

Once you’ve selected your microphone and mount, and assuming you won’t be attaching it to the camera with a hot shoe adapter, you’ll need something to attach it to. To the casual observer, boom poles and hand grips are very much alike with little difference to choose from. For the majority of uses this is true, although there are options available depending on the intended use of the equipment. For example, extendable booms allow for greater flexibility of mic placement and enable the mic to follow the sound source more precisely. For extended scenes, balanced booms that use lightweight aluminium construction are particularly appreciated by the audio crew!

Windshields, Windsocks and The Duck

Recording outdoors in the UK can be a wonderful experience if you manage to schedule a shoot during the summer (for reference, this will last for about two days and can fall in any month between April and October). If you understandably miss the summer weather, it can generally be a bit windy. Or rainy. Normally, it will be both. Luckily, your audio doesn’t need to suffer because of prevailing weather.

One of the most useful tools to location recorders is the windshield, commonly known as a zeppelin due to the similarities in its appearance to… well, a zeppelin. Windshields will be used in conjunction with a boom pole or handheld mount and will be designed to fit over microphones of a certain length – although you could house a short shotgun mic in a windshield designed for a long shotgun mic, the best results are obtained when the windshield matches the specifications of the mic correctly. The primary function of the windshield is to filter out extraneous wind noise and air movement from indoor sources such as air conditioning units, but it is also tasked with protecting the microphone to some extent.

For challenging conditions, a windshield may need to be fitted with a windsock such as the Windjammer by Rycote. Windsocks are typically synthetic fur-covered sleeves that will be attached to a windshield using Velcro or a zip. While these are excellent for filtering out wind noise, it’s worth being aware that they will also filter out some high frequency content from your audio source before it reaches the microphone capsule. With this in mind, it’s important not to use a full windshield and windsock system all of the time, although some windsocks have been designed with a much shorter fur that consequently attenuates high frequencies to a lesser degree at the expense of protection from stronger winds. For static outside broadcasts, additional protection may be required to shield the microphone and windshield from rain. Rycote, for example, offer a waterproof roof for their Modular Windshield and S-Series system called The Duck. As well as protecting the mic and windsock, solutions such as The Duck also minimise rain noise and further improve the quality of the recorded audio.

As a compromise, many companies now offer intermediate systems like Rycote’s Softie Windshield. Instead of being a fully encapsulating enclosure for a microphone, these intermediate windshields slip over the capsule of the mic quickly and easily. This ease of setup along with the practicality of the smaller size of the unit allow them to be transported and put into action quickly and efficiently, making them particularly useful in news-gathering situations.

Wireless Systems and Lavalier Microphones

Despite wireless systems and lavalier microphones being inherently separate technologies, they are most commonly used together to provide a mobile and covert recording solution. With the exception of adding enhanced functionality to portable recorders like the M-Audio Microtrack (which we’ll look at shortly), lavalier mics – when used in combination with wireless transmitters and receivers – are excellent for spot-mic’ing and for maintaining constant and consistent close-proximity audio signals from talent. Lavalier microphones are particularly suited to this latter situation due to their small size, which enables them to be hidden within an actor’s clothing or in and around a set to accurately capture nuanced sound from a performance.

In my experience, the understanding of wireless systems is polarised between end users, some of whom understand the benefits and limitations while others understand that they require this technology to get a job done, but have little experience in its use and operation. The main question that comes up (largely, I suspect, as a result of budget constraints) is whether one receiver can accept the signal of more than one transmitter. The answer, unfortunately, is no. Well, strictly speaking you can, but the results will be largely unusable. With that said, there are a number of units available that include twin receivers and modular systems for multi-channel setups that also reduce the amount of rack space required for large systems.

Should you need to, however, the output of one transmitter can be received by multiple receivers. This means that the output from one transmitter can be sent to multiple receivers on multiple cameras for larger productions.

One thing worthy of mention is that wireless systems, even those operating in the less cluttered UHF range, are susceptible to interference. Wherever possible, wired audio connections are preferable to preserve signal fidelity, although the benefit of having wireless transmission of audio can often outweigh the slight degradation in audio quality. The location you record in will also have a bearing on the quality of the signal between the transmitter and receiver – on one occasion I remember arriving at a recording with a myriad of wireless systems only to find out that the proximity of the location to the local airport meant that we effectively ended up in a wireless ‘shadow’ due to the airport’s communication systems. In certain situations like this, having a wired backup is always a good idea!

Want to know more? Call 03332 400 222, email broadcast@Jigsaw24.com or take a look at our full broadcast range.


Sound for Picture I: A Brief History…

Sound for Picture I: A Brief History…

The idea of combining recording sound with picture is nearly as old as film itself, although restraints imposed by technology made this possible only in the 1920s with the introduction of the Vitaphone system. Used on features and nearly 2,000 short subjects by Warner Bros. and its sister studio First National from 1926 to 1930, Vitaphone was the most successful of the sound-on-disc processes.

Rather than the picture and audio co-existing on the same film, the soundtrack was issued separately on 16″ phonograph records. Using projectors, amplifiers and speakers enabled simultaneous playback of both audio and picture – this was achieved by aligning a start mark on the film with the picture gate whilst at the same time placing a phonograph record on a turntable and aligning the phonograph needle with an arrow scribed on the record’s label. When the projector rolled, the phonograph would turn at a fixed rate of 33 1/3 r.p.m. to match the 11-minute maximum playing time of a reel of film and, in theory, picture and audio would play back in sync. Unlike most phonograph discs, the needle on Vitaphone records would move from the inside of the disc to the outside.

Towards the end of the 1920s, the studio system further strengthened the relationship between picture and audio and saw the Radio Corporation of America (RCA) move into the movie business in an effort to exploit the RCA Photophone – a technology owned by RCA’s parent company, General Electric. The Photophone was a sound-on-film “variable-area” film exposure system that allowed electrically-recorded audio to be synchronised to a motion picture image.

While the “variable-area” format saw competition in the form of “variable-density” systems, neither was perceived to be significantly superior and indeed both systems were marketed equally by RCA and another company, Western Electric. Ultimately, the RCA systems were abandoned and the Western Electric (Westrex) system was renamed Photophone after the Western Electric and Westrex registered trademarks were sold for use in cinema sound systems. Renaming of the Westrex system to Photophone was facilitated by the demise of RCA’s cinema sound business unit and by General Electric’s failure to secure the Photophone trademark. Although technology has now moved on considerably, the Photophone system is still in production, with more than 100 systems currently in active service world-wide.

Of course, sound for picture has become inherently more efficient and simultaneously more complex with the advancement of digital tools for audio creation. With large post-production facilities at hand and with the development of non-destructive editing, producers have far more options and tools available to them to create some phenomenal soundscapes for picture. Arguably one of the most creative professions for the audio engineer, audio post-production requires a wide range of skills and often involves teams specialising in various stages of production, from initial audio capture to post-production Foley and ADR recording to editing and, finally, mixing and mastering to any one of many digital multi-channel formats.

Want to know more? Call us on 03332 400 222, email broadcast@Jigsaw24.com or take a look at our full broadcast range.