In This Issue
May 2004 In This Issue Click article title to open Reviews
People
BIAS Peak 4
Business End
Stereo Editor: Mac OS X With the move to version 4, BIAS's aptly-named Peak editing package leaves Mac OS9 behind, but offers a wealth of new features: built-in CD burning, a new user interface, Audio Units support, and processing tools including a real-time convolving reverb.
Readers' Recordings More constructive comments from MPG (Music Producers Guild) members on SOS readers' submitted recordings.
Boss BR1600CD 16-track Digital Multitracker Boss\'s range of digital multitrackers is designed to allow musicians to make recordings with a minimum of technical knowledge. The BR1600 is the most sophisticated yet, providing 16-track recording along with amp modelling, vocal harmony generation, and built-in drum machine and loop sequencer.
Emagic Logic Pro & Express Recording Software For Mac OS 9/OS X The big news for Logic users is not the new features in Logic Pro and Express, but the new (and extremely competitive) pricing. So which of the two is the bundle for you?
Eventide Reverb 2016 Digital Reverb Processor Eventide's classic SP2016 reverb unit has been recreated and enhanced by the original designer. But how does the sound match up to high-spec multi-effects units of today?
Educate Your Ears Leader If you've never heard top-quality recording gear with your own ears, what reference do you have when assessing midprice and budget gear?
Emulating Brian May's Guitar Amp Sound TC Electronic's Stefan Moeller For many years, German designer Stefan Moeller has been on a quest to perfectly emulate Brian May's guitar amp sound. Now, with TC's new TC Thirty plug-in for Powercore, he believes he's cracked it...
Graham Fellows (aka John Shuttleworth) Making Engineering Funny Comic actor, has been recording and editing his own radio series for years — and now he's putting his engineering skills centre stage.
Hardware design paradigms: why use them? Sounding Off When studios are increasingly dominated by software, why are we still hanging on to hardware design paradigms?
Gmedia ImpOSCar
Matmos
Virtual Instrument Plug-in: Mac & PC Gmedia continue their quest to render the best of the world's keyboards in software form. But will it give OSCar fans a case of the grouches?
The Art Of Cut & Paste With an emphasis on cut-and-paste sound collages and bizarre field recordings, San Franciscan duo Matmos have created some unique and often disturbing electronic music.
Korg Triton Extreme
Studio SOS
Synth Workstation Korg have redefined the workstation synth many times, and each time it has become harder to see how they could better their previous achievements. The Triton Extreme is a bold new colour, but does the rest of it measure up?
Jiang Li The SOS team apply themselves to the task of mixing Chinese traditional instruments with mainstream Western sounds at reader Jiang Li's home studio.
Kurzweil K2661 Synth Workstation Kurzweil's high-quality K-series workstations have always
Todd Rundgren Recording Liars As a musician, engineer, producer and multimedia artist, Todd Rundgren has been a pioneering figure in rock music
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In This Issue
commanded a premium price tag in the UK, but it's been a while since the range's underlying technology has been updated. Is their latest offering still competitive in 2004?
Latest Sample CDs Sample Shop Checkout the hottest Sample CDs...
M Audio Firewire Audiophile Firewire Audio & MIDI Interface Until recently, the benefits of the high-bandwidth IEEE 1394 interface were available only to musicians on a relatively generous budget, but now M Audio are making Firewire peripherals available at new low prices to PC and Mac users.
Mackie HR626 Active Studio Monitors Mackie continue to expand their range of studio monitors by combining an innovative D'Appolito design with their existing passive radiator technology.
Presonus Eureka Recording Channel This smart new unit offers EQ and compression, plus a transformer-coupled Class-A preamp and an unusual Saturation control.
Preview: Korg Legacy Collection Virtual Instrument Suite/Hardware Controller Korg's Legacy Collection recreates some of their bestloved keyboards in software form. It wasn't finished as we went to press at the end of March, but we managed to grab this sneak preview as the final touches were being applied... (Full in-depth hands-on Review next month.)
Rode NT2000 & NT1A
since the late 1960s. His first album of new material for 10 years was, like many of its groundbreaking predecessors, almost entirely played, produced and recorded by Rundgren alone.
Your Letters Crosstalk We respond to another batch of reader emails and letters. Technique
Automatic Beat-slicing Logic Notes Although beat-slicing is something more immediately associated with programs like Propellerhead Recycle or Bitshift Audio Phatmatik Pro, Logic also has facilities to automatically chop up your loops.
CLASSIC TRACKS: 'What's Love Got To Do With It?' Artist: Tina Turner; Producer: Terry Britten; Engineer: John Hudson In 1984, a dose of British soul resurrected Tina Turner's flagging career in spectacular style. For engineer John Hudson, the recording of 'What's Love Got To Do With It?' also provided a memorable example of the 'less is more' principle in action...
Consoles: taking control with Digital Performer Digital Performer Notes If you've never investigated DP's Consoles feature, you may be missing out on interesting MIDI and Audio control possibilities. We explain, as well as introducing a couple of indispensable new utilities.
Controlling Pro Tools
Capacitor Microphones These two new large-diaphragm mics from Rode improve on their previous technology, offering a classy sound with exceptionally low noise.
Pro Tools Notes You don't need a Pro Control to get some hands-on control of Pro Tools. You don't even need one of the 'supported' MIDI control surfaces: pretty much anything with knobs on will do. Read on...
Tascam 2488
Creative Synthesis With Delays
24-track Digital Portastudio Tascam's new Digital Portastudio is the most affordable 24track workstation on the market. In our exclusive hands-on review, we find out how easy it is to use, and whether its recording quality is as impressive as its track count.
Synth Secrets Effects can play just as important a role in sound creation as the elements in a synth's signal path — provided you have access to their constituent parts. We take a closer look at effects synthesis using simple delays.
Waves IR1
Creative Synthesis With Yamaha XG
Convolution Reverb Plug-In: Mac & PC If you want reverb 'sampled' from real spaces, there are now quite a few options available. But Waves have managed to come up with something a little special — a convolving reverb that offers detailed control over the
XG Masterclass: Part 2 Layering voices is one of the best ways to maximise the potential of your XG sound module, so here are a variety of ingenious ways you can use this technique — you can even turn your synth into a high-spec step sequencer!
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In This Issue
sound.
Competition
WIN Apple eMac computer recording system Sound Advice
Q. Are there different types of MP3? Q. How should I set my compressor for recording rock guitar? Q. Should I build a vocal booth? Q. What can I do to improve my stereo recording setup? Q. What should I look for in a PC laptop? Q. Where should I put my monitors?
Cubase SX 2: Using Plug-ins in a Surround Mix Cubase Notes Continuing our exploration of surround sound in Cubase SX 2, this month we turn to using stereo and multi-channel plugins in a surround mix.
Demo Doctor Reader Recordings Another batch of lucky SOS readers' demos wait nervously for the Doctor's prognosis. What is the verdict? Listen while you read on...
Hard Drive Size Cutting Edge Hard drives are getting bigger and that can only be good, right? Well, up to a point...
Loading REX Files... and more tips Sonar Notes Find out how to load REX files into Sonar, discover which digital audio editors work best with loops, and explore some useful envelope techniques.
OS X Kernel Panics Apple Notes If you're suffering from kernel panics, if no one else can help, and if you can find a log file, maybe you should hire (well, read, at least) this month's Apple Notes...
PC Systems: What Do You Get For Your Money? PC Musician Just what can you expect in a PC system costing £600? Or over £1500? We round up the likely specs and the possible pitfalls.
Questions & Answers Solutions to Reader Problems More valuable advice to help you solve common problems.
Super-quiet Power Supplies PC Notes We take a look at a new range of super-quiet power supplies, as well as explaining how to protect your data and settings in case of crashes.
Using Mics & D.I. Boxes On Stage Live Sound Find out how to mike up a typical live band — and also discover when it\'s best to Direct Inject instead.
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BIAS Peak 4
In this article:
What's New? Peak At Play Working With Peak Peak For Sound Design Editing Regions Late News Peak Performance
BIAS Peak 4 Stereo Editor: Mac OS X Published in SOS May 2004 Print article : Close window
Reviews : Software
BIAS Peak 4 £349 pros Improved user interface. Very flexible but still largely intuitive. Comprehensive set of DSP tools. Supports Audio Units and VST plug-ins.
With the move to version 4, BIAS's Peak editing package leaves Mac OS 9 behind, but offers a wealth of new features such as built-in CD burning, a new interface, Audio Units support, and processing tools including a real-time convolving reverb. Paul White
cons Nudge editor would still be better if it covered three regions.
summary Peak gets better with every revision, and version 4 is bristling with useful features, both corrective and creative, yet remains easy to use. It also comes with a well written PDF manual, with a paperback version optional at $25 US.
information £349; upgrade from version 3 £130; Peak LE £89; Peak DV £179; printed manual $25. Prices include VAT. SCV London +44 (0)20 8418 0778. +44 (0)20 8418 0624. Click here to email www.scvlondon.co.uk www.bias-inc.com
Test Spec Peak versions 4.03 and 4.1.
BIAS Peak is arguably the bestestablished stereo audio editor for the Mac, and has now reached version 4, with a new look and new features including a convolving reverb. Version 4 onwards supports only Mac OS X, but for any OS 9 diehards out there, the installation CD also includes version 3 for OS 9. Peak 4 requires Mac OS 10.2 or later, and though you can run it on a basic G3, a G4 or G5 is needed if you want to use the convolution process in real time. Upwards of half a gigabyte of RAM is recommended, and your audio interface must support Apple's Core Audio standard. Peak makes effective use of dual-processor G4 and G5 systems. Peak has to be authorised via the BIAS web site, otherwise it will fall asleep after 14 days. You must provide your serial number and other details in order to gain an authorisation code that unlocks the program permanently. Most of the features that have been carried over from earlier versions of Peak were explored in more depth in our review of version 3, so I won't dwell on them in any depth — to find out more, you can read that review in the January 2003 issue, or on-line at www.soundonsound.com/sos/jan03/ articles/biaspeak.asp. As before, Peak's main claim to fame is that it offers more sophisticated stereo
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BIAS Peak 4
Apple iBook G4 933MHz with 384MB RAM, running Mac OS 10.3.2, and Apple G5 dual 2GHz with 2GB RAM running OS 10.2.7.
editing facilities than most multitrack sequencer packages. It's also very much a 'Swiss Army knife' program, as it can handle and convert a number of commonly used audio file formats, including AIFF, WAV and SDII (including SDII Regions), and it has an impressive armoury of DSP tools and effects. Audio files can be accommodated up to 32-bit and up to whatever sample rate your hardware can support. All editing tasks are non-destructive until a project is saved, and file conversion and other processes can be organised as batches for unattended processing. There's also unlimited undo/redo with a full undo history.
What's New? One of the first things I noticed about Peak 4 was that the whole 'Aqua-style' facelift introduced in version 3 has been given a Panther-style brushed-metal makeover that enhances clarity as well as the overall aesthetics. I always found Peak's icons a bit difficult to interpret, but they are much better in this version and the meters are also easier to read. The toolbar and buttons are resizeable, and as before, you can create your own custom control bar containing only the tool icons of your choice — though there's still no obvious way of creating and storing multiple toolbar setups for different tasks, as the toolbar settings are stored as preferences. I still feel this could be a useful future addition. As with earlier versions of Peak, you can create a playlist file that's compatible with Roxio's Jam CD-burning software, but you no longer need Jam to burn audio CDs, as version 4 of Peak has full Red Book CD-burning capabilities built in. As with Jam, you can create playlists using 24-bit files, which are dithered to 16-bit immediately prior to burning, and you can edit the transitions between songs, with the ability to create crossfades between tracks. For those users who'd rather stick with Jam, a full version of Jam v6 is included with Peak 4.1 onwards, which became available shortly before I finished this review.
The Nudge editor has been improved in Peak 4, though it still only covers two regions rather than three.
The core of the program has been optimised for use on the new generation of multi-threaded, multi-processor Apple computers, full Audio Units support is now included for insert effects (though it's still VST-only for the Vbox effects routing matrix), and there are up to five effect insert points that can accept VST or AU plug-ins. Vbox, for those who haven't encountered it before, is a matrix into which you can drop VST plug-ins and then connect them in series/parallel combinations to create multi-effects or mastering processing chains. Each plug-in has its own mini-metering and bypass button, and there are also master Vbox input and output level meters. In addition, Peak 4
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BIAS Peak 4
includes a selection of new processing tools, the most notable of which is the Impulseverb convolving reverb.
Peak At Play The editing ethos of Peak is that regions of an audio file can be selected and saved in a 'drawer' for later reassembly in a playlist (or several different playlists). Playlists may be used for creating albums, with Vbox effects and processors applied with different snapshot settings for each track (Region) if necessary. Vbox may also be used to apply a blanket effect, such as limiting, to a whole playlist, which allows small level changes to be made to individual songs within the playlist without the risk of clipping. However, the Playlist is also an essential tool for editing and rearranging individual songs where you need to remove sections, copy sections or rearrange the order of a piece of music. Playlists can use regions created from different source files and regions may be viewed on a file-by-file basis or as a global Contents list showing all regions for all the files in a project. Peak caters for the independent processing of left and right channels where such is required but, as you'd expect, edits can only be made to both parts of a stereo file, so as to always keep the two channels in sync.
Working With Peak Most stereo editors owe a debt of gratitude to Digidesign's now-defunct Sound Designer II. There's invariably a main window showing an overview of the audio, with a larger, zoomable window that shows you the section of audio you're currently working on. Peak sees the original concept embellished considerably, but for the benefit of old SDII users like myself, it is possible to configure the key commands and control strip icons to work in almost exactly the same way. Furthermore, Peak can open multiple audio documents simultaneously, each document with an individual undo history, and should you need to, Peak allows the user to trigger up to 10 audio documents using the number keys 1-0 on the Mac keyboard. This could have some interesting applications in theatre or as an alternative to CD backing tracks for gigs. The other major piece of SDII legacy is the way in which sections of audio can be marked, saved as Regions, then reassembled in a playlist. This is the tradition way of editing a song or creating a good mix from two or three takes. New regions appear in the Contents drawer and multiple playlists can be created using the same audio material (as you may need to do when doing several alternative remixes of a track). When a project is stored, the audio file and playlist files are saved separately. Peak can record audio via your interface or you can import existing files — in either case, you start work with a New Peak Document. Peak also allows audio to be recorded through a VST plug-in when using Core Audio, which can save file:///H|/SOS%2004-05/BIAS%20Peak%204.htm (3 of 8)9/22/2005 6:07:36 PM
BIAS Peak 4
time when you know exactly what process you need. Existing files in AIFF, Sound Designer II, WAV, AAC, 24-bit Wave and Broadcast Wave, Quicktime, Raw, System 7 Sound, Sonic AIFF, Paris, Jam Image, AU or MP3 formats can be opened by dragging and dropping them onto Peak's icon or by opening them from the File menu. Peak can work with numerous compressed audio file formats, such as AIFF/AIFC and Quicktime files that have been produced using the MACE 3:1, MACE 6:1, IMA 4:1, QDesign or Mu-law codecs. Peak's own MP3 encoding has been improved and now works at up to 320kbps, while the new version can also provide AAC (MP4) encoding. As before, there's Quicktime Movie and DV (Digital Video) clip support offering sync of video to audio to an accuracy of better than a single frame. Peak will also open split stereo files such as may be generated by Pro Tools or BIAS Deck, which it then saves as interleaved stereo audio files. After Previous versions of Peak featured off-line editing, files can be converted to a convolution processing, but Peak 4 adds the real-time Impulseverb process. reduced bit depth using POWR dithering, but as the CD-burning function offers dithering within the playlist, it makes more sense to keep projects at 24-bit until you come to burn the album master. Playback may be started by clicking anywhere in the overview waveform or by placing the cursor in the main waveform display and then hitting the space bar. Users with smaller monitors may wish to turn off the overview display to conserve space. If you select only one half of the waveform, then only that channel plays back, which is useful when looking for problems that exist in one channel only. You can opt to have the audio scroll beneath a stationary cursor or the cursor can move over the audio, which means the screen is redrawn every time the cursor gets to the right-hand edge. Those with marginal computers may be better off switching their monitors to a lower colour resolution if the screen redraw places too much demand on their available CPU resources. Once a section of audio has been selected, it can be played back with a user-determined amount of pre-roll and post-roll, and markers can be inserted or deleted as required. Many aspects of Peak's appearance can be configured as user preferences, including defining custom keyboard commands and determining which icons are displayed in the toolbar. A Pencil Tool is available for redrawing sections of the waveform, with various smoothing options, and there's also an automatic clickfile:///H|/SOS%2004-05/BIAS%20Peak%204.htm (4 of 8)9/22/2005 6:07:36 PM
BIAS Peak 4
removal facility that works well providing the click is recognised as such in the first place (it must be a sharp transient). Peak also offers dynamic or tape-style audio scrubbing.
Peak For Sound Design BIAS Peak's sound-design capabilities have been used on a significant number of major-league movies (not to mention for sample library creation) where its inbuilt processing, such as pitch-shifting, convolution, time compression/ expansion and now the new 'harmonic rotation' can be put to good use. Apparently a number of sound designers also make use of Peak's batch processing facility so that they can speculatively try convolving a whole batch of audio files with the audio they're trying to process, just to see what comes out of the other end. Often the least expected combinations throw up the most interesting results. SMIDI support is available for transferring samples to and from SMIDI-compatible samplers, which can be used via SCSI (or Firewire-toSCSI adaptors); supported models are the Emu E4, ESI32, ESI4000, E64 and E5000, Kurzweil K200, K2500 and K2600, Peavey SP/SX and Yamaha A3000, A4000 and A5000. Peak includes all the standard DSP tools we've come to expect from serious editors, such as sample-rate conversion, normalisation, audio reverse, gain changes, fade-in, fade-out and so forth. It also includes the well-respected POWR dithering algorithm for bit-depth reduction (also used by Emagic and Digidesign), a workmanlike four-band paragraphic EQ, automatic click repair and a waveform redrawing tool that has an effective smoothing option to help avoid clicks or other artifacts. And of course Peak has always had a few more off-the-wall effects up its sleeve, such as Rappify, Reverse Boomerang, Convolve, Amplitude Fit (a granular normalisation process that can make everything equally loud) and now the new Impulseverb. Impulseverb is a convolution-based reverb processor, which comes with over 250MB of impulses to play with and offers real-time preview of the effect type and wet/dry mix prior to saving a convolved version of the file. Peak has included off-line convolution for many years now, and it is incredibly useful for sound-design work, but now you can also 'misuse' Impulseverb in a creative way to achieve very strange effects for long sound files by using a short section of audio (up to two or three seconds is typical) in place of a reverb impulse response. If you like this kind of weirdness, you might also want to try the new Harmonic Rotate function (it sounds like a cross between ring modulation and an off-tuned radio), but if you're the type who likes to settle down with a good book, then the new Sqweez VST dynamic compressor/limiter plug-in might appeal. For the user with slightly different requirements, there's envelope-based dynamic tempo change with markers and a whole range of sample-manipulation tools. Peak is particularly adept at working with sample loops using tools such as Loop Surfer, Loop Tuner and Crossfade Loop. A Phase Vocoder (Fourier analysis and resynthesis) process is available for time or pitch massaging, but how natural it sounds depends very much on how far you need to deviate from the original. I tried a 15 percent speed increase of a mix I'd done and the result sounded rather disembodied! Loop Surfer is ideal for anyone wishing to create or edit sample loops, and file:///H|/SOS%2004-05/BIAS%20Peak%204.htm (5 of 8)9/22/2005 6:07:36 PM
BIAS Peak 4
crossfade looping is supported with user-variable duration and shape. If you know the tempo of the material, Loop Surfer can use that to extract loops of the correct length, but if not, there's a Tempo Calculator that will attempt to work it out based on the length of a selected section of audio. There's also an intelligent Guess Tempo that tries to calculate the tempo of a selected section of rhythmic audio, and Peak can also automatically divide files into regions dependent on audio threshold levels in a similar fashion to the Strip Silence command in Logic.
Editing Regions A key part of any stereo editor is the Nudge window, which allows you to manipulate and fine-tune the transitions between selected audio regions. Peak's Nudge window is also loosely based on that found in Sound Designer II, though it's somewhat more elaborate and includes the ability to manipulate crossfades with a visual display of what's happening. You can view the waveform either side of the edit and you can adjust the region start and end times at the transition on the fly as you audition the result. Two Nudge buttons, with a user-definable nudge time, can be used to fine-tune the edit placement, and fades can be dragged into place as needed. Regions boundaries may also be adjusted via scroll-bars in the Nudge window, and these offer a fine degree of control. In previous versions of Peak, I found the limitations of the Nudge window quite frustrating, but now it's a lot more user-friendly, with plenty of visual feedback. A number of significant improvements have been made, though it still seems that you can only audition one transition at a time. Most Playlists can be exported into Jam for of the time this is perfectly fine, but there are occasions where you may be burning to CD, but discs can also now be burned directly from Peak. pasting in a very short section of audio, such as a single note, drum hit or syllable, where you need to audition two transitions for the edit to make sense. On the plus side, there's now a scrolling cursor that enables you to correlate what you hear with what you see. When Preserve Timing is off, both Region markers can be moved independently, and the waveform display may be zoomed horizontally down to single-sample resolution. The ability to view three Regions rather than two would still be a welcome addition, but on the whole, the Nudge editor is now a joy to use.
Late News Just as I was about to close this review, an update to Peak 4.1 became available, the big news being that Roxio's Jam v6 Red Book CD-burning software is now file:///H|/SOS%2004-05/BIAS%20Peak%204.htm (6 of 8)9/22/2005 6:07:36 PM
BIAS Peak 4
included at no extra charge, as is SFX Machine LT, a cut down version of the fullblown SFX Machine plug-in. Peak still has its own CD-burning facilities, but Jam is included for the benefit of those audio professionals who are used to working with it and who may also need to use some of its more advanced features, such as the ability to write ISRC codes when mastering. Jam may also be configured to launch Peak as its default audio editor. SFX Machine LT, which supports both AU and VST plug-in formats, comes with 21 effects presets and has a randomise feature to allow serendipity to take a hand when you run out of ideas! It is especially useful for creating unusual effects and as such is as useful in sound design as it is in standard music production. Users can upgrade to the full version at a discount price via an upgrade link built into the plug-in and you can read a review of the full version in the July 2003 issue of Sound On Sound (www.soundonsound.com/sos/jul03/articles/ plugin0703.asp)
Peak Performance I have to say that Peak is the first stereo editor package that's made me think I could finally abandon my old copy of Sound Designer II without feeling too insecure. Peak is of course a much more sophisticated program than SDII, but as an editor, it is fast and largely intuitive. I had a couple of crashes with version 4.02 but after updating to 4.03, and later 4.1, I found Peak very solid and the visual redesign has made the program much friendlier than earlier versions. The PDF manual has now been augmented by an optional ($25) well presented paper manual and the introduction of Audio Units support makes perfect sense now that it is the official Apple plug-in format. Because nothing is made permanent until a file is saved, multiple levels of undo are always available, and it is easy to delete segments or add sections of silence to a recording, which is vital for any dialogue-editing applications. All the traditional DSP functions work predictably well and the new convolution-based reverb is also very effective as well as being simple to use, though I'm less convinced by some of the weirder inclusions. Nevertheless, there are creative The just-announced update to version 4.1 people out there who will love them, so sees Roxio's Jam CD-burning package bundled with Peak, along with the SFX it's better to have them than not. For Machine LT plug-in suite. me, it's things like editing, waveform drawing and declicking that are the most important, as they are fundamental to any editing task. Those provided here work extremely well, though automatic click recognition only works on very
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BIAS Peak 4
obvious fast transients. The drawing tool has an effective smoothing option, so most short-duration artifacts can be disguised without leaving any audible glitch or thump. Vbox seems to have changed little since the last time I visited Peak (other than being reformatted as a four-by-four matrix to give the user more screen space), and the one criticism I had of it remains — there's no simple way to pan parallel sound streams to create stereo multi-effects other than by inserting a dedicated panner plug-in. A pan control at the end of each output stream is all that's needed! The ability to set up snapshots of VST plug-in settings for each Region in a playlist was included in version 3, but it's a great feature and well worth another mention. Peak has really matured as a stereo editor, and its comprehensive set of tools makes it so much more than just a stereo editor, though of course there's always the lower-cost Peak LE or Peak DV for users with simpler requirements. Especially in light of the recent announcement by TC Electronic that there will be no more development on the rival Spark, every Mac user involved in editing, sound design or mastering should take a very close look. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Boss BR1600CD
In this article:
Overview Getting Started Automation & Sync Making A Recording Get Rhythm (And Bass) Adventures In Sound Further Adventures Audio Editing Final Analysis
Boss BR1600CD £1099 pros Much more than just a multitrack recorder — many users will get by without the need for a separate drum machine or guitar effects unit. In some respects, it's more immediate to use than most multitrackers. Eight XLR mic inputs, all of which can be recorded simultaneously. Very quiet in operation. Records without data compression. Good complement of effects, with eight dedicated compressors and 16 track EQs. Some of the 'traditional' features that have been left out aren't missed at all...
cons ...but some, like monitor mixing and track soloing, are missed quite a lot. Poor metering. Input gain structure is confusing, and places digital gain controls in the record path. Effects can't be sync'ed to tempo or controlled by tapping the Tap Tempo button. Some aspects of the BR's operation are frustratingly slow, especially loading and
Boss BR1600CD 16-track Digital Multitracker Published in SOS May 2004 Print article : Close window
Reviews : Multitrack Recorder
Boss's range of digital multitrackers is designed to allow musicians to make recordings with a minimum of technical knowledge. The BR1600 is the most sophisticated yet, providing 16-track recording along with amp modelling, vocal harmony generation and a drum machine and loop sequencer. Sam Inglis
Every manufacturer has their own take on the all-in-one digital multitracker, and Boss's is summed up by the slogan on the BR1600CD's box: 'The easy way to record your band'. This is a product intended not for the seasoned audio engineer, but the musician who wants to get results without having to calibrate signal levels or delve into routing matrices.
Photos: Mike Cameron
Like its junior siblings the BR532, BR864 and BR1180, the BR1600CD thus has what you might call a 'streamlined' feature set. Everything has its own dedicated button — none of your shifted functions here. There's no choice of sample rates or bit depths, no monitor mixing, and not much by way of routing options. There's no waveform editing, side-chaining, mix bussing or grouping, nor even a track solo function, and the design encourages recording with effects rather than tinkering at the mix. The drive towards ease of use hasn't just led Boss's designers to make a bonfire of inessential options, however. They have also included some attractive extras such as guitar amp modelling, a drum machine, a bass synth and a loop sequencer, not to mention a selection of hand-holding features designed to make
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Boss BR1600CD
saving Songs. Only one output pair, and that on RCA phonos. Too many functions are buried in the Utility menu. Undo button doesn't work for the majority of actions. Transferring WAV and AIFF files via USB may produce that 'life's too short' feeling.
recording faster and more immediate.
Overview
The BR1600CD is described as a 16-track recorder, although there are a few compromises along the way. The first is that tracks 9/10 to 15/16 are always configured as stereo; the second is that the built-in drum-machine, bass synth, and loop sequencer each take up a stereo track if you use them; and the third is summary that, unless you want to use a separate two-track recorder, you'll need to keep Boss's BR-series recorders are designed for the musician, tracks 9/10 free to bounce your mix to. On the plus side, however, each track has not the engineer, and though 16 virtual tracks for storing alternate takes and the like. All audio is recorded at the BR1600CD lacks the 16-bit resolution and 44.1kHz sample rate, with no data compression, and drive flexibility of some competing capacity is a generous 40GB. Up to eight inputs can be recorded simultaneously, multitrackers, it provides an with the socketry on offer consisting of eight XLR mic inputs, eight balanced effective and reasonably quarter-inch jack inputs, a high-impedance jack for DI'ing electric guitars and simple way of recording a band. basses, and co-axial S/PDIF I/O. Globally switched phantom power is available for the mic inputs, though it resets to 'off' every time you power the unit down. information £1099 including VAT. Roland UK +44 (0)1792 515020. +44 (0)1792 799644. www.roland.co.uk www.roland.co.jp
Test Spec Boss BR1600CD OS v1.01. Apple G4 iMac 700MHz with 256MB RAM running Mac OS 10.3.2.
The output arrangements are rather less impressive: there is only a single stereo output pair, and that emerges on RCA phonos, nobody's favourite connector (although it will suit musicians who are hooking the BR up to a hi-fi, rather than a roomful of studio gear). There are two front-panel quarter-inch headphone sockets with independent volume controls. This is as many as most of the BR1600CD's rivals provide, but still not enough for many band-recording applications, and there's no way to set up a headphone mix that's different from the main mix. Audio data can also enter and leave the BR1600CD via a USB connection to a Mac or PC (an upgrade to version 1.01 of the BR operating system is required for use with OS X), and of course there's the built-in CD rewriter, which fulfils the You can connect the BR1600CD to your computer via USB, and this usual backup and allows you to transfer audio files to and from your choice of software file-transfer MIDI + Audio sequencer or sample editor. The multitracker's drive is seen by the computer as a separate external hard drive. functions, as well as writing audio CDs. It's handy to be able to back up BR projects via USB, but transferring WAV or AIFF files between the BR and a computer is no fun at all. They have to be converted to or from the BR's native format, a process which is about as snappy
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as continental drift, and the BR also takes ages to mount as a USB drive (on a Mac running OS X, at any rate). This might not be so bad, except that you can only import or export one file at a time via USB, so transferring a multitrack project involves dismounting and remounting the BR1600CD for each track. The BR also refused to recognise some of the AIFF files I tried to transfer to it, whether via USB or CD-R. The unit itself is fairly large at 494 x 339 x 99mm, with a robust plastic case, and draws its power from an external line-lump adaptor. All the information is presented on a 130 x 36mm backlit LCD, which is navigated with cursor keys and a data wheel. There are also four function keys and four rotary encoders, which take on different roles depending on what you're doing with them. Each track has a fader and a single button which serves for track selection, record arming and muting. It's all pretty easy to navigate, though I was occasionally confused by the system used to display the function of the assignable buttons. When you're editing a compressor and the screen says 'Link on', for instance, what it actually means is that stereo linking is turned off, but that pressing the button below that message will turn it on. Three-band EQ is available on every track, and there are eight dedicated track compressors; the two sides of each stereo track can be unlinked for independent dynamic and EQ processing. There are also two global send effects, or Loop Effects as the manual describes them, one devoted to reverb and the other to chorus and delay-based effects. Finally, there's a powerful 'floating' insert effects processor that can take on various tasks — you can spread its resources across eight tracks to provide additional compression and EQ, or focus them to provide more sophisticated functions for one or two tracks. The options available include COSM guitar-amp emulation, a Vocal Toolbox pitch corrector and harmony generator, a Mastering Toolkit, and speaker modelling for use with Roland's digital monitors.
Getting Started When you call up a new Song, the BR1600CD asks you whether you intend to use the drum machine, bass synth, or loop sequencer, and configures the tracks accordingly. If you do want to avail yourself of these rhythm aids, you'll need to tangle with the BR's Pattern sequencer — see the 'Get Rhythm (And Bass)' box for more detail. If you just want to go straight ahead and record audio, the BR does its best to guide you through the setup procedure. First, you choose one of four input assignment options by pressing the appropriate button. In Guitar/Bass mode, the BR assumes you've plugged your instrument directly into the front-panel quarter-inch jack, and that you want to play and record through the built-in COSM amp modelling and effects. Likewise, Vocal mode is designed to cope with a mic plugged into input two, and automatically places a chain of vocal effects, EQ and dynamics processors in its path. Pressing the Guitar/Bass and Vocal buttons at the same time allows you to use both modes file:///H|/SOS%2004-05/Boss%20BR1600CD.htm (3 of 11)9/22/2005 6:07:41 PM
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simultaneously. The Multitrack and Stereo Tracks modes make use of all eight inputs, if needed. In Multitrack mode, each input is assigned to the corresponding track, or to its equivalent in the 9-16 bank, while in Stereo Tracks mode, all eight are mixed down to a stereo pair and recorded to the track pair of your choice. In Multitrack mode, the BR's insert effects processor is automatically configured to place a lowcut filter, compressor and EQ across each track (in addition to the dedicated track compression and EQ that's always available). You can choose from various preset settings such as Rock Band, Jazz Band and Multi Drums, but the thinking behind the compressor and EQ settings for the different channels isn't always obvious. The default setting in all the input modes has the insert effects placed in the record path as well as the monitor path, though you can choose to switch them into the monitor path only, into the playback path as conventional mixdown inserts, or off altogether. The usefulness of recording with effects such as (digital) compression and EQ is debatable. On the one hand, it makes recording a more immediate process, which is in keeping with the BR1600CD's raison d'être, and frees up DSP resources at mixdown. On the other, if you overdo things at the tracking stage, there's little you can do to remedy them at the mix, and the preset effects settings that are applied by default might do more harm than good. I also found it a bit odd that neither the insert effects processor nor the dedicated track compressors can provide multi-channel gating.
Automation & Sync The BR1600CD can save up to 100 mixer scenes per Song, which can be assigned to markers for scene-based automation; and though the faders aren't motorised, they are also capable of dynamic automation when used in conjunction with an external MIDI sequencer. Channel volume messages are sent and received to indicate fader positions, and the default arrangement has fader one sending and receiving on channel one, fader two on channel two, and so on. This works fine, provided that your MIDI sequencer is capable of recording data on multiple MIDI channels simultaneously and keeping it separate (a process which in Pro Tools, for example, involves creating and record-arming 16 individual MIDI tracks). A helpful Fader option in the Utility menu displays the virtual fader positions as they play back. As well as controlling an external MIDI sequencer using MIDI Machine Control, the BR1600CD can also send MIDI Time Code, or control another unit using MIDI Clock and Song Position Pointer data. It can't act as a sync slave, but can record timing information from an external sequencer to use as a sync track.
Making A Recording
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Navigating a BR1600CD Song is as straightforward as loading one from disk is slow (and it is slow: even the most basic Songs take upwards of 40 seconds to load). The Time/Value dial can be used to wind forward or back through the Song and there's the usual complement of transport controls. Each Song can contain up to 100 markers; there are dedicated search buttons to jump to the next and previous marker, but all other marker-related functions are buried in the Utility menu. Tracks are armed by pressing the track button once, whereupon the button flashes red; recording is started by pressing Rec, then Play. You can also punch in using a footswitch (not supplied), or set in and out points for dropping in automatically. The mic preamps are obviously not in the Avalon or GML class, but they're perfectly usable, with only a little hiss creeping in even at the highest gain settings. Classical recordists and those with a large collection of ribbon mics would be unlikely to choose the BR1600CD anyway, and there's enough gain available for its intended applications. The sensitivity of each input is adjusted using a trim pot in the usual way, and each is equipped with a single red LED which lights when the The BR1600CD's unusual Level Calibration analogue signal reaches -6dB. These button can be used to set up your recording LEDs are your only guide when setting input levels automatically. input levels: the manual instructs you to turn up the gain until they 'light up occasionally when the instrument is played hard'. Metering on the digital side isn't a lot better — you get a fairly dinky multitrack bar-graph display in the LCD, with nothing at all to indicate digital overs. The internal effects and processors are equally devoid of gain-reduction meters and the like, which is a shame. While recording it's impossible to set up a monitor balance that's different from the actual levels going to disk, because the faders only affect playback levels of previously recorded tracks. If the drummer wants to hear less tambourine and more snare in his headphones, in other words, you'll have to record the tambourine more quietly and the snare louder. You could do this using the preamp gain knobs, but in Multitrack and Stereo Tracks modes each track has an additional 'soft' gain control that works in the digital domain, following the insert effects. The gain controls are calibrated from zero to 127, with values below 100 providing attenuation and values above 100 increasing gain, up to a maximum of 6dB. The BR1600CD will adjust these gain controls automatically if you like — simply press the Level Calibration button, play the instruments for a few seconds, then hit Calculate. Recording purists who don't want their signals degraded by digital level controls must also contend with a global Input Level knob. According to the manual, this also sits in the record path after the insert effects, but it's not calibrated and no
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explanation is forthcoming as to whether it simply offers attenuation or provides an additional gain boost. The lack of information makes it hard to know what the optimum setting should be. Again, the design has been compromised for the sake of simplicity, but whilst experienced engineers will undoubtedly find it restrictive, the bottom line is that it is possible to make good-sounding recordings with this unit. In the real world, the quality of recordings made on the BR1600CD is going to be limited by mic placement, ambient noise, room acoustics and instrument quality, not by the quality of the preamps, the lack of 24-bit recording or the degradation caused by digital attenuators. In fact, for all its fixed 16bit/44.1kHz nature, the BR1600CD will deliver better recordings than its competitors in many situations, because it's almost completely silent in operation. I recorded a quiet finger-picked acoustic guitar part with the mics quite distant and the BR in the same room, and no trace of hard drive noise made it onto the recording.
Get Rhythm (And Bass) As well as being 'the easy way to record your band', the BR1600CD also caters for those who aren't lucky enough to have a real rhythm section on hand, or who want to blend live performances with sequenced material. A built-in drum machine and bass synth are controlled by a Pattern sequencer, which can also be used to string together sampled loops, known in Boss-speak as Loop Phrases. The drum machine takes over tracks 15/16, the bass tracks 13/14, and the Loop Phrases tracks 11/12, so if you use them all, the BR1600CD is reduced to a 10-track recorder. Drum and bass Patterns can be created by real-time recording, either from an attached MIDI keyboard or using the BR1600CD's track buttons. You can choose to have your recording quantised as you play it in, but there is no way to undo this, or to quantise a Pattern that's already been recorded. Alternatively, drum Patterns can be created in a friendly grid editor, while step-time recording is available for composing bass parts. I always find these methods laborious and restrictive compared to software sequencers, but the implementation here is as good as most. If you want to make detailed adjustments to your drum and bass Patterns, you can do so via the Microscope list editor. There's a limited range of bass sounds; a few different electric sounds are represented, along with acoustic and fretless basses and a couple of synth-bass tones. They're perfectly usable, though it's not clear why any of them needs to waste a stereo track. Nine drum kits likewise cover the basics, but don't offer too much scope for experiment, and there's no way to edit drum kits or create your own. You can, however, connect up a MIDI module and have your drum and bass parts played on that. You can also create your own Loop Phrases from audio recorded onto the BR1600CD, or import loops from sample CDs in WAV or AIFF format. The process of browsing a sample CD, auditioning loops and importing them to Loop Phrases is straightforward, but involves a bit of hanging around. A nice touch is the bundled Discrete Drums CD-ROM, which contains well-played and wellrecorded drum parts for 14 songs, each broken down into loopable verses, choruses, and so on. If the tempo of your Song is different from the tempo of a Loop Phrase, the phrase will be time-stretched to fit. I've heard better time-
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stretching, though, and I found the results were only acceptable within a fairly narrow range. The Arrangement sequencer works pretty much as you'd expect, although the scale of the graphical timeline is so small as to be of little value. As well as specifying which drum Pattern, bass Pattern and Loop Phrase to play back, each sequencer measure can store tempo and chord information, allowing you to create a tempo and chord map across your Song. The latter is essential if you want to use the automatic vocal harmony generator, and offers a healthy range of extended, diminished, and augmented chords as well as the basics and a 'non-chord' option. However, I found it tough to get my head round the way chord sequences affect the bass Patterns. It seems that all Patterns, both preset and user-defined, are regarded by the BR1600CD as being by default in the key of 'C', so when the Pattern sequencer encounters a different chord, it modifies the bass part to fit. You can audition how your bass parts will sound in different keys from the Pattern editor, but you can't specify that a Pattern should have a root key other than 'C'. The upshot of this is that if you want to employ the chord mapping and you're using your own bass Patterns, you need to transpose them all into the key of 'C' before you enter them. This would be annoying anyway, and combined with the lack of a clear explanation in the manual it undermines the BR1600CD's supposed simplicity and intuitiveness. The BR1600CD's sequencing facilities are OK for stringing together a few loops off a sample CD, or making a very rough backing track for a demo, but creating anything more ambitious would involve a lot of fairly tedious work. No doubt it would get easier with practice, but I found myself scurrying back to Cubase at the first opportunity...
Adventures In Sound In Multitrack mode, the eight-channel compressor/EQ setup is the only configuration available from the insert effects processor. If you are recording a solo instrument or vocal, however, you can get stuck into some more adventurous effects. Various different Guitar choices are available, beginning with a COSM-modelled overdriven guitar amp providing the usual choice of amps and speakers, from a sparkly JC120 emulation through bluesy Voxes and Marshalls to high-gain metal tones. Additional modelling is used to recreate a variety of overdrive and distortion pedals, or you can choose an alternative signal path with COSM compressor modelling instead. Four compressors are modelled: Urei 1178, Dbx 160X, MXR Dynacomp, and Boss's own CS3. It's all easy to use and versatile enough for most applications, though I found many of the amp models had a fizzy and synthetic-sounding edge to them, and given the choice I'd rather use my Johnson J Station. Penurious electric guitarists can also turn to an Acoustic Simulator, which gives usable results in a busy mix, and a surprisingly reasonable Bass Simulator. You have to be careful how you play to use this, as any hint of a chord or overlapping notes will throw the pitching way off, but it would be fine for simple bass lines. If
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you're lucky enough to own a real bass you can choose a bass amp simulator or a multi-effects patch, while the Acoustic Guitar option is designed to make a DI'd steel-string sound more like a miked one. Vocal Multi is the default choice for a vocal mic, and provides compression, de-essing, enhancement, EQ, a noise suppressor, chorus and delay. COSM Vocal Comp is similar, except that it dispenses with the chorus and uses a COSM modelled compressor/limiter rather than the basic dynamics algorithm. If you're after something more experimental, you could route your mic through the Voice Transformer algorithm, derived from Boss's dedicated vocal processing units, or a microphone modeller, which allegedly 'modifies the sound recorded using a conventional dynamic mic, lapel mic or direct line, making it sound as though it had been recorded using an expensive condenser mic or a special studio mic'. Don't put that AKG C12 on Ebay just yet... There are also three Simul algorithms, which divide the DSP resources between a guitar and a vocal input. These combine basic vocal processing with amp simulation, acoustic guitar emulation or the de-DI'ing acoustic process. Finally, two Stereo algorithms are available for use across a track pair: a multi-effects patch incorporating ring modulation, dynamics, flanging, EQ, and delay, and a lofi patch. In all, there is a total of 18 different insert effects configurations on offer. In each case, you can turn individual modules off and on, change the order, edit their parameters, and store the results as User patches, but you can't create your own routings or swap the effect modules out for different ones. It's another area where Boss have chosen to compromise the BR1600CD's flexibility in favour of ease of use, but as compromises go it doesn't seem too terrible. What is a real shame, given the emphasis on ease of use, is that none of the effects can be sync'ed to the BR's tempo. Nor can you use the Tap Tempo button to set delay times and the like.
Further Adventures If you don't mind forgoing the insert effects, you can use the same DSP resources to see what your mix will sound like on a different set of speakers using the Speaker Modelling, or avail yourself of the Vocal Toolbox or the Mastering Toolkit. Both of the latter are actually insert effects of a sort, except that neither can be used on input signals; the Mastering Toolkit can only be used on track 9/10. All of these are familiar from previous Roland and Boss products, and the Speaker Modelling is optimised for Roland's DS-series monitors, which file:///H|/SOS%2004-05/Boss%20BR1600CD.htm (8 of 11)9/22/2005 6:07:41 PM
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can be connected digitally. The Vocal Toolbox provides pitch-correction and automatic harmony generation. These can't be used at the same time, and the latter will only work if you have created a rhythm arrangement with a chord sequence. In the sections of the Song where you've specified a chord rather than the non-chord options, up to three artificially generated voices will sing harmony, with healthy doses of reverb and chorus available to smother the results. If my experience is anything to go by, you'll need it. The pitch-correction is not exactly transparent, either, though it still might be the best option in some circumstances! It's certainly easy to use — you simply select the type of voice you're working on (male or female, high or low) — and adjust the Smoothing amount to taste. Some glitching will be audible, but with luck it won't be too obvious. There's no way to specify a scale, and it doesn't seem to take any notice of the chord sequence. Finally, the Mastering Toolkit is designed to even up the levels of your mixes, give them that 'produced' sheen, and impose automated fades in and out. The processing on offer combines EQ and enhancement with a three-band dynamics unit, followed by a limiter and output dithering. It's no substitute for a trip to Metropolis, but doesn't seem to offer any nasty surprises either. You'll need to bounce your mix down to stereo track 9/10 before using the Mastering Toolkit, and its output is bounced to one of track 9/10's virtual counterparts. When you do bounce your mix to a stereo track using the BR's Bounce recording mode, the inputs are muted by default, but it is also possible to use them to bring external sound sources in at mixdown.
Audio Editing You never get to see a waveform on the BR1600CD's display: that level of editing is obviously considered too daunting for prospective users. It is, however, possible to cut, erase, copy, paste, and move chunks of audio, either within the same track or between tracks. Editing functionality is accessed by pressing the Utility button and choosing the Track option. You delimit the section of audio to be moved or copied by specifying its start and finish position in bars and beats, in minutes and seconds, or by using marker numbers. To place markers precisely you can scrub through an audio track using the data wheel, though you can only scrub one track at once. It should be pointed out that what Roland call 'scrubbing' is not very much like tape rocking past a playback head; instead, you hear a half-second loop on constant repeat, sounding not unlike a piece of experimental German glitch music. Some people prefer this method, but I'm not one of them, and I found it much easier to put markers in roughly the right spot in normal playback, then fine-tune their position later — although doing so requires you to delve once again into the Utility menu. Overall, the BR1600CD's audio editing is adequate for simple tasks like compiling a vocal from several takes, but not for more complex work such as multitrack edits on drum parts.
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Final Analysis It's tricky to review a machine like the BR1600CD in the pages of a recording magazine, if only because it seems to be aimed at exactly those people who can't be bothered to read recording magazines! Anyone who already has a good grasp of the principles underlying a 'traditional' studio will find this machine inflexible and sometimes frustrating. With its woeful metering and tendency to insert effects and digital gain controls in the record path, it doesn't exactly encourage classical recording practices, either. But the BR1600CD isn't intended for the Al Schmitts of this world: its target market is the impatient musician who wants to record his or her band with the minimum of fuss. I know several people in exactly this position who've bought the smaller BR1180 10-track, and their experiences really do bear out Boss's claim to have made recording easier. It's certainly interesting to see just how many of the features we traditionally associate with products of this kind can be removed without compromising the utility of the recorder too much. And in many cases, the resulting simplification does make the machine more approachable. Not having to deal with an input routing matrix is a blessing, for instance, and I didn't find myself bemoaning the fixed bit depth and sample rate at all; nor did the semipreset effects setups prove restrictive. Yet for all the streamlining that has gone into its design, the BR1600CD is fundamentally a much more complicated machine than its predecessors. The resulting lengthy boot-up, loading, and saving times make for a less immediate recording experience all round, and in other respects it's just plain confusing — the input gain structure and the bass chord sequencing spring to mind. The designers' aim to have no shifted functions is laudable, but the practical upshot of it is that lots of commonly used features are buried in the Utility menu, because there's no space for them to have dedicated buttons. Boss have also missed a few simple tricks that would have helped make it easier to use, like setting delay times with the Tap Tempo button or sync'ing them to the BR's tempo. The friendly Undo/Redo button is not as useful as it could be, because very few actions are actually undoable: it doesn't work on any changes made in the Arrange or Pattern editors, for instance. And there are definitely areas where Boss have thrown too much onto their features bonfire. The inability to solo tracks is infuriating, for example, as is the fact that you can't change the input balance in the monitor path without changing the recording levels. There's a clear trade-off in multitracker design between versatility and ease of use, and like Boss's previous BR-series machines the BR1600CD deliberately sacrifices flexibility for the sake of simplicity. I don't think the balance works quite as well here as it did with the BR1180, and a little time invested in learning a more versatile 16-tracker might mean less frustration in the long run. But let's not lose sight of the bigger picture: in the UK £1099 still buys you a powerful 16-track recorder, and the drum machine, amp modelling, and loop sequencer are valuable extras. Realistically, no multitracker is utterly intuitive in all its aspects and, as long as you're aware of its limitations, the BR1600CD does indeed file:///H|/SOS%2004-05/Boss%20BR1600CD.htm (10 of 11)9/22/2005 6:07:41 PM
Boss BR1600CD
provide a straightforward way to record a band. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Emagic Logic Pro & Express
In this article:
Whither Sound Diver? At The Top End Those Lovely Plug-ins, Part 1: Effects The Big Express Those Lovely Plug-ins, Part 2: The Instruments For Future Reference Pricing & Upgrades Summing Up
Emagic Logic Pro & Express £699/£199 pros Excellent value. OS X and OS 9 still supported.
Emagic Logic Pro & Express Recording Software For Mac OS 9/OS X Published in SOS May 2004 Print article : Close window
Reviews : Software
The big news for Logic users is not the new features in Logic Pro and Express, but the new — and extremely competitive — pricing. So which of the two is the bundle for you? Paul White
cons
By now, pretty much every Logic user must have heard that Apple are making changes to the way Logic is being marketed, the most noticeable being summary that the Logic range is being The introduction of Logic repackaged as just two products: Logic Express and Logic Pro means Pro and Logic Express. This would the end user gets an seem to mark the end of separately incredible deal, either as an upgrade or as a new sold plug-in products for Logic — the purchase. impression we're getting from Apple is information that all future plug-ins will be added to the Pro program during future upgrades. See 'Pricing & Upgrades' We saw some intriguing new software instruments in preview form at the January box. NAMM show, though these aren't in the first Pro version, and because Apple Sound Technology +44 have a policy of not announcing shipping dates until they're pretty sure they can (0)1462 480000. meet them, they can't give us any clues as to when they will ship. So, what is in +44 (0)1462 480800. these two new packages and what does it mean for the existing Logic user? Seems a little tough on the loyal Emagic users who have already bought everything.
Click here to email www.soundtech.co.uk www.emagic.de
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Emagic Logic Pro & Express
Whither Sound Diver? What is still unclear at this point are Apple's plans for the Emagic Sound Diver synth librarian. It is still available as an OS 9-only product, but a beta version of Sound Diver for OS X is available for free download from the Emagic web site. This requires an XS key and therefore can only be used by existing Logic v5/6 users. Emagic and Apple are making no comment at present about the future of Sound Diver.
At The Top End Starting with Logic Pro, this is essentially Logic Platinum plus every additional plug-in Emagic have ever produced, namely ES1, ES2, EVOC20, EXS24 MkII, EVB3, EVP88, EVD6 and the new Space Designer convolution reverb. If you're not familiar with any of these formerly optional plug-ins, see the boxes. In addition, Logic Pro naturally includes all Logic's usual built-in processing and virtual instrument plug-ins, plus a good selection of samples to use in the EXS24 sampler. On top of this, the previously optional Digidesign support add-ons, ESB TDM System Bridge and the PTHD Extension, have been included too, whilst EPIC TDM and Host TDM Enabler are discontinued. Looking very briefly at the key features of Logic Pro, it can handle up to a massive 255 stereo audio tracks, CPU power permitting, plus a further 64 plug-in instrument tracks. Its software mixer can be configured with up to 64 busses and 64 aux tracks, where you can have up to 15 inserts and eight sends per channel — not that you'd ever be likely to need that many! There are now over 50 built-in effect plug-ins, plus all the previously optional plug-ins, real-time and fast offline mix bouncing, and support for a number of popular surround formats including quad, 5.1 and 7.1. POWR dithering is built in for reducing bit depths from 24 to 16, while the ESB TDM System Bridge allows Logic users to combine their Mac's native processing engine (and hence all Emagic instruments and plug-ins, plus any installed third-party Audio Unit plug-ins) with a DAE Digidesign Audio Engine within Logic and route all audio through the Pro Tools hardware. Support for TDM hardware and TDM plug-ins (with compatible systems) is built-in, as is support for the newer HD and Accel systems. OMF and Open TL file import and export eases the job of moving projects between platforms and of course there's the new Project Manager for keeping track of audio files and samples. Like Logic Platinum 6, Logic Pro 6 is optimised for Apple G5 and G4 machines. Purchasers of Logic Pro get a new paperback-style Logic 6 manual, plus all the individual paper manuals for the plug-ins and electronic versions of all the manuals in PDF format. With the Logic Pro repackaging, the version number has now crept up to 6.4 — but in truth, little has changed since version 6.3 other than some enhancements to Space Designer. For example, when IR responses are loaded, they are now file:///H|/SOS%2004-05/Emagic%20Logic%20Pro%20&%20Express.htm (2 of 8)9/22/2005 6:07:46 PM
Emagic Logic Pro & Express
automatically converted to match the sample rate Logic is currently working at, pre-delay and IR Start are now separate variables, and a 'Grace Time' feature has been added that delays the recalculating of the impulse response when you're adjusting parameters so that Space Designer is not always trying to work out the answer before you've finished asking the question. The graphical user interface has been upgraded slightly to provide faster access to key parameters, and the filter cutoff is now expressed in Hz. If you don't have Logic Pro and don't plan on getting it any time soon, you can download an upgrade to 6.4 from previous versions of Logic 6 free of charge and enjoy all the benefits of Logic Pro other than the additional free plug-ins — your XS key will work as normal and only allow you to use additional plug-ins that have already been authorised to your key. However, I suspect lots of users will jump at the the new upgrade path to the full Logic Pro with all the previously optional plug-ins. The winners are current owners of Logic Gold or Platinum versions 5 or 6 (including the versions of Gold that came as part of a bundle) who have bought few or no additional plug-ins, because for a one-off payment of £149 including VAT, they can move directly to Logic Pro with all the trimmings. In the UK, just contact Sound Technology directly or your Emagic retailer. This will also make a few serious users, especially those who have just bought Space Designer, want to throw themselves under the nearest passing Cubase delivery truck. The 'sick as pigs' brigade will be made up of serious Logic users who have previously bought most or all of the optional plug-ins and who have recently bought Space Designer as well. While they don't actually lose any money, they are in the unenviable position of seeing other Logic users upgrade very cheaply to what actually cost them quite a lot of money. However, this is an Apple strategy, and if it keeps costs down in the future, we could all be winners. After all, Apple are really in the business of selling computers, so if they can tempt us with low-cost but CPU-hungry new features, such as the forthcoming Sculpture physical modelling synth (which I'm assuming will be part of a forthcoming revision to Logic Pro later this year), we're more likely to go out and buy faster Macs. Of course that may not be their reason at all!
Those Lovely Plug-ins, Part 1: Effects EVOC20 is essentially a vocoder that can be used to combine two audio sources (one via its side-chain) such that the frequency spectrum of one is imposed on the other. Up to 20 bands of analysis and synthesis are available, but further controls such as modulation LFOs, resonance and offset allow the sound to be bent and mangled in a number of very creative ways. There's also a built-in simple synthesizer that's optimised for creating the classic talking synth vocoder effects. Additionally, you can also use the filter bank and its modulation sources as an insert effect in its own right, where the simple graphical interface makes it easy to just dabble until you hear something you like. I particularly like the filter bank section for processing drum loops and percussion parts. Space Designer is one of the latest generation of convolution reverb plug-ins and works by using an impulse response taken in a real space (or passed
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Emagic Logic Pro & Express
through a hardware reverb processor) to recreate the acoustic of the original by convolving it with any audio signal that happens to be passing through it. This technique works very well and is largely dependent for its success on the library of impulse responses that are available. However, convolution is very processorintensive so G4 users will have to be careful how they use it — and the longer the decay time you set, the more processing power you need. G5 owners can be somewhat more cavalier about the whole thing but if you have a G3, don't even go there! Space Designer comes with a disc containing over 1000 impulse responses from natural spaces, outdoor locations, odd buildings, and of course everyone else's hardware reverbs. One restriction of convolution reverbs is that there's a limit to how much tweaking you can do to the effect without making it sound unnatural, but this one has a reasonable amount of scope and the included library is excellent. Once you've used this type of reverb processor, you probably won't want to go back to synthesized reverb plug-ins at all.
The Big Express Logic Express is equivalent to the previous Logic Audio and comes bundled with ES1, EVP73 (a slightly scaled down version of the EVP88 'tine piano' plug-in instrument without all the effects) and EXS24P, a play-only version of the EXS24 sampler. Although this does not load Akai-format discs or permit editing sampler instruments or creating them from scratch, it does import EXS24 instruments from existing libraries, thus making it a very powerful sound module. In addition, purchasers get the Xtreme Analogue and Xtreme Digital sample libraries (which are not included with Logic Pro), and all this for a new bargain-basement price of under £200. Though Logic Express, now at version 6, is a slightly cut-down version of Logic Pro, other than having fewer plug-ins, it offers as much in the way of features and flexibility as most users will every need. What's more, it will still run any of the optional Logic plug-ins you might have kicking around, though these are no longer available separately. There's also a simple upgrade path from the popular Logic Audio Big Box or Logic Express to Logic Pro for £499, which may well be worth it simply because of the value of the plug-ins that come with it. Express can handle up to 48 stereo audio tracks and 16 instrument tracks, its
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Emagic Logic Pro & Express
internal mixer can be configured with up to eight busses, and the system handles audio resolutions of up to 24-bit/96kHz depending on the hardware you're running. Like Logic Pro, the audio engine has 32-bit floating-point internal resolution and the relatively new Track Freeze function is included, allowing processed audio tracks or software instrument tracks to be temporarily rendered as audio files to save on CPU load. Express has slightly fewer plug-ins than Pro, but users still get 28 effect/processor plug-ins plus six software instruments. You can run so many MIDI tracks there's no real practical limit and both versions feature a decent Score Editor with real-time notation and printout. For newcomers, there's a Setup Assistant in the OS X version, while OS X users can also import and export MP3 files. Though Apple are trying to coerce everyone to switch to OS X, however, Logic Express and Logic Pro still support OS 9, and anyone moving up to OS X needs to be aware that VST plug-ins are not directly supported, so any third-party plugins will need to be upgraded to the Audio Units format or used with a third-party wrapper such as FXpansion's VST To Audio Units Adapter. VST plug-ins are still supported in the OS 9 version.
Those Lovely Plug-ins, Part 2: The Instruments ES1 was one of Emagic's first optional plug-ins, and arguably also their most visually tasteless. Blessed with a front panel so green that it reminds one of freshly squeezed caterpillars, ES1 is a comprehensive monosynth. It's polyphonic and follows the analogue paradigm, but has just a single oscillator and works well for bass sounds, dance lead lines and so on. In addition to a good choice of basic waveforms, ES1 also has a sub-oscillator, distortion, glide, a fairly flexible modulation system and a choice of resonant filter types. It also has an 'Analog' control setting to alter the pitch and cutoff frequency of each note in a random but subtle manner to make the sound more analogue. ES2 is a far more sophisticated polysynth, which still works on the analogue paradigm but is also capable of various digital synthesis methods including wavetable synthesis, and comes with an impressive library of presets to get you started. ES2 has a seriously file:///H|/SOS%2004-05/Emagic%20Logic%20Pro%20&%20Express.htm (5 of 8)9/22/2005 6:07:46 PM
Emagic Logic Pro & Express
heavy-duty modulation matrix not unlike that offered by some of the Oberheim analogue machines of yesteryear — there are 10 source/ destination modulation channels that can be used simultaneously. The oscillator section comprises three tone oscillators with multiple waveforms, though there are also over 150 single-cycle waves taken from complex digital sources. These can be processed via two resonant filters, which may be set up in series or parallel, and there are distortion, chorus, phaser and flanger effects built in. The instrument features a super-fat unison mode, and there are various morphing features that can be used to create evolving sounds and textures. ESX24 MkII is a fairly straightforward, but at the same time powerful sampler. It can read samples in EXS24, Akai S1000 or S3000, Samplecell, Recycle, WAV, AIFF, Giga, Soundfont 2 and Vienna Symphonic Library formats. It also has the ability to stream samples from disk. It operates via just two windows, the front panel and an editor, and though it doesn't have the fancy drag-and-drop facilities of some of its competitors, it is solid and easy to use. It's also simple to copy and paste layers of samples between programs. It now supports multiple outputs as well, although the way it does this isn't particularly elegant. The processing section is much like that of an analogue synth, with switchable resonant filter types, separate level envelope and filter envelope modulation, distortion and a very serious modulation matrix. Non-EXS24 sample material must be imported and converted to EXS format prior to use, so having a large hard drive to house your sample library is a good plan. EVP88 uses physical modelling to recreate the sound and performance dynamics of classic electric pianos, and is inspired by models such as the Fender Rhodes and Wurlitzer EP200. Its simple interface makes it easy to choose a piano type, adjust the basic elements of the sound and add the essential electric piano effects such as overdrive, chorus and phasing. The general consensus is that EVP88 provides a very good emulation of these classic instruments, and it also exhibits a good response to playing dynamics, which is what you miss out on with sample-based emulations. Like EVP88, EVD6 uses physical modelling, but this time its mission is to emulate the sound of the Hohner D6 Clavinet and similar instruments. It has a 60-key range and goes as far as to emulate the string buzzes and key clicks of the original. However, it goes much further than impersonating the classics as the user has plenty of control over the key parameters, including the imaginary
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Emagic Logic Pro & Express
pickup position beneath the imaginary strings. By deliberately taking the sound away from that of the Clav, you can coax a whole range of ethnic and abstract sounds from this instrument as well as some very passable harps. EVB3 is another physically modelled instrument, this time seeking to emulate the Hammond B3 drawbar tonewheel organ. It has an authentic-looking control surface, complete with drawbars, and also features the overdrive and rotary speaker effects associated with the original as well as additional effects, such as wah. Comparisons between EVB3 and Native Instruments' B4 are inevitable, and in many ways, they're just slightly different (and both excellent) takes on the same concept. The EVB3 effects, including its wonderful rotary speaker, can be used as separate insert plug-ins on other sources if necessary.
For Future Reference This isn't a review in the traditional sense, because very little has changed since last time we reviewed Logic other than the way the products are being sold and packaged. However, that doesn't mean I'm going to let Emagic/Apple off seeing a list of features that I feel need fixing or amending in subsequent revisions of Logic — though I can perfectly well understand why few changes have been implemented in the present version. I was very pleased when Project Manager was added, as I'd always felt Logic handled audio projects in a rather sloppy way, but now it's here, it still doesn't allow users to start new job as a new project and to know that all their files will be stored in one place. There's also the issue of linking and Instrument windows. At the NAMM show, I saw three different Emagic demonstrators trying to adjust a plug-in window, only to find that the window they were working was related to a different track to the one they thought they were working on. As linking works so well elsewhere throughout Logic, it would be nice to see an option that brings up the plug-in instrument window for the currently selected track when the Link button is on. It would also be useful to have software instrument icons come up as the track icon by default. On a more fundamental note, the surround support provided in Logic still doesn't allow the balance between the centre speaker and the front left and right speakers to be adjusted, yet this is something all surround mix engineers need — sometimes you want a centre-panned sound to come out of the centre speaker, sometimes you want it to come out of the left and right speakers, and sometimes a mix of the two would be good. An additional slider next to the LFE slider would fix this. And finally, not because it's my last comment, but because I'm rationing myself... would it be possible to build in a master fader for Logic that comes post the Bounce mix levels, so that we could adjust the level of any externally connected active monitors (where no mixer is being used) without affecting the mix level being bounced within Logic? This is particularly relevant with surround monitoring, as a decent surround volume control costs almost as much as Logic Pro does! Actually, you can now email us with your own comments about what you'd like to see fixed or changed in Logic because we have in place an arrangement to file:///H|/SOS%2004-05/Emagic%20Logic%20Pro%20&%20Express.htm (7 of 8)9/22/2005 6:07:46 PM
Emagic Logic Pro & Express
forward these to the Emagic design team — who will look at them. The only stipulation is that you limit your list to five features and never use the phrase "And another thing!"
Pricing & Upgrades If bought independently, Logic Pro costs £699 and Logic Express £199. Existing users of Logic Gold or Platinum versions 5 or 6 can upgrade to Logic Pro for £149, while for those with version 4 or earlier, the upgrade will cost £499. Logic Express and Logic Audio Big Box users can move up to Pro for the same price. There is no way to upgrade to the Express version. Prices include VAT.
Summing Up Leaving aside my personal feature wish-list, the outcome of these changes is that there will be many more winners than losers in the Logic upgrade market, and perhaps it is because the XS key has been so effective against piracy that Emagic have been able to drop their prices to this extent? The luckiest people are those who have only a bog-standard version of Logic Gold version 5, as they can go straight to Logic Pro without passing Go, paying only £149 into the Community Chest. However, anyone without Space Designer who has the computing power to run it should jump at the chance to upgrade. There will also be new users who can now afford to buy seriously high-end sequencing and audio software for less than ever before, which as Ian Cullen at Sound Technology pointed out to me, leaves them with more money in their pockets to buy a faster Mac to make ther best use of it all. Equally, the affordability of Logic may tip the balance for those Windows users who'd like to switch to a Mac/Emagic combination but who were previously dissuaded by the cost. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Eventide Reverb 2016
In this article:
Eventide Reverb 2016
A Leaf Out Of The History Digital Reverb Processor Book Eventide Reverb 2016 Plug- Published in SOS May 2004 in Print article : Close window The Reverb 2016 Reviews : Effects Using The Rotary Encoders Reverb Presets The Classic Sound
Eventide Reverb 2016 £1798 pros Elegant user interface. Faithful recreation of classic algorithms. Useful enhancements to algorithms. Flexible I/O.
Eventide's classic SP2016 reverb unit has been recreated and enhanced by the original designer. But how does the sound match up to that of the high-spec multi-effects units of today? Hugh Robjohns
Although most reading this will be Photos: Mark Ewing aware of Eventide and its range of Algorithms can become sophisticated digital effects processors, unstable. few — if any — will have come across Case is wider than most, Princeton Digital before. This new and won't fit many racks. company was founded a little over two summary years ago by Tony Agnello — but over A very elegant recreation of 25 years ago he was the Eventide an early digital reverb which engineer who developed the original became a classic. The reverbs all work extremely algorithms for the still highly regarded well in the context of a mix, SP2016 reverb processor, as well as and are surprisingly complex the software in the original Harmonizer. and lifelike. The 'one knob per function' user interface makes The foundation research began in 1978, and the original SP2016 — one of the first digital reverbs — was launched in 1983. It quickly became very popular, it a joy to tweak a preset program, and the displays partly because of the naturalness of its reverb programs, but also because it was provide very clear status so controllable and immediate — qualities sadly absent from so many modern information. Considerably hardware reverbs. more reliable and with greater cons
fidelity than the original, although it is still possible to find an unstable algorithm occasionally.
A Leaf Out Of The History Book
information £1797.75 including VAT. HHB Communications +44 (0)20 8962 5000. +44 (0)20 8962 5050. Click here to email
The SP2016 — a 3U rackmount box with vertical faders to adjust the parameters, chunky '80s buttons, and a small LED alphanumeric display — was fairly crude by modern standards, but three of its algorithms have stood the test of time: Stereo Room (with source-related stereo early reflections), Room Reverb (with a
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Eventide Reverb 2016
mono input) and High Density Plate. These reverb programs all sounded far more natural than any other reverb technology could manage at the time (plates, www.eventide.com www.princetondigital.com springs, tape echoes, and tiled rooms) and they always seemed to sit well in mixes. Furthermore, all their parameters were instantly adjustable, which was revolutionary. The SP2016 reverbs sounded quite distinctive, but they were also Test Spec very believable thanks to the way the unit generated a surprisingly complex earlyEventide Reverb 2016 OS reflection pattern, complete with a realistic build up of reverb density over time, v1.0.7 with PLD software combined with a smooth and natural decay tail. www.hhb.co.uk
v1.0.1.
Agnello formed Princeton Digital to apply the advances in DSP technology to the reproduction of high-quality, modern versions of some of those 'legendary' Eventide effects. The first results of this work are now available both as a bespoke hardware unit — the topic of this review — and as a software plug-in. The original SP2016 digital reverb ran on a purpose-built 'array processor' — the state of the art twenty years ago. This approach wasn't unique to Eventide; DSPs as we know them today were simply not available, and microprocessors were incredibly basic compared to those available now. Consequently all of the early digital reverb manufacturers — including AMS (the Ambience program on the RMX16 is still one of the best) and Klark Teknik (the DN780 remains one of my favourite reverbs) — had to adopt variations on gate array hardware techniques to achieve the sophisticated digital processing required. Although it may seem surprising, current DSP technology has only recently reached a level where it can match the computational complexity of those early bespoke arrays. So the Reverb 2016 employs a 120MHz DSP to do the number crunching, making it possible not only to recreate the SP2016's original algorithms, but also to take them a stage further, beyond the limits imposed by the original array-processor hardware. As a result, the new Eventide Reverb 2016 contains not only the original algorithms, but also new, more sophisticated versions of each of them. The 'enhanced' facilities include the improving of the Diffusion parameter — it was rather limited and subtle on the original, and is now far more expansive and useful. There is also provision of Diffusion and EQ controls to the Room and Plate algorithms, and the Position control is active in the Plate algorithm too. The high EQ response has also been tweaked slightly to give a brighter sound.
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Eventide Reverb 2016
Eventide Reverb 2016 Plug-in Princeton Digital have also produced a software-only plug-in version of the Reverb 2016 for the Pro Tools HD platform. Not surprisingly, the plug-in is also called Reverb 2016 and provides a rather stylish graphical user interface which replicates the look and feel of the hardware unit. The reverb algorithms are also apparently identical, which explains why it is also pretty DSP hungry! A free fullyfunctional 30-day demo version is available from the Princeton Digital web site and the full version can also be purchased on-line.
The Reverb 2016 The Reverb 2016 is a 1U rackmount box, with black-painted metalwork throughout, including the front panel. It measures roughly 205mm behind the rack ears, making it an easy fit (in terms of depth) into a flight case. However, the main part of the box measures about 449mm in width — the standard is about 435mm — and that means it won't actually fit in many racks or cases because there simply isn't sufficient space between the rack strips. This really is a case of try before you buy, because you'll be very frustrated if it won't slot into that rack space you had in mind! The rear panel is straightforward but well equipped. As the 2016 is capable of true stereo-in/stereo-out processing using certain algorithms, it is provided with two channels of inputs and outputs. The analogue I/O is catered for with both XLRs and TRS sockets, switched globally for balanced +4dBu or unbalanced 10dBV operation via a rear-panel button. The converters are standard 24-bit designs, but the sampling rate is not standard. The original SP2016 ran at 40kHz — typical of the technology available in the era when it was conceived — and the new version remains faithful to that. As a result, the frequency response of the processed signal is limited to 16kHz (±1dB) at the high end. The dynamic range is quoted as greater than 98dB ('A' weighted) and distortion is below 0.006 percent. The D-A output is capable of driving a peak output level of +24dBu. It is interesting to note that, although the digital processing and sample rate (as already mentioned) are identical to those of the original, the analogue interfaces are radically different, particularly in respect of the A-D and D-A stages. The originals were crude 16-bit devices with early brick-wall anti-alias and reconstruction filters which mangled the high-end response with amplitude errors and horrendous phase shifts. The new design uses modern 24-bit oversampling filters which provide a far better dynamic range and a smoother, more open highend sound character. file:///H|/SOS%2004-05/Eventide%20Reverb%202016.htm (3 of 8)9/22/2005 6:07:50 PM
Eventide Reverb 2016
A pair of phono (RCA) connectors provides co-axial S/PDIF digital audio in and out, and a pair of MIDI sockets offer In and Out ports. The last socket is another TRS type which allows an optional momentary-action footswitch to either 'kill' the reverb input (in other words stopping the reverb send signal, but allowing the existing decay to continue) or bypass the unit completely. Just to finish off the rear panel, on the extreme left (as viewed from the back) is an IEC mains socket for the integral switched-mode power supply (SMPS) which accepts any mains voltage between 90V and 240V, at 50Hz or 60Hz. There are no user-accessible fuses, and the unit consumes 45W of power. The mains power switch is on the front panel at the extreme right-hand side. The front panel is plain and simple, but arguably the more attractive for it. A pair of simple input-level LED bar-graph meters occupy the leftmost part of the control panel, showing signal levels above -40, -20, -6 and 0dBFS, with an additional pair of lights to indicate overload of the A-D converters. A pair of simple black rotary controls set the analogue input and output levels, and a yellow LED adjacent to the input control illuminates when mono algorithms are in use. In this case, the two input signals are internally summed to mono before being passed to the reverb processor. Running above the meters and these control knobs are three black buttons, each with an associated indicator LED. The first selects the S/PDIF digital input, while the others provide manual control of the features operated by the optional footswitch — the reverb input 'kill' and bypass functions.
Using The Rotary Encoders Most of the other front-panel controls are rotary encoders, all but one with a ring of LEDs around the base to show the current setting. These provide real-time control of the wet/dry mix (zero to 100 percent), pre-delay time (0-999ms), decay time (maximum varies with program), notional distance from the source (which affects pre-delay, diffusion, early reflections, frequency response and more besides!), diffusion, and high and low EQ. All of the encoders have an over-press facility, and in all but the two EQ knobs, pushing the encoder causes its numerical value to be shown for about five seconds on the LED display. (The numerical value is also shown while the knob is being adjusted.) The LED displays are different for each parameter, but selected to show the relevant setting as clearly as possible. For example, the Mix control is shown as a solid block of light from the zero position at seven o'clock around to the current Mix level. The Pre-delay is shown with a dual pointer, one part circling faster than file:///H|/SOS%2004-05/Eventide%20Reverb%202016.htm (4 of 8)9/22/2005 6:07:50 PM
Eventide Reverb 2016
the other. The first shows blocks of 30ms, while the other steps for singlemillisecond increments. A nice operational feature here is that, by pressing and turning the knob simultaneously, the pre-delay is adjusted in 30mS steps, while simply turning the knob normally allows 1ms step precision. Both the control operation and display are unusual, and take a little getting used to, but the combination provides great precision over a huge range of values. The Decay control value is shown in the same way as that of the Mix control (using a solid block of lights), while the Position control shows its setting with a single line. Diffusion is shown as another block of lights, but this time building outwards symmetrically to either side of the 12 o'clock position. The EQ controls are slightly unusual. The low band provides between +4dB and 8dB of gain centred on any one of ten frequencies between 50Hz and 500Hz (in 50Hz increments). The high band only provides attenuation (up to -8dB) at frequencies between 1kHz and 8kHz (in 500Hz increments). For both controls the gain is adjusted by rotating the knob in the usual way, while the turnover frequency is adjusted by pressing in the knob while turning it. This action is easy enough with the unit mounted in a rack, but the pressure required is such that you can easily end up pushing the box all around the table if it is left free standing! It is also worth noting that applying too much low boost with a long decay time can cause runaway feedback within some programs.
Reverb Presets To the left of the numerical display window is a column of four LEDs and a button used to select the reverb algorithm. Stereo Reverb programs are shown by a red LED, Room Reverb programs are yellow, High Density Plates are green, and the three new algorithms (stereo, room and plate again) are shown in combination with a blue LED. Pressing the button repeatedly simply cycles through the modes. Under the three-digit LED display window is another pair of buttons, each with its own indicator LEDs. The first accesses the System menu — of which more in a moment — and the second is used to save the current settings to a preset memory. The last encoder, mounted to the right of the display window, enables presets to be selected for storing or recall. There are 99 preset memories, all initially loaded with factory programs, and any can be overwritten with user settings if required. Presets 0-59 are all configured with no dry signal at all, whereas presets 60-99 are identical programs with an appropriate wet/dry mix. The presets are organised in groups of ten effects, based on each of the six algorithms (the three originals and the three new variations). The effects in each group gradually increase in scale from small to large environments. Pressing the encoder knob loads the selected preset, at which point the indicators around all the parameter encoders change to reflect the new settings. Any program can then be further adjusted by manipulating the relevant encoder. To save a preset you simply have to press the Save button, dial up the required preset number, and press Save again. file:///H|/SOS%2004-05/Eventide%20Reverb%202016.htm (5 of 8)9/22/2005 6:07:50 PM
Eventide Reverb 2016
The System button mentioned earlier provides access to the machine's configuration parameters. Pressing the button repeatedly causes the display to cycle through the various options, starting with the input source (both analogue and digital). Options here are input one, input two, or both — enabling the use of a single-source signal even for the stereo programs. Next is the MIDI channel (Off, 1-16, or All). The next two modes are labelled dC (Dump Current) and dA (Dump All), which provide a MIDI data dump of the currently selected program or of all programs. The Save button also flashes while in this mode, and if it's pressed the data is output via MIDI to an external data filer or sequencer, for example.
The Classic Sound The sound of the Reverb 2016 is pretty magical. There is something about the way this neat box generates reverb — and more importantly, the way that its reverbs sit in a mix — that makes this a great machine. To be honest, the SP2016 was not a processor I spent much time with in my early audio career — I'm more familiar with the AMS and Klark Teknik reverbs from the same era. Nevertheless, I can certainly appreciate the quality of these algorithms, and the benefits afforded by Eventide having to build the hardware necessary to implement them. Pretty much all subsequent digital reverberators worked the other way around — algorithms had to be created (or compromised) to run within the available hardware structure, at least until affordable DSP power reached a certain level, after which point there was an explosion in the number and sophistication of really good-sounding digital reverbs. The Reverb 2016 is certainly very intuitive to use — when I first opened the box I thought the manual had been omitted, so just started playing with the unit and found it very easy to master. Only later did I discover that the two sheets of A4 notes tucked in with the machine really are all that is needed to fully understand it. Compare that to the 100-page booklets that usually accompany Yamaha, TC Electronic or Lexicon reverbs! The controls feel perfect, the displays are simple but clear and informative, and the presets all provide useful starting points. Having said that, this box is so easy to fine-tune by ear that, after an initial survey of the presets, I didn't use them again! I preferred instead to quickly recall the appropriate algorithm and dial in the necessary parameters. Unless you can remember the specific presets appropriate to a particular task, it's just as quick to configure the unit manually, and a lot more fun! It seemed a little odd that some of the parameter controls were blank and inoperative for certain algorithms — until, that is, I read the notes, which explained that the original machine didn't include EQ or Diffusion controls for the
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Eventide Reverb 2016
Room and Plate algorithms. I guess that's the whole point about faithful recreations of existing products. However, I was equally impressed that these facilities are provided when using the 'New' versions of the algorithms — the turbo-charged updates — which also have other subtle tweaks and refinements to ensure the reverb quality and versatility matches the best of the current products in the market. The High Density Plate algorithm is probably my favourite, although the Stereo and (mono) Room programs are also very good and extremely usable in the appropriate contexts. I found that recalling some presets occasionally resulted in nothing useful at the output — just an unstable kind of whistling noise. Often changing the preset and then going back to the required one again cured the problem, which seemed completely unpredictable. Whether this is part of the faithful recreation (there are several warnings about inherent algorithmic instabilities) when changing presets in the presence of audio signals, or a bug in the current operating software, I don't know. However, although a little frustrating at first, it became largely irrelevant when dialling in parameters from scratch. Of all the parameter controls, the Position knob was probably the most heavily used parameter for all the programs apart from the original Plate algorithm (where it is not available). This control really does simulate moving closer to, or farther from, the source, changing diffusion, early reflections, wet/dry balance, EQ and so on. It's just the job when everything else seems well balanced but its all just a bit too close or distant! The New variants of the original algorithms retain most of their inherent character, but add slightly more controllability and a subtle extra richness or reduced coarseness, depending on your point of view. In other words, it could be said they sound a little more modern than their classic siblings. However, both versions of all three algorithms were eminently usable across a wide variety of music genres, always seeming to sit perfectly in the mix rather than on it, bringing space and perspective without acting like acoustic Polyfilla — a common failing of lesser reverbs. The original algorithms could, at times, reveal their '80s heritage with a characteristic coarseness (especially if auditioned in isolation during a solo-in-place check, for example), but this rarely became noticeable within a mix. Maybe there's something here for other reverb manufacturers to take heed off. The real-world precision of modern high-spec reverbs and convolution systems may not always be the most musically appropriate choice, especially for pop mixes. The Reverb 2016 is expensive when compared to the glut of very high-quality and versatile reverbs and multi-effects processors now available in the UK. However, it does offer a unique sound quality that works wonderfully in the context of a mix, and is fantastically ergonomic to use. If the budget can stand it, this is definitely worthy of close personal attention — but measure your rack space first to avoid disappointment! Published in SOS May 2004
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Eventide Reverb 2016
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All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Gmedia ImpOSCar
In this article:
Gmedia ImpOSCar
Getting Started The Oscillators Virtual Instrument Plug-in: Mac & PC Additive Synthesis Published in SOS May 2004 OSCar Overdrive Print article : Close window Filters, Envelopes & More... Modulation & Performance Reviews : Software Facilities The Effects MIDI The ASG Realised? Paul Gmedia continue their quest to render the best of the Wiffen Writes... world's keyboards in software form. But will it give In Use OSCar's fans a case of the grouches? Conclusions
Gmedia ImpOSCar £100 pros Not just an OSCar... it fulfils much of the promise of OSC's unreleased ASG polysynth. It offers additional features in all areas of synthesis, such as velocity sensitivity, the simple but useful effects section, and the 16-voice polyphony. It generates a huge range of analogue, digital, and hybrid sounds. I've been hammering it for over a month, and it hasn't crashed yet!
cons There's no stand-alone version available yet. There's no aftertouch. The pitch-bend response is slewed too slowly. Saving to a new location and compiling banks of patches from an existing library is rather convoluted.
summary ImpOSCar is an excellent attempt to recreate the sound and feel of the original OSCar. However, its hugely enhanced facilities make
Gordon Reid
Some people feel that the OSCar is overrated. Its operation seems obtuse, and its lack of a screen or dedicated programming controls makes it feel oversupplied with features. This is barely ameliorated by the fact that most of it is quite straightforward once you get the hang of it, and that nowadays you're unlikely to use its more arcane features. I also rate the OSCar as one of the ugliest synths ever designed, and its construction, while distinctive, is not well thought out. Three aluminium rods turn the two rubber end-cheeks and the guts into a chunky synth sandwich, and if the bolts holding these rods work a little loose, the stresses on the board connections and the panel hardware are almost certain to lead to failures. Then there are the horrible buttons, and pitch-bend and modulation wheels that are partially obscured by the left-hand end cheek... the list goes on and on. If that weren't bad enough, there's the price, which now ranges from £700 to £1400. Given that you could start equipping a home studio for the upper of these, the OSCar seems extraordinarily over-valued. On the other hand... had Mr Spock been presented with an OSCar (the synth, not the award) he might have raised a single eyebrow and said, "Fascinating". Once you get used to the controls, it becomes very playable, and its combination of digital oscillators, primitive additive synthesis and an unusual filter topology give it a character that makes it stand out from the crowd. Sure, it can create all manner of Moog-y imitations and, perversely, it was for its bass and lead sounds that it eventually became known, but the OSCar is also capable of a huge range of delicate and brittle sounds that you could never obtain from a purely analogue monosynth. Oh yes... and it sounds gorgeous (for more on the OSCar, check out the SOS Retrozone feature at www.soundonsound.com/sos/sep99/articles/oscar.
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Gmedia ImpOSCar
head-to-head comparisons with the original synth a little pointless, so I prefer to view it as a powerful, polyphonic softsynth rather than as a slavish imitation of the OSCar. Highly recommended.
information £99.95 including VAT. Time + Space +44 (0) 1837 55200. +44 (0)1837 55400. Click here to email
htm). So, given my dislike of the OSCar's body, but my fondness for its musical soul, I was intrigued when I heard that Gmedia were developing a software synth based upon it. I thought their Mtron imitation of the Mellotron was a brave attempt at recreating the flavour of an almost inimitable original, and I also rated their Oddity plug-in. I therefore hoped they would prove to be the right company to undertake the non-trivial task of recreating this quirky and unusual synthesizer.
Getting Started
www.timespace.com
Test Spec 1GHz Apple G4 Titanium Powerbook with 512MB of RAM running Mac OS v10.2.8, and Plogue Bidule. ImpOSCar version reviewed: v1.0.
ImpOSCar arrived with a brief but Photos: Mark Ewing beautifully produced manual and a CD containing four versions of the software: Mac OS 9.x VST, Mac OS X VST, Mac OS X Audio Units, and PC (Windows 98, SE, ME, 2000 and XP) VST. Despite this cornucopia, I was rather disappointed to find no Mac stand-alone version (which, I understand, is planned for a future Like it or hate it, the original OSCar was release at an as-yet unspecified time). certainly distinctive! I'm a MAS man (ie. a user of MOTU's Digital Performer), and tend not to use VST applications, so a stand-alone version can eliminate many headaches. Anyway, I was unable to use ImpOSCar on my studio's G4 tower because it does not support VSTis, so I loaded the Mac OS 9.x version onto a 1MHz G4 OS X Titanium Powerbook on which I keep a couple of OS9.x VSTi hosts. Everything seemed to be going well until I played a few notes on my MIDI controller keyboard — there was a huge latency, possibly caused by a mismatch between the OS X drivers and the Classic environment hosting my VSTi software. Fortunately, when we contacted Gmedia to resolve the problem, they had the answer ready, in the form of a Gallic VST host entitled Plogue Bidule (French, I am told, for 'Plug-in Thingy'). I downloaded the program from http://plogue.com, spent a while getting everything to talk to each other, and... bingo! It worked beautifully (see screenshot on the next page). With an original OSCar to my left, and the Powerbook to the right, it was time to let battle commence. At first sight, ImpOSCar seems to be an accurate recreation of the OSCar, but all is not what it seems. With the removal of the sequencer, the addition of neater Load and Save buttons, and improvements in the User Wave controls (which we'll discuss later) there is room on the panel for a fourth row of knobs. This has made it possible for Gmedia to extend numerous synthesis functions, and to add a bunch of new ones. So let's look at each of the sections in turn.
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Gmedia ImpOSCar
The Oscillators Much of the OSCar's fame is due to its unusual oscillators. These generate five conventional waveforms: triangle wave, ramp wave, square wave, pulse wave, and PWM. Since both the OSCar and ImpOSCar are digital, it's reasonable to assume that these waveforms will be all but identical, and so they initially seemed to be. Starting with Osc1, the triangle waves are almost indistinguishable from one another. Likewise, the ramp and square waves are similar to the original, although the OSCar's are a tiny bit brighter. After a few moments, I began to suspect that this was due to distortion in the OSCar's filter, so I backed the level off by a hair. Yep! That did it. Significant differences began to appear, however, when comparing the pulse waves. On the OSCar, the PW knob offers a clean sweep from square (50 percent) to a tight pulse. On ImpOSCar, the emulation was near perfect at some settings, while at others there was an unpleasant aliasing, which was particularly noticeable at high frequencies. This was not nice. Hosting a MIDISport 2x2 and ImpOSCar
The next setting is PWM, which — on within Plogue Bidule. both the OSCar and ImpOSCar — offers a single parameter: depth. On the original synth, a dedicated LFO produces the modulation, and turning the PW knob from fully anticlockwise to fully clockwise provides a smooth transition from a static waveform to a rich chorus. This is as true at high frequencies as it is at low ones. However, ImpOSCar displayed an alarming artefact at any pitch higher than about an octave above middle 'C'. This manifested itself as two unrelated frequencies going 'dee-daa dee-daa' at the modulation rate. When SOS made these problems known to Gmedia, they had an updated version back to us within 48 hours, and following the update, everything worked as it should. The pulse wave is now, in my view, indistinguishable from the OSCar's, and I can only tell the difference between the OSCar's and ImpOSCar's PWM waves because ImpOSCar's is the cleaner of the two! In addition to the quasi-analogue waveforms, ImpOSCar offers a further five preset waves derived from the OSCar: Full Organ, Harpsichord, Strong Lead, Double Pulse, and Gritty. But perhaps the greatest trick up the OSCar's electronic sleeves is its primitive additive synthesis...
Additive Synthesis
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Gmedia ImpOSCar
Without a manual, it is perhaps unlikely that you will stumble across the original OSCar's additive synthesis, since programming it involves an arcane combination of knobs, buttons and the keys themselves. But it can be done, and the results have a brash quality that allows the OSCar to explore sonic territory forever inaccessible to purely analogue synths. So, in addition to the 10 preset waveforms, ImpOSCar offers two further settings — User Wave 1 and User Wave 2 — that allow you to create patches based on either of two waveforms that you program in the grandly named User Wave Matrix. (shown left). Click on either 'W1' or 'W2' on the Edit Mode knob, and a representation of an LED matrix will appear in place of the Creating a user pitch-bend and modulation wheels. Within this, you can Wave. drag the amplitudes of the first 24 harmonics up and down, doing so in real-time so that you can hear how your changes affect the timbre of the sound. As Paul Wiffen notes elsewhere in this article, this is little different in principle from the method employed on the OSCar itself, but the clarity of the representation, and the ability to see what you've done, makes something that was previously haphazard into something clear, straightforward and useable. Once you have created a User Wave, you can name it, store it, and recall it as shown in the screenshot on the next page. And, although it may not be immediately apparent, Gmedia have been rather clever here. If you insert a User Wave — say, 'Gordon's Wave' — into two patches, and then edit the harmonic content of one of them, you do not damage the other patch. This is because the patches contain the actual harmonic data; they do not use the name to look up that data within a library. So, instead of wrecking your patches, all that happens when you change the contents of 'Gordon's Wave' in patch 1 is that it becomes 'Unnamed' in patch 2, and the sound remains unaffected. Nice!
OSCar Overdrive Unlike on ImpOSCar, where they have separate dedicated controls, the OSCar's filter overdrive and main volume were both accessed via the same control, the main volume knob — although they were still independently adjustable. To set the filter drive, you moved the volume control while holding down the Store button. This is at variance with the operation of the control as described by Paul Wiffen in his SOS OSCar retrospective, but following recent discussions with Chris Huggett, Paul has agreed that time had taken its toll on his memory of this when he wrote the SOS retro! This explains, of course, why the Volume control on the OSCar's front panel has the words 'Filter Drive' printed underneath, with an arrow pointing to the Store button (see right).
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Gmedia ImpOSCar
Filters, Envelopes & More... Once you have determined your oscillator settings — preset or user-defined waves, octave and so on — you can select Osc2's octave and/or semitone transposition, apply detune, balance Osc1 against Osc2 in the Mix section, and balance the combined sound against a white-noise generator. That's all as it should be and, in the case of Osc2's tuning, much simpler than the method used by the OSCar itself. The signal then passes to the filters (shown below). The OSCar offers a pair of analogue two-pole filters that you can configure as two low-pass 12dB-per-octave filters in series, as a 12dB-per-octave band-pass filter, and as two 12dB-peroctave high-pass filters in series. Moreover, the cutoff frequencies of the two devices need not be the same; a Separation control allows you to space them by anything up to a factor of four. This means that you can use the lowpass mode as a traditional 24dB-peroctave low-pass filter, or as dual 12dBper-octave low-pass filters with two ImpOSCar's oscillator section. 'knees' in the cutoff profile. Alternatively, you can determine the broadness of the band-pass mode, use the high-pass mode as a traditional 24dBper-octave high-pass, or as dual 12dB-per-octave high-passes, again with two 'knees' in the cutoff profile. On the front panel, the OSCar provides six filter modes: low-pass, high-pass and band-pass, each with or without keyboard tracking, which is nominally 100 percent. In addition to all of this, it offers overdrive of the filter input (see the 'OSCar Overdrive' box on the previous page for more on this). ImpOSCar provides all this, but goes further by adding six modes that you could have obtained on the original instrument only by violently molesting the filter circuits with a soldering iron. The nine ImpOSCar modes (of which the first three are equivalent to the OSCar's) are shown in the table below. These modes would have little significance were it not possible to determine the filters' cutoff frequencies independently. However, despite Gmedia's claim that this is unique to the OSCar and ImpOSCar, a similar configuration also appeared on the Crumar Spirit, which was also released in 1983, and which produces a similar range of powerful sounds and 'formant'-style timbres.
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Gmedia ImpOSCar
IMPOSCAR FILTER MODES OSCAR MODES Low-pass 24dB-per-octave Dual low-pass filters in series. Band-pass 24dB-per-octave Dual band-pass filters in series. High-pass 24dB-per-octave Dual high-pass filters in series. PARALLEL MODES Low-pass two-pole
Dual low-pass filters in parallel.
Band-pass two-pole
Dual band-pass filters in parallel.
High-pass two-pole
Dual high-pass filters in parallel.
MIXED MODES Low-pass/band-pass
A low-pass and band-pass filter in parallel.
Low-pass/high-pass
A low-pass and high-pass filter in parallel, creating a bandreject filter.
Band-pass/high-pass
A high-pass and band-pass filter in parallel.
Instead of offering just 0 percent and 100 percent filter/keyboard tracking, ImpOSCar offers a Keytrack knob that allows you to select values between -200 percent to +200 percent in steps of 10 percent. Likewise, instead of offering a single knob for volume and filter overdrive, ImpOSCar has two: one a conventional loudness control, the other an overdrive control. Turning both to maximum creates the fat sounds for which the OSCar is best known, but you can also use the two knobs on ImpOSCar to create quiet, driven sounds, and loud, clean sounds. Like the original, ImpOSCar has a dedicated ADSR contour generator for the filter, with an Amount control that determines the depth applied to the cutoff/ centre frequencies, and a Delay control that allows you to delay the onset of the filter envelope by up to two seconds after you depress a key. The last of these improves upon the Delay function on the original, which you obtain by turning the Sustain to 'zero', and then determining the length by adjusting the Release. Just above the filter contour generator, the OSCar and ImpOSCar both offer a dedicated VCA contour generator (shown above). Gmedia claim a fastest response for both ImpOSCar envelopes of 1ms, and a slowest of between 15 seconds and 60 seconds, depending on which knob you're twiddling. However, I wasn't sure that the minimum settings were getting close to 1ms. All other things being set correctly, you can set the filter and/or amplifier envelope Attack and Decay to their furthest anti-clockwise positions, whereupon you should — if the
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OSCar front-panel detail, showing the arrow connecting the Volume and Store button. The legend 'Filter Drive' is just visible under the volume control.
Gmedia ImpOSCar
specifications are correct — obtain a precise click lasting just 1/500th of a second. In contrast, the OSCar and ImpOSCar seemed to produce a brief squelch; faster than some 1980s hybrid synths, but not in the millisecond range. I decided to put this to the test by sampling numerous ImpOSCar clicks and measuring their duration. The results were very interesting and, to give ImpOSCar its due, the VCA envelope's transients ranged from 0.4ms, which is astonishingly fast, to around 3ms, which is still respectably rapid. The tails of the clicks lasted somewhat longer; from almost non-existent to around 30ms at their longest. On the other hand, if I opened the VCA fully by setting its sustain to maximum, I was able to measure the filter response, which was much slower. This time, there was an initial transient lasting around two milliseconds, but this was followed by a tail extending for up to 160ms, and I suspect that this is what I was hearing. I checked this with a range of filter drives and volume settings, just to make sure that I wasn't overloading my A-D converter, or anything else in the signal chain, but the results were consistent. Like the OSCar, ImpOSCar has numerous triggering and gate modes, and faithfully recreates the OSCar's single- and multi-triggering, external triggering and variable gate time, as well as the envelope repeat modes that work so well with its simple arpeggiator. Of course, Gmedia have extended the principles of the original, in particular expanding the voicing from the OSCar's monophonic and Loading a user Wave. duophonic modes to include four-, eight-, 12-, and 16-voice polyphony. You might think that this would detract from the essential OSCar-iness of ImpOSCar, but I don't think that it does. I prefer to think that it makes ImpOSCar a descendent of the Oxford Synthesizer Company's fabled polyphonic OSCar, the ASG (for more on this, see Paul Wiffen's box later in this article). Given the hushed tones in which people sometimes refer to the ASG, this can be no bad thing. Finally (in this section) where ImpOSCar now offers the Function knob that controls the polyphony, the OSCar has a knob that activates a 'Hold' function and a basic arpeggiator (see above). ImpOSCar has moved these toward the lower left of the panel, regrettably detaching them from the Triggering and Tempo controls that are explicit to their operation. On a more positive note, the arpeggiator now offers up, down, up/down and random modes, and you can synchronise it to the internal clock or to MIDI.
Modulation & Performance Facilities file:///H|/SOS%2004-05/Gmedia%20ImpOSCar.htm (7 of 15)9/22/2005 6:07:56 PM
Gmedia ImpOSCar
As with everything that has gone before, the LFO (below) expands upon the facilities of the OSCar, upping the original six waveforms to 12. These include 'stepped' waves, a couple of patterns, a unipolar 'guitar' wave for realistic vibrato effects, and a random setting. As on the OSCar, you can direct the LFO to the filter cutoff The filter section. frequency and oscillator pitch, but — inevitably — the external synchronisation is way beyond what was possible in 1983. There are four options. Firstly, free-run allows the LFO to, umm... run freely. Not an obvious benefit when playing ImpOSCar as a monosynth, the use for this becomes apparent when playing polyphonically, because the LFOs in each of the voices drift in and out of sync with one another. Secondly, Lock does what you would expect; it locks the polyphonic LFOs together. Next, Clock synchronises the LFO to the internal clock driving the arpeggiator, whereas MIDI bypasses the internal clock, and synchronises the modulation to MIDI Clock, if present. If you set the Sync to MIDI, but do not provide a MIDI Clock, you may experience an unpleasant 'hash' as the LFO goes hunting for a synchronising source. I obtained this on the OS 9 version, but not the OS X version, although the problem was my fault for setting everything up incorrectly, not ImpOSCar's. In addition to LFO modulation, you can direct the filter envelope (Env2) to the oscillator pitch. You might think this a strange addition to the LFO section, but that's as nothing compared to the 'Env' option in the LFO itself. This allows you to use Env2 in conjunction with the Gate knob to create new LFO waveforms, the rate of which is determined not by the LFO Rate control, but by the Tempo knob (ie. the internal clock) in the envelope section. Weird! The original OSCar was neither velocitynor pressure-sensitive, so I am happy that ImpOSCar allows you to control four aspects of the sound using MIDI velocity. These are the rate of the Attack and Release times of both contour generators, the amount of filter overdrive, and the amount of Env2 applied to the filter cutoff/ centre frequencies. While not a full complement of velocity-sensitive The amplifier, triggering, and polyphony parameters, these nonetheless facilitate a controls. goodly measure of expression. Unfortunately, aftertouch is conspicuous by its absence. Given that ImpOSCar is superior to the OSCar in so many other ways, it's a shame this wasn't added to Gmedia's version as well. Conventional 1970s' performance duties are fulfilled using the pitch-bend and modulation wheels, with the oscillator pitches and the filter cutoff frequency as destinations (the relevant controls are shown highlighted on the next page). file:///H|/SOS%2004-05/Gmedia%20ImpOSCar.htm (8 of 15)9/22/2005 6:07:56 PM
Gmedia ImpOSCar
Strangely, the pitch-bend response is slewed... flick the joystick from the centre to one extreme, and the pitch slides upward or downward over a second or two. Apparently, this was added to avoid audible zippering on the Mac version of ImpOSCar, but it limits the use of pitch-bend somewhat. Understandably, Gmedia are hoping to implement a faster response in a later revision. On a happier note, ImpOSCar's emulation of the OSCar's six portamento/glissando modes is already spot-on.
The Effects In addition to its extensions and improvements in other areas, ImpOSCar offers something completely new: a stereo effects unit. Accessed by clicking on the Panel Open message to the right of the instrument, this reveals 12 knobs and a button that otherwise remain hidden from view (see below). Together with the three 'panel' knobs and the effect On/Off button, these determine the nature of the available chorus and delay.
The arpeggiator.
Of the 16 controls (not including on/off), only three apply to the chorus. These, as you might expect, are Depth, Rate and Level. I rather like this chorus; it adds a subtle dimension to the sound without overpowering it. The delay unit is more extensive, with independent Length, Feedback, and Mix controls for the left and right channels, and Delay Level and Tempo knobs available on the main panel. In addition to these, there's a Gate knob, a Delay Units knob that sets the baseline for the Delay Lengths, and a simple Hi/Lo EQ that helps you to seat the delay in the mix. The final control, marked 'CF' is a button that switches between a cross-channel 'ping-pong' delay, and a conventional delay with the left and right channels treated independently. Of these, the Gate needs extra explanation, because its action is not immediately apparent. The LFO. When turned anti-clockwise from the 12 o'clock position, it suppresses the delayed signal while a direct signal is present. As explained in the manual, this is an old studio trick that lets the sound cut through without being cluttered by echo while you are playing, but which brings the delays to the fore when you stop, adding ambience at the end of solos and so on. Turned clockwise, the opposite occurs, and at its furthest clockwise extreme, you only obtain delay while you are playing. As the name suggests, this is like gating an echo unit using the direct signal, and it creates some interesting and unusual sounds. file:///H|/SOS%2004-05/Gmedia%20ImpOSCar.htm (9 of 15)9/22/2005 6:07:56 PM
Gmedia ImpOSCar
MIDI One of the things that made the OSCar special in 1983 was its patch storage. OK, there were only 12 memories, plus 24 factory sounds in ROM, but OSC later upgraded this so that all 36 locations were programmable (don't forget, this was non-trivial stuff in those days!). Perhaps in deference to this, ImpOSCar sound banks comprise 36 patches that you can access using the Load menu provided by your host software or, as I discovered, by dragging the FXB file over the image of ImpOSCar. The manual states that ImpOSCar will also load original OSCar sound banks via SysEx, and I was keen to make some direct comparisons of ostensibly identical patches. However, the utility that makes this possible was not available at the time of writing, so I was unable to do so.
RESERVED IMPOSCAR CONTROLLER NUMBERS CC# NAME
ACTION
64
Sustain
Holds all notes.
66
Sostenuto Holds all notes held at the moment CC66 was received.
67
Soft
Divide MIDI velocities by a factor of two.
68
Legato
Toggle between single- and multi-trigger modes.
69
Hold2
Multiplies release times by a factor of two.
123 —
All notes off.
Apart from the lack of aftertouch, ImpOSCar boasts an impressive MIDI specification, and you can control all of its parameters using MIDI CC messages. Many of these are pre-defined, but you can change any of them by turning the Edit Mode knob to 'CC'. Each knob and switch then displays a window containing the letter 'L' (for 'Learn') and the number of the currently assigned CC. If you click on the 'L', the next controller you send will be assigned to that parameter. Six CCs are, however, reserved, as shown in the table below. Finally, while in Edit Mode, you may also like to change the way in which the mouse turns the knobs. There are two options: Circular and Linear. Unfortunately, the Linear option refuses to work for me, and even prevented me from exiting the mode. I tested this using both the OS 9 and OS X versions of ImpOSCar, and found it to be true in both cases. I suspect that it may be an unwanted side-effect of the unconventional host software I'm using, but it's strange that it happens when running under both VSTi Host and also Plogue Bidule.
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Gmedia ImpOSCar
The ASG Realised? Paul Wiffen Writes... As I was paid to help design presets for Gmedia's ImpOSCar, there was no way SOS were going to let me review it. However, my close involvement with the sound design and marketing of the original OSCar (the reason, of course, why Gmedia wanted me involved on ImpOSCar) also meant that I was well placed to write some additional historical background to supplement Gordon Reid's review, as well as some subjective comments about ImpOSCar. Those who worry about such things should therefore be aware that everything I say here about ImpOSCar is against a backdrop of having been commercially involved with the design of the presets for the instrument — make of that what you will. The full story of my involvement with the original OSC project can be found in my September 1999 SOS Retrozone article (a web link to which can be found near the start of this review). I programmed 32 sounds for the OSCar back in 1982 on a work-in-progress development unit with no front panel and paper labels for each knob at designer Chris Huggett's house, and loading these same sounds into ImpOSCar last year was quite a spooky experience — it was as if a missing section of my life had suddenly been returned to me. Within minutes of encountering ImpOSCar, I was re-living the experience of programming one in a way I hadn't been able to do for nearly 20 years. In the OSC days, my demo OSCar was always being taken away to fulfil an order somewhere (I particularly remember serial number 1000 going to Vangelis when I had hoped to hold onto it) and although it always got replaced, it meant there was never one to call my own. More recently, I was given one by a kind soul for my 40th birthday, but unfortunately, I left it in the studio of another friend who has now emigrated to South Africa (if you're reading this, Jethro, please get in touch!). As Gordon says elsewhere in this article, ImpOSCar offers much more than the OSCar ever did, and the two big improvements in this regard really struck me when I demonstrated ImpOSCar for Gmedia at this year's NAMM show in January. When you're programming sounds, you don't notice polyphony so much, as you tend to be striking a single note over and over, and until you come to play for real in front of people, you don't realise how important velocity sensitivity is to performance. Those with very long memories might remember that there was to have been a velocity-sensitive polyphonic product from OSC, which was announced and even demonstrated by yours truly at the 1985 Frankfurt Musikmesse. Called the ASG (Advanced Sound Generator), it was housed in a rack to avoid the problems of having to develop a velocity-sensitive keyboard (that then-recent innovation, MIDI, meant that any other keyboard could be used to trigger it with velocity), and featured a large LCD display so that the additive waveform facility could be more easily programmed. As one of the few people ever to play the ASG, I can attest to the fact that velocity sensitivity and polyphony made a big difference to the sound of the OSCar and unsurprisingly, the same is the case with ImpOSCar. It reminded me of the first time I played the ASG just before we took it to its German launch — I thought at the time how much it changed the character of the OSCar to be able to trigger it polyphonically, with each voice changing subtly depending on how the velocity was routed to amplitude and filter. I always thought that the ASG had a very Oberheim-like quality which I never noticed on the monophonic OSCar (although the voice cards at the heart of both synths were exactly the same) and the same is true of ImpOSCar. Used this way, ImpOSCar is in fact the equivalent of 16 OSCars (£11,184 at file:///H|/SOS%2004-05/Gmedia%20ImpOSCar.htm (11 of 15)9/22/2005 6:07:56 PM
Gmedia ImpOSCar
1984 prices) or a fully loaded ASG (which would have been better value at its £2999 target price, had it ever been released). I suppose we should no longer be surprised that a virtual synth can outperform its original real-world inspiration, because all the physical requirements for each voice of polyphony in a hardware synth are sidestepped. Once the voice architecture has been recreated in the digital domain, then creating multiple voices is simply a question of assigning CPU power sufficient for as many voices as you want to trigger. This is a far cry from what happened with OSC's ASG, which housed up to 16 OSCar voice cards. I seem to remember that the plan was to offer it in an eightvoice version for around £1999 and a 16-voice version for £2999. However, to my knowledge, that 1985 Frankfurt show was the only time the prototype held a full complement of OSCar cards. The problem was that once we got back from the show to business as normal, we never had enough voice cards and Chris Huggett would have to keep 'borrowing' them out of the ASG to make up an order. By the time OSC ceased trading as a manufacturer, all the voice cards had been used to sell the last few OSCars, and the last time I saw the ASG prototype, it was mute, as all its voice cards had been cannibalised. At last, I feel that the concept has been fully realised. I particularly enjoy creating additive waveforms with ImpOSCar. Although the way that you built up waveforms on the OSCar was fun (repeated hitting keys on the keyboard to increase the proportion of the corresponding harmonic in the resulting waveform) and could be used live to subtly change the harmonic content of the waveform (I seem to remember one of the tracks on Jean-Michel Jarre's Rendezvous opening with this effect), it was very difficult to keep track of how many harmonics were present, and what the relative strength of each one was. With the graphical harmonic display of the ImpOSCar, you can see at a glance which harmonics you are using, and what their relative levels are, and altering this balance is simplicity itself with the mouse. It's also great to have the Delay and Chorus built in and tempo-sync'able. All in all, ImpOSCar really delivers on the potential of the OSCar in a way that Chris Huggett and I were never able to do in the OSC days because of lack of capital (we had several meetings with Ultravox's management about investing in OSC to enable us to get the ASG to market, but they decided to buy their parents all houses instead). Now if we can just get Gmedia to make ImpOSCar load and process samples, as Chris always planned to do with the ASG, then it really will be the instrument that I always dreamt of. Paul Wiffen Many thanks to Tim Goodyer, former editor of Studio Sound, and now the Editor of Pro Audio Asia, for the loan of his much-loved and beautifully preserved OSCar throughout this review.
In Use On the simplest level, using ImpOSCar couldn't be easier. Select a preset from the drop-down list in the 'Selected Patch' window, and play. If you want another sound, select another patch, and away you go. Likewise, editing is straightforward and intuitive, albeit with one frustrating twist. I didn't discover until the end of the review that I could use the Apple key and click on a knob to set it to a sensible default, such as 0 percent, 50 percent or 100 percent (as
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Gmedia ImpOSCar
appropriate). It would be nice if this was mentioned in the manual. Furthermore, despite the clarity of the control panel, it initially seemed that some of the knobs are rather coarsely quantised, although they are not. To obtain fine adjustment, you have to click on a knob and then move the cursor a fair distance away from it to obtain smaller angular changes. Using this method, values that jump by, say, 0.15 at a knob's rim change by as little as 0.01, as they should.
Performance features.
Having twiddled a few knobs, it took me a while to realise that there is no need to 'Save' changes because edits are retained in the current patch (you can return to the unedited patch at the click of a button, so all is well). However, this leads to another minor annoyance... you can only edit a patch and save the results elsewhere through a long-winded method of storing it in a temporary buffer, resetting the current patch to avoid losing it, choosing a new patch, and then recalling the edited version, entering it into the new location from the buffer. What's more, saving complete banks of sounds is achieved using the host application, but for me there was no 'single patch' load, and therefore no way to combine patches from disparate files into a single bank of presets. Apparently, many users will obtain this facility via their host software, but others will not be so lucky. Nonetheless, I'm more than happy with the sounds produced by ImpOSCar, and I was soon using the conventional waveforms to develop patches ranging from the expected (and still superb) bass and lead sounds, to rich imitations of solo strings, solo brass and traditional bleepy and squelchy synth sounds. I then sent myself wandering within the OSCar's digital realm, finding it easy to create patches that stepped way beyond the norm. These were bright, often percussive, and sometimes harsh sounds including bells, chimes, and all manner of strange effects. If ever a synthesizer had two distinct characters, it was this one. Of course, what many people will want to know is whether it sounds like the real thing. The short answer is, "it can be astonishingly similar, particularly in the bass registers... in fact, it's sometimes all-but indistinguishable from the OSCar". But I'm too old and ugly to leave things at that, so now I'm going to be a bit more clinical and point out that there are small differences when comparing each of the OSCar's sections against those of ImpOSCar. This is not just a question of the envelope responses and shapes being slightly different (though they are) or of the filter and resonance characteristics being slightly different (though they are) or even of the overdrive and volume characteristics being slightly different (though they are, too). It's simply that, if you create a patch on the real thing, and then try to recreate it on ImpOSCar, it's not always the same. Try setting up a basic sound on both: one oscillator, fully open low-pass filter, organ envelopes, and no modulation. You'll be hard pressed to differentiate one from the other. But
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Gmedia ImpOSCar
do something as simple as close the filter and use the filter envelope to open it, and you may hear a difference. On a superficial level, you can attribute some of this to differences in the visual calibration of the controls. For example, if you set all the knobs that control the filter envelope to 12 o'clock, you'll obtain a much faster sweep on ImpOSCar than on the OSCar, with a higher cutoff frequency in the sustain segment. But even with experimentation, you may not be able to program the sounds to be identical. Are these differences significant? I don't think so, but you might disagree.
The Effects window.
So let's move on to the aspects of ImpOSCar that cannot be compared to the OSCar, simply because they do not exist on the original synth. Primarily, this means the Effects section and the polyphony. Set ImpOSCar to four-, eight-, 12-, or 16-note polyphony, and you'll find that it's a master of rich strings, punchy brass sections, and evolving pads that shriek 'Prophet' or 'Oberheim' at you. I was impressed by the depth of the sound, especially once the pulse and PWM waves were fully available to me. I was also amazed at how organic it could sound, and even used it to create an imitation of one of my favourite Polymoog organ sounds (Genesis, circa 1978). Adding a touch of chorus and delay, I was even creating lush imitations of 1970s string synths. In short, the combination of the OSCar's quasi-analogue voicing with 16-voice polyphony and effects is gorgeous. The polyphony and effects also bring many digital patches to life, and I can imagine fans of the PPG or even some DX owners being happy to experiment with ImpOSCar to obtain percussive sounds as well as ethereal, digital pads. Finally, I found that invoking one ImpOSCar with the effects and user waves or more parameters of velocity visible. sensitivity adds yet another dimension, and although one shouldn't be too keen to sling lashings of effects, polyphony, and velocity response at every patch, this is good stuff. If I have a reservation, it's this: ImpOSCar perhaps tries too hard to be an OSCar. Think of it as an excellent polyphonic softsynth inspired by the OSCar, and I'm certain that you won't be disappointed.
Conclusions file:///H|/SOS%2004-05/Gmedia%20ImpOSCar.htm (14 of 15)9/22/2005 6:07:56 PM
Gmedia ImpOSCar
For a long time, I was not a fan of software synths, finding them less intuitive and less musical than their hardware brethren. However, my view is changing, and much of the credit — or blame — lies at Gmedia's door. I like Mtron, and the Oddity was the first software synth I used that combined an accurate representation of its inspiration, a good user interface and watertight operation. With ImpOSCar, the company has gone still further and, now that everything works correctly, the additional benefits outweigh any reservations I might have had. So, would I buy ImpOSCar? On the one hand, there's a certain kudos in owning a real OSCar, although I couldn't recommend that anyone purchase one at current prices. On the other hand, ImpOSCar is very affordable and, while you lose immediacy, it's going to be more reliable, it's more flexible, and you gain hugely in areas such as additional LFO waveforms, extra filter modes, improved overdrive/volume controls, keyboard tracking, and velocity sensitivity... to say nothing of MIDI control, 16-voice polyphony, and effects. The final point to make is that I've used OSCars off and on for two decades, and yet if I hadn't had one here for direct comparison, I wouldn't have been able to say for sure that the sound of ImpOSCar was any different from that of its inspiration. 'Nuff said, I think. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Korg Triton Extreme
In this article:
Description Extreme Spec In Brief Synthesis/Architecture Extreme Sounds Sequencer & Sequencing Effective Treatment Sampling Burning Desires The Arpeggiator Weighty Issues Conclusion Extreme Pricing
Korg Triton Extreme pros
Korg Triton Extreme Synth Workstation Published in SOS May 2004 Print article : Close window
Reviews : Keyboard workstation
Korg have redefined the workstation synth many times, and each time, it's become harder to see how they can better their previous achievements. The Triton Extreme is a bold new colour, but does the rest of it measure up?
A lot of sound for the money. USB connectivity for Derek Johnson computer interfacing and the addition of affordable external media. It has to be a coincidence: Korg's new Digital I/O (optical S/PDIF) Extreme flavour of Triton workstation must as standard. have been under development before Yamaha Some sample RAM released their enhanced Motif ES (reviewed in provided as standard.
cons Not silver any more! Sample RAM only expandable to 96MB.
summary Shame about the non-silver look, but I'll get over that with all those extra waveforms, and the warm feelings induced by the favourable price.
information See the 'Extreme Pricing' box, left. Korg UK Brochure Line +44 (0)1908 857150. +44 (0)1908 857199. Click here to email www.korg.co.uk www.korg.co.jp
Photos: Mark Ewing
January of this year). Yet here we are with what feels like Korg's answer to that challenge: it adds massive amounts of waveform ROM and USB connectivity to the high polyphony count and integrated sampling, sequencing and synthesis functions that the Triton family has come to represent. So close is the competition in this market that it seems that anyone entering it has to produce a silver- or chrome-liveried instrument: Motif ES certainly is, and even The basic 61-note synth-action Roland's imminent Fantom X will adopt this Triton Extreme (SOS used both the 61- and 88-note keyboards for this colour scheme. But prepare for a shock. The review). latest Triton has a deep metallic-blue finish that reminds you of synths from the days before the release of Korg's Trinity — the original silver machine from 1995. Without dwelling on aesthetic issues, it's hard to see why Korg would go for this colour change and still keep the Triton name, but then I suppose the new case is not so expensive to produce, and of course the new Triton once again stands out from the crowd.
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Korg Triton Extreme
Whatever the reasoning, and whilst I personally miss the sleek, now classic, silverness, you shouldn't judge a synth by its colour. The physical layout of the instrument is largely the same as what has gone before, and sonically and operationally Extreme is very much one of the Triton family. At least the most remarkable piece of front-panel real estate, present on Korg's flagships since the Trinity, remains largely unchanged. After using a touchscreen, so much other music-technology hardware feels less immediate, and it's still centre-stage here.
Description In some ways, Triton Extreme is to Triton Studio what Triton LE was to the original Triton: it repackages the essence of the Studio at a significantly lower price point. However, while it loses a handful of Studio features, Extreme has been enhanced in a number of ways. For example, gone is Studio's SCSI connector, but in its place are two USB sockets, joined by a Compact Flash/ Microdrive memory card slot. More significantly, perhaps, Extreme has a lot more samples in ROM — 160MB to Studio's 48MB. In fact, the new instrument's overall value for money is enhanced since the extra waveforms are largely sourced from previously optional PCM cards: the piano, orchestral, Trance Attack and Vintage Archive cards are represented in their entirety, whilst Dance Extreme, Studio Essentials and Pianos/Classic Keyboards are present in 'best of' form. At £155 a card (£275 for the double orchestral set), that's a significant example of added value. In addition, Triton Extreme includes a load of new sampled material, much sourced from Korg's recent top-of-the-range 'arranger' keyboard, the PA1X Pro. Physically, the Extreme's layout is virtually identical to that of the Studio, with the touchscreen in the centre, and the various patch-selection, real-time control and sequencer/sampler operation buttons scattered around the area. Korg's Valve Force tube circuit (first seen on the latest Electribe phrasecomposition devices) provides a surprise that's more subtle than the Adding sample RAM (in the form of standard casing's colour change, but it's just as SIMMs) is easily done via the slots on the visually striking when it's switched on. top panel (the slot cover is situated just It's housed behind a prominent plastic under the valve blister). blister erupting from the front panel, and its blue LED backlighting ensures that you'll know it's in-circuit even if you can't hear the result. The valve itself hardly glows: it seems Korg have developed some electronics that extract all the device's valviness without driving it very hard. Good for device longevity, I'm sure, but no fun if you'd like to see the blighter working! And working it is: the file:///H|/SOS%2004-05/Korg%20Triton%20Extreme.htm (2 of 12)9/22/2005 6:08:01 PM
Korg Triton Extreme
blister gets warm. Though only one tube is specified (each Electribe featured a pair), it seems that a single valve can be coaxed into processing stereo audio. Valve Force is available for master or insert processing, and can be applied to audio entering the Extreme via its audio inputs. The remaining feature-set is largely that of the Triton Studio (see SOS July 2002, or www.soundonsound.com/sos/jul02/articles/korgtriton.asp). So you get 120note polyphony (at most — and even this is only available under certain circumstances, depending on how waveform ROM is being addressed), dual programmable arpeggiators, a fully specified 16-track sequencer, a comprehensive sampler (offering handy integration with the sequencer) and so on. Extreme is also available in three versions, with 61- or 76-note synth-action keyboards, or an excellent 88-note piano-action affair. There are slight differences on the back panel, since there's no SCSI socket and no blanking panels for expansion options; the only extras that can be installed in the Extreme are a MOSS synth-modelling board and extra RAM for the sampler. Instead, what you do get around the back is six audio outs, two audio ins (with mic/line level switch and gain control), S/PDIF I/O on optical connectors, MIDI In, Out and Thru, and three foot-controller sockets. There is also a Compact Flash/Microdrive card slot, and two USB connectors. One of the latter allows the user to interface the synth with a computer, so that Extreme can talk MIDI with your favourite software. This process is automatic in Mac OS X, and drivers are supplied for Windows XP. Other versions of Windows and Mac OS 9 don't appear to work with MIDI, but as long as the host computers have USB, they'll 'see' any optional Compact Flash card in the Triton Extreme's rear-panel slot and mount it on the desktop, allowing you to use the card to swap samples and MIDI files. The Extreme's other USB connector allows Extreme to play host to its own external devices — hard drives or CD burners, for example (see the box on page 182 for more on this).
Extreme Spec In Brief Maximum polyphony: 120 voices. Multitimbrality: 16 parts. Waveform ROM: 160MB of 16-bit, 48kHz PCM samples, arranged as 962 multisamples and 1175 drum samples. Patches: 1536 Program memories, 1536 Combination memories, 144 drum kits, 256 GM2 Programs, nine GM2 drum kits. Sample-format compatibility: AIFF, WAV, Akai S1000/S3000 and Korg-format sample data. Sample RAM: 16MB, expandable to 96MB. Sequencer: 16 tracks, offering 1/192-note resolution. 200 songs, 20 cue lists, 200,000 notes maximum.
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Korg Triton Extreme
USB: one Type A connector (for external USB devices), and one Type B (for computer interfacing). Option: EXB-MOSS synth-modelling board.
Synthesis/Architecture Summarising Triton Extreme's 'HI' (Hyper Integrated) synth architecture is straightforward; it's exactly that of previous Tritons, and has lots in common with Trinities and earlier Korg synths. If you'd been cryogenically preserved just after buying and mastering a Korg M1 in 1988, and received a Triton Extreme as a thawing-out present in 2004, you'd still stand a good chance of being able to figure the new instrument out. At the most basic level of the synthesis structure, there's 160MB of waveform ROM, organised as 962 multisamples and 1175 drum samples. Next up in the hierarchy is the Program, which can be assigned one or two multisamples that become that Program's 'oscillators'. The output from the oscillators then passes through a signal path akin to that in a analogue subtractive synth. Twooscillator Programs use up double the polyphony of Programs with just one oscillator. The two oscillators can be velocity switched, and it's even possible to assign two velocity-switched multisamples to a single oscillator slot with no impact on polyphony (as they won't ever be playing simultaneously). Coarse and fine tuning options are also available, along with delay and reverse parameters for each oscillator. The signal path offers a pair of expressive and potentially aggressive multi-mode filters, plus filter, amplitude and pitch envelopes, two MIDI-sync'able LFOs for each oscillator, and a ton of modulation options, allowing you to add movement within a sound or to apply further sonic tweaks from the front-panel knobs or an external hardware control surface. In short, there appear to be no real changes to the synthesis system compared to that of previous Tritons. But since Korg have been getting it right, in many musicians's opinions, I'd guess they saw little need (beyond the addition of all that extra waveform material) for tweaking. The sonic open-endedness of the Triton is ensured by its sampling capabilities, which allow you to create your own multisamples from scratch, or import them from Akai-compatible CD ROMs (more on the built-in sampler later in this review). And though Extreme lacks most of the expandability of previous Tritons, file:///H|/SOS%2004-05/Korg%20Triton%20Extreme.htm (4 of 12)9/22/2005 6:08:01 PM
Korg Triton Extreme
an optional MOSS physical modelling board adds some of the functionality of a Z1 modelling synth to the Extreme. Drum Programs are different: rather than selecting drum samples from within a Program itself, you choose a whole kit of pre-mapped samples, to which a set of synth parameters is applied. There are 144 editable kit locations on board Extreme (plus nine preset GM 2 kits), 50 of which are factory-set. The kits can be edited via a sub-page of the synth's 'Global' mode. Here, one or two samples, with velocity switching, can be assigned to every key, and a very basic set of synth parameters — based around a filter and simple envelope generator — can be applied. A simple mixer lets you set up levels, pans and effect sends for each 'drum' in the kit. Both sorts of Program have access to insert and master effects, plus an arpeggiator. I'll return to the arpeggiator later, and there's more on the effects in the box on the left. Next up is the classic Korg Combination, first seen on the M1. In a Combi, up to eight Programs can be layered, with options to velocity and/or key-split individual elements. Each part may also be addressed on its own MIDI channel, like a miniature multitimbral setup complete with basic mixing options. Extreme's array of insert and master effects start to come into their own in this mode, with the user free to treat individual elements with dedicated effects. The arpeggiator also magically grows a twin: arpeggiators A and B can have their own pattern and note-resolution settings, making one-finger performances a definite option, since many of the presets are drum or riff-based patterns, a feature exploited in the factory Combis. Yamaha's Motif ES has just one arpeggiator, and it's just not the same once you've had two running at once! Note that the insert effects included in a Program are not automatically replicated in a Combi. You can manually recreate the appropriate settings, but it's easier to copy them, which you can do from anywhere on the Triton, including Programs, other Combis and Songs. The number of Programs and Combis that can be stored on the Triton Extreme appears to be the same as on the Studio; however, the new instrument comes with a lot more preloaded material. There are still 1536 each of Programs and Combis, but Korg fill up 1344 of the former and 1280 of the latter with factory settings; that's up from 512 each on the Studio. In addition, there's one bank of 128 Programs that becomes available with the MOSS board installed, and there's a non-overwriteable General MIDI 2 bank, complete with variation patches, to a total of 256.
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Korg Triton Extreme
Extreme Sounds In general, the sound of Triton Extreme is rather like a fully-packed example of an earlier Triton — all the basic waveforms, Program and Combis will be the same. That said, it's nice to have them all in one place, without having to shell out for the extra PCM cards, or fuss with installing them. It's also good to have some of the excellent sounds and waveforms developed for the PA1X Pro arranger keyboard. Not that I have a soft spot for arranger keyboards, you understand, but this is a particularly well-voiced example. Demo a Triton Extreme, and you'll be impressed right from the start, whether you just scroll through Programs and Combis sequentially or take a more measured approach by auditioning presets within the 16 themed Categories (the Categories differ slightly for Programs and Combinations). The synth fires up with some piano-based Programs and Combis in the first slots. These have a really nice, upfront sound, and play especially well with the top-of-the-range 88-note weighted-action Extreme. Only the lower octave seems a little too artificial, as if there aren't enough multisamples to stretch right down to the bottom of the range. For once, I enjoyed some of the string simulations, though mainly the massed section examples (for example D018: '14 Violins ST', and D033: 'Largo Celli'); the solo and string quartets were not so good. The synth strings and string-like pads were effective, especially in layered examples, but it's hard to point you at any particular examples as stand-outs, since there are so many! As I mention towards the end of this article when discussing the arpeggiator, I found some of the one-finger Combis to be really impressive, quite contrary to my usual feelings in these matters! Most of the World-themed Combis had something to offer, though inevitably a little cheese becomes evident if you check out every one (particularly in a couple of Combis with the word 'Irish' in the name). If I have one overall comment it's that, in common with factory sound collections everywhere, there is a reliance on effects to create impressiveness. Nevertheless, the Triton's basic waveforms can sound excellent without the window dressing.
Sequencer & Sequencing As mentioned above, Combis can be used as mini-multitimbral setups, but for real multitimbral use, you need Sequencer mode. Here, you're presented with MIDI recording facilities that rival those of computer-based MIDI software. From the variety of potential time signatures, to individual track looping, and the comprehensive quantise and cut-and-paste editing, you may find yourself writing more with the workstation than your computer! Just remember to save your work somewhere, as all Songs are lost when you power-down. Briefly, an Extreme Song is made up of a maximum of 16 tracks, each assigned its own Program and accessing a wide range of mix controls and track offsets such as transposition, but also including voice parameter offsets, thus allowing file:///H|/SOS%2004-05/Korg%20Triton%20Extreme.htm (6 of 12)9/22/2005 6:08:01 PM
Korg Triton Extreme
you to fine-tune a Program in situ without having to re-edit it and re-save it. Each track also has its own MIDI channel, and of course Extreme can be played multitimbrally from the sequencer of your choice. The mixing options include access to global and insert effects. You can record your data in step or real time, or a mixture of both. Post-record editing can happen on a global level (there are plenty of ways to move, correct and change data globally), or right down at the single-event level. As on the recent tube-equipped Korg Electribes, the impressive glow actually comes from an LED (here, a blue one) rather than the built-in valve! Nevertheless, Valve Force widens your sonic palette nicely.
The sequencer has quite a few nifty features when you start to investigate it more closely. For example, both arpeggiators can be used, and their output can be recorded into a sequencer track. And for those of you who might bemoan the way in which a Combi can't be used in a song, you'll be pleased to hear that the settings of any Combi (or Program, for that matter) can be copied into a sequence, via a feature called Auto Song Setup. This is a great option when you happen to be noodling with an inspirational sound, and would like to start sequencing with it right away. You won't lose that inspiration by having to set up a Sequence and replicate all the effect settings manually. You're a couple of button pushes and a metronome click away from starting to record your performance. The settings from another Song can also be copied, and if you'd like a regular starting point, you can use Templates. A good selection of factory-produced choices are provided, with memory locations for your own creations. Further automatic song creation assistance comes in the from of RPPR — Real-time Pattern Play/Record — which lets you trigger user-defineable or factory phrases with single key presses. Another aid to speedy song construction is the Cue List. With this facility, you can write verses, choruses and bridges and so on as individual 'Songs', organise how you'd like them to play back in a Cue List, and then convert the result, when you're happy, to a new Song. Couldn't be simpler. RPPR and Cue List could offer some interesting results as live tools, but for straight-ahead set-list creation, there's Song Play Mode. Though aimed pretty squarely at the playback of General-MIDI-compatible Standard MIDI Files from card or disk, your SMF could easily be an Extreme Song, full of program changes aimed directly at the main sound engine...
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Korg Triton Extreme
Effective Treatment The 24-bit effects system remains much the same as that of Triton Studio (and previous Tritons). The major addition is the Valve Force circuit — a first for a workstation such as this — which functions as an insert or a master effect. Generally speaking, the effects system offers five insert effects 'slots' which can be assigned flexibly within Combis and Sequencer setups, plus two global sendtype effects and a master three-band EQ. There is a central pool of 102 effect and processor types; any of these can be assigned to an insert effect, while 89 can be used as a master effect. The usual collection of reverbs and delays is included (all tempo-sync'able, of course), plus modulation, pitch-shifting, and amp-simulation algorithms. Ring modulation, doppler, and sundry straight and dynamic distortions add to what's on offer. Though insert effects can be chained, there are already-chained varieties that can be used in one slot. Then there are the special 'double-size' effects which use up two slots, such as the vocoder and the more complex delay options. The sends to the main master effects are accessed, in confusing fashion, from Menu 8, entitled 'Insert Effect'. This seems to be a bit of leftover OS from somewhere, and should really be cleaned up at some point. This is a good collection of processes, for the most part well-implemented, and capable of adding the final touches to your mix. What's more, in the same way that synth parameters can be controlled in real time, so can those of the effects, for moving beyond sweetening to purely creative uses. And then there's that Valve Force circuit, which is something of an interesting concept: using a real valve on the output of a digital instrument. Valve Force lends a warm, miked-up feel to anything you put through it, but note that the effect can't be used on the actual digital out of the Extreme.
Sampling The Triton Extreme, like previous full-strength Tritons, offers all the facilities of a very well-specified stand-alone studio rack sampler, recording in 16-bit, 48kHz quality. That fixed rate is no problem, since resampling to lower rates is one of Extreme's many DSP tools (including normalisation and reverse). Custom multisamples of real instruments can be created, or imported from Akai-format sample CDs. The synth's own outputs can be resampled, of course, and beats and loops can be time-sliced and time-stretched. Sampling can be to RAM or to optional Compact Flash card. On the subject of RAM, I'm pleased to note that 16MB is provided to get you started, enough for nearly three minutes of mono sampling. The sample RAM complement can be upgraded to 96MB (via the access panel shown on page 177). By way of comparison, Yamaha's Motif ES comes with no RAM at all as standard, but it can be expanded to a more impressive maximum of 512MB. One of the most interesting aspects of Triton sampling is the way in which the sequencer can be integrated, using what's called 'In Track Sampling'. This is file:///H|/SOS%2004-05/Korg%20Triton%20Extreme.htm (8 of 12)9/22/2005 6:08:01 PM
Korg Triton Extreme
almost like having a dedicated digital recorder on board — almost. You record audio into a track, and trigger data is created to play the audio along with the sequence on playback. With careful planning, it's even possible to add vocals to a sequence. There are limits to the length of audio that can be recorded, but that audio can be processed by insert effects on the way in, and fully edited afterwards. Keeping track of the resulting samples is a little tricky, but the saving/loading routines Sampling and keygrouping remains as easy as ever via the touchscreen. and file system are reasonably logical in their own way. And if you invest in a large hard drive (or one of the larger-capacity Compact Flash cards), you'll have no trouble keeping your work safe and backed up. You will have to buy something, though, as samples disappear on power-down. The rear panel's S/PDIF input can be used to record into the sampler, but it is fixed at a sampling rate of 48kHz, with no sample-rate conversion available, so you'll have to supply your own sample-rate converter if your digital devices can't be set to output at this rate. The USB computer connection comes in really handy for sample editing, provided, of course, that you've added a Compact Flash card to the Extreme. If so, you can then move samples seamlessly back and forth from synth to computer and vice versa, allowing you to record or edit with whichever device seems most appropriate to you. Korg call the Extreme's sampler an 'Integrated' device, and it really is: sampling can be entered from any mode, so you could sample yourself playing a riff on a Program or Combi, or just resample a particularly interesting sound effect or rhythmic effect produced by the arpeggiator. The output of the sequencer can also be resampled to RAM or Compact Flash card. Keep your mind open to the possibilities, and you'll be able to stretch the effects complement even further, by adding effects to the stuff you sample, freeing up the onboard processors for other work.
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Korg Triton Extreme
Burning Desires As I've said elsewhere in this review, the Triton Extreme allows you to add USBcompatible storage devices to your setup. As there's no CD burner built in, you might well wonder whether you can hook up a USB CD burner to create reference CDs of Extreme mixes. The short answer is yes, although the process isn't exactly straightforward, and requires additional expense. Not only do you require a CD burner, but also a large card for the Compact Flash/Microdrive slot. Microdrives are available only in 1GB sizes, and Compact Flash cards come in a variety of sizes up to 1GB, so finding a card shouldn't be a problem. Thankfully, prices aren't too high, and they make a great backup medium for general piece of mind, not to mention for use at gigs. Assuming you are prepared to get a burner and a suitable card or Microdrive, the CD-burning process is fairly straightforward. You use the sampler's resample option to record a complete Extreme Song to the media installed in the card slot, and then go to the 'Make CD' page in Extreme's Media Mode. You have the option of creating a list that'll be burned all in one go, or you can add mixes to the CD piecemeal. The CD then needs to be finalised before it can be played on a normal CD player. I wasn't able to try this for myself, but certain guarded comments in the manual lead me to believe that the process may not be all that fast, with 1x (ie. real-time burning) the best that's on offer. Still, it's a nice option to have, and does mean your tracks can be started and finished wholly within your Extreme. Note that although Triton works at 48kHz, and has no sample-rate converter on its digital input, audio burned to CD is converted to 44.1kHz first.
The Arpeggiator These days, arpeggiators go far beyond merely breaking up held chords, and most provide autoaccompaniment-like patterns, giving their host instruments an appeal that's as wide as possible. Korg do allow you to define your own patterns, of course, which at least means you can create automatic music that's based on your own ideas. If you create the right sort of pattern yourself, and mix note resolutions, you could even generate algorithmic-type musical output.
At first glance, not much seems to have happened around the back since the days of the Triton Studio. But look more closely and you realise the Studio's SCSI expansion slots are no more, nor can the mLAN board or the ADAT out/word-clock in option can be fitted to the Triton Extreme. By way of compensation, there are dual USB sockets on the left of the Extreme's rear panel for the connection of peripherals, or for computer interfacing, and there's a Compact Flash/ Microdrive slot on the right.
The Triton Extreme's dual polyphonic arpeggiators are provided with five preset patterns and 507 user patterns, 489 of which are filled at the factory. Basic
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Korg Triton Extreme
chord arpeggiations are available, over up to four octaves with a variety of note resolutions, but the phrases range widely from guitar strums and drum patterns to bass lines. It's a simple matter to set up key ranges for arpeggiations, and if you combine this flexibility with tempo-sync'ed delays and LFO modulations, the potential for instant customised grooviness is quite high. Try out some Combis for a good taster — for example, A001, 'Film Sound Track' doesn't even have its arpeggiators enabled as standard, yet the result is excellent when they are. Also check out Combi A007, 'EXP Film Strings', for an example with the arpeggiator explicitly engaged. Many of the Combinations in the 'World' category of factory settings are also pretty stunning. Be inspired, and come up with your own variations.
Weighty Issues As I mentioned in the main body of this article, it was my good fortune to have access to the 88-note weighted piano-action version of the Extreme for much of this The 88-note weighted version of the Triton review. Generally speaking, it's an Extreme, as used for much of this review. excellent keyboard action, offering a good halfway house that provides something like the feel of a genuine acoustic piano without being too hard to play if you spend the rest of your time with plastic-action synths. As with all hammer-action keyboards, I felt a little odd at first (especially playing straight synth sounds and drum kits), but this soon passed. I wouldn't be too unhappy to have the Extreme 88 at the heart of my studio! The action itself is Korg's RH2 (Real Weighted Hammer Action 2) keyboard. It's apparently designed to simulate a grand piano's touch, with light resistance as you play higher up the keyboard. It certainly feels like a player's keyboard, and would be just the thing to grace the stand of a monied lounge performer in search of feel, portability and a good piano sound-set, but who might need to occasionally branch out with backing tracks and a more varied sound.
Conclusion Extreme offers the best of everything Triton has become in the few years since its introduction. I can't say it leaps ahead of the rest of the range, but it still provides, on balance, more facilities for a lot less money than the previous Studio variety. The internal hard drive and optional internal CD-burner of the Studio might be missed by some, but adding your own external USB-equipped devices will hardly make a dent in the savings offered here. Sure, they won't be internal any more, but I'm not complaining. This leaves me wondering what's to become of the rest of the Triton range.
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Korg Triton Extreme
Officially, Triton Studio is still a current product, but I don't know who'd buy a Studio model when Extreme is so much cheaper, and when you consider how much you get in Extreme compared to what you lose — just think of all that extra waveform ROM. I'd say Extreme sounds even more fabulous as a result. I know which option I'd go for!
Extreme Pricing As I finish this review, there are 10 possible Tritons to choose from: 61-, 76- and 88-note versions each of LE, Studio and Extreme, plus the Triton Rack. Just for the record, here's what's available, with current pricing: Triton LE 61 — £899; Triton LE 76 — £1029; Triton LE 88 — £1499; Triton Studio 61 — £2149; Triton Studio 76 — £2549; Triton Studio 88 — £2999; Triton Extreme 61 — £1749; Triton Extreme 76 — £1899; Triton Extreme 88 — £2499; Triton Rack — £1099.
Some attrition in this range must surely be forthcoming soon, although it's hard to say what's on the way out. My guess would be that many potential Studio customers will be tempted by Extreme when the see the price difference — a 76note Extreme costs less than the Studio 61! Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Kurzweil K2661
In this article:
Kurzweil K2661
Goodbye Floppy To Sample Or Not To Sample Synth Workstation Previous Kurzweil K-series Published in SOS May 2004 Reviews In SOS Print article : Close window Ribbons & Blows Reviews : Keyboard workstation Ain't ADAT A Shame Effectively The Same K2661 Specification Conclusion
Kurzweil K2661 £1999 pros The power of a K2600 with the portability of a K2000. Superb sound quality. VAST technology still the best sample-and-synthesis implementation out there. Compatibility with older library material, as well as other formats. KB3 tonewheel emulation. Kurzweil's proven commitment to providing upgrades/updates. Familiarity for previous K2series users.
cons Looks too much like old technology with a new hat on. KDFX now looks limited. Some features (such as audio input to the synth engine) require the sampling option. 48-voice polyphony is tight with full arrangements and layering. Small display. The price is high relative to the competition, on a spec that looks distinctly oldfashioned in some key areas.
summary Looking at my pros/cons list is eerily familiar. I would have
Kurzweil's high-quality K-series workstations have always commanded a premium price tag, but it's been a while since the range's underlying technology has been updated. Is their latest offering still competitive in 2004? Paul Ward
When Kurzweil's K2000 first took the Photos: Mark Ewing stage back in 1992, I felt that it represented a considerable leap forward for synthesizer workstation technology. Admittedly, no single aspect of the machine's feature list was startlingly new at the time, but Kurzweil had succeeded in bringing together a number of mouth-watering goodies into a very well-integrated package. Not only that, but the quality of the hardware, software and sample set were exemplary. Kurzweil had developed a proprietary synthesis system based on DSP building blocks which they termed 'VAST' (Variable Architecture Synthesis Technology). From the first, VAST proved incredibly powerful and flexible, with a modulation architecture that was the answer to many a synthesist's prayers. As the '90s moved on, Kurzweil continued to develop and expand the K2-series with major software updates (including a highly developed filing system and fully featured sequencer), expansion options and more advanced versions of the hardware, culminating in the K2600, released in 2000, with its choice of 76 semi-weighted or 88-note piano-weighted keyboard, faders, ribbon controllers and wide variety of controller inputs. There's no doubt that the K2600, in both its 88-note and 76-note form, is
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Kurzweil K2661
expected so much more of a new Kurzweil workstation. The lighter package and nonweighted keyboard are welcome, but the disappearance of the ribbon controllers is certainly not. The K2600, good though it is, failed to deliver any huge developments in VAST technology, with the possible exception of KB3 (although even this was an earlier optional K2500 development that was incorporated as standard on the K2600) and I'm afraid that's even more the case with the K2661, three to four years on. If you want K2600 abilities in a familiar, portable K2000-like package then this machine may be the one you've been waiting for, but don't expect anything earth-shatteringly new.
formidably large and heavy. I well recall carrying one to my upstairs studio to conduct the SOS review — phew! While this kind of bulk may not be a problem for professional musicians with a road crew to hand, to the amateur and semiprofessional with live performance in mind, it may well be a strong reason for crossing the 2600 off their short list. The earlier K2000, on the other hand, with its five-octave (61-note) non-weighted keyboard, was of manageable proportions and weight, making it ideal for gigging, rehearsals and generally being carried around in the back of a car. In my review of the K2600 (see SOS April 2000, or www.soundonsound.com/sos/apr00/articles/kurzweilk2600.htm), I voiced my doubts that we would see a non-weighted version (given the lack of a similar nonweighted version of the earlier K2500). It's taken four years, but Kurzweil have finally proved me wrong! So is this a case of 'better late than never'? Since the K2661 is essentially the soul of a K2600, this review will give a broad outline of its features, coupled with comparisons with the earlier K2600. For a more detailed review of the capabilities of the K2600 itself, take a look at that April 2000 review.
Goodbye Floppy
information K2661, £1999; SMP61 sampling option, £500; ROM3 Stereo piano ROM option, £129; ROM4 Electric piano ROM option, £200; 2661 ribbon controller, £53. All prices include VAT. Team Kurzweil Europe +45 7556 9644. +45 7556 9655. Click here to email
Gone is the floppy disk drive of the earlier K2-series machines, and in comes a Smart Media memory card slot. This slot appears on the back of the unit, which seems odd to my way of thinking, but the port is internally illuminated by a beautifully eerie blue light (this seems wasted around the back too...) which makes finding it very easy! Using a Smart Media card is as easy as using any other storage device on these machines: simply go into Disk Mode and dial the device up. Given the relatively low cost and availability of these cards, I'd say this is a good move by Kurzweil. A SCSI port is still available for connection of other storage devices (the K2600 had two SCSI ports) and it happily worked with the chain of Jaz, Zip and CD-ROM drives I usually have connected to my faithful old www.kurzweileurope.com K2000. The 2661's operating system software resides in Flash ROM and is www.kurzweil updateable from Smart Media, over MIDI, or from a SCSI device.
musicsystems.com
Still with us is the 64 x 240 fluorescent backlit display. This has not changed since the K2000 and is badly in need of an update. I can't help feeling that the power of VAST would be now better served with a screen similar to Korg's Triton — VAST is still the cream of the crop, in my opinion, and there is so much capability in here that is locked behind pokey editing screens and cryptic abbreviations. I'm convinced that giving VAST a friendlier face would win it a lot more converts.
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The assignable sliders are really handy, not to say invaluable when acting as drawbars under the 2661's built-in KB3 tonewheel
Kurzweil K2661
organ emulation (note the footage legends 32MB of sample ROM is provided under each 'drawbar'). onboard, including as standard all of the K2600's samples, Kurzweil's previously optional ROM1 and ROM2 Orchestral and Contemporary blocks and a General MIDI sample set. A further pair of 8MB expansion slots will take the total up to 48MB of ROM at maximum. The optional ROM blocks Kurzweil currently have to offer (numbered 3 and 4) are the Stereo Dynamic Piano and the recently announced Vintage Electric Pianos. It would have been nice if these were also included.
The K2661 will accept a maximum 128MB of sample RAM, although at the time of the review, Kurzweil were supplying the K2661 with no sample RAM preinstalled. However, apparently the plan is to ship the K2661 with the full complement of 128MB pre-installed from April onwards for no extra cost. I'm pleased to hear this, although I have to say that 128MB still isn't much these days (even my old K2000 has 128MB of RAM in it). However, a full complement of 1.5MB of P-RAM is also included as standard, and this is arguably even more important than sample RAM. What's more, in the past it has always been inexplicably expensive as an add-on. P-RAM (Parameter RAM) holds usercreated objects and those loaded from disk, including Keymaps, Programs, Setups and Sequences.
To Sample Or Not To Sample Sampling is available as a cost option. It's certainly not as important an upgrade as it once was, since there are plenty of ways of getting sampled material into the K2661 without it, and sampling is arguably better done on a computer where it can be edited on a large screen before porting the samples to the synth. The sampling upgrade brings with it both analogue and digital sample inputs and permits external access to the entire VAST DSP engine. However, the sampling option is not required to edit and make use of RAMbased samples within VAST. The K2661 inherits the formidable sample-editing capabilities of the K2-series and it is fair to say would still put a fair number of dedicated samplers — hardware and software — to shame. VAST itself is more than capable of the real-time enhancing or mangling of sample material beyond all recognition. One significant new feature that Kurzweil have given to VAST is 'Triple Modular Processing'. What this essentially means is that you can 'chain' three VAST Layers together in series, passing the results of Layer 1 via an audio insert into Layer 2, and the result of Layer 2 similarly into Layer 3. This hugely increases the number of possible DSP block combinations (to something around 30 billion, according to the manual — and who am I to argue?). Each Layer can still have its own audio input, so you don't lose any voices in a Program, and there's no extra cost in polyphony either. Kurzweil have added 94 new DSP Algorithms to take advantage of Triple Mode. True to form (for Kurzweil have always been file:///H|/SOS%2004-05/Kurzweil%20K2661.htm (3 of 9)9/22/2005 6:08:06 PM
Kurzweil K2661
exceptionally conscientious at keeping users of older products well-furnished with updates), this new feature is also available to K2600 owners via a software update — check the Kurzweil web site for details. Editing the K2661 follows a pattern that will be familiar to existing K2-series owners. I did have a problem with the alpha-wheel, since I kept reaching for it only to find my fingers jabbing the cursor buttons. I put this down to a combination of the screen having been pushed to the right (in comparison with earlier Kurzweil synths) to accommodate the faders, and the off-putting circle drawn around the cursor buttons! Kurzweil's KB3 tonewheel organ and Leslie emulation mode makes a welcome return to my studio on the K2661. I was mightily impressed by this technology when I first saw it on the K2600, and it has lost none of its charm here. OK, so it is primarily aimed at getting Hammond sounds, but it can be pressed into other uses, and if tonewheel organs are your thing you're going to love this facility. The range of control on offer is simply mind-bending and the results can be very convincing. The faders act as drawbars, and there are enough assignable switches to set up toggles for Leslie rotor speed, percussion, vibrato and such like. Due to the processing required to achieve it, KB3 can still only be used on a single MIDI channel.
Previous Kurzweil K-series Reviews In SOS K2000: March 1992. K2000 sampling option: August 1993. K2000 v3: February 1995 (see www.soundonsound.com/sos/1995_articles/feb95/ kurzweilk2000.html). K2500R: October 1995 (see www.soundonsound.com/sos/1995_articles/oct95/ kurzweilk2500r.html). K2500/2500X: April 1996 (see www.soundonsound.com/sos/1996_articles/apr96/ kurzweilk2500.html). K2000VP: September 1998 (see www.soundonsound.com/sos/sep98/articles/ kurzweil_k2000.html). KDFX effects board: January 1999 (see www.soundonsound.com/sos/jan99/articles/ kurzweilkdfx.159.htm). DMTi digital multitrack interface: May 1999 (see www.soundonsound.com/sos/may99/ articles/kurzweildmti.htm). K2600X: April 2000 (see www.soundonsound.com/sos/apr00/articles/kurzweilk2600. htm).
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Kurzweil K2661
Ribbons & Blows The K2-series sequencer is still the best hardware sequencer I have ever used. There, I've said it. But... given that I spend most of my time using a computer as my sequencer of choice, I've never had to spend that much time using it. Certainly it is highly capable, with many of the features you'd expect from a dedicated software package. Songs can be tied into Setup objects, allowing them to be triggered by key-presses and enabling grooves to be edited on the fly, even adding fills and introducing new parts as you go. For live use these features are fantastic and certainly easier to run with than a computer-based system. The use of a ribbon controller enhances the experience greatly — yet sadly, this is no longer built into the K2661, as two were on the K2600, but is an add-on option. I was unable to try out the ribbon input, having no suitable device available. It seems to me a real shame that Kurzweil have not included the K2600's ribbons on board — especially the short one. These were one aspect of the older K2600 that seemed so perfectly implemented. Cruelly, the manual seems to enjoy explaining the neat integration of the sequencer with the ribbon as if you already have the ribbon option installed... Though the loss of the ribbons is a shame, it's fair to say that you'll never be short of hardware control options with a K2661. In addition to the mod/ pitch wheels, the 10 front-panel switches and the eight faders, there's provision for connecting up to four switch pedals, two continuous control pedals, a ribbon controller and a breath controller. I'm glad to see a breath controller socket; they are still a very much underexploited control source in my opinion. It worked fine with my Yamaha BC1.
Whatever the cost/features ratio looks like, the rear panel of the K2661 leaves you in no doubt that this is a professional workstation. The Smart Media card slot is a welcome modern addition, but there's still ADAT interfacing and even SCSI built in as standard. MIDI In, Out and Thru are present, and the Thru socket may even be switched to act as a second MIDI Out port. On the right can be seen the blank holes where the high-impedance jack, XLR and optical S/ PDIF/AES inputs to the sampler are found if the sampling option (not included on the review K2661) is installed.
All of the K2661's control sources transmit MIDI. You can define the MIDI controller number of each of them on an individual Setup basis — in fact you can define many aspects of their functionality, including the range of transmitted values and initial position when a new Setup is recalled. This is fantastic for live use, and also for creating control templates for software synths and effects. Four individual analogue outputs are provided (as opposed to the eight of the K2600) with 24-bit D-A converters, and there is also a pair of analogue mix outputs. The analogue outputs are all balanced, but can also be used unbalanced (Kurzweil have apparently patented the output circuitry used in these output stages). Unfortunately, in changing the analogue outputs, they have also lost what I rated as one of the K2000's best features. The K2000 outputs were actually configured as send/return loops and it was possible to tap other file:///H|/SOS%2004-05/Kurzweil%20K2661.htm (5 of 9)9/22/2005 6:08:06 PM
Kurzweil K2661
instruments into the return. I used this facility to take a pair of keyboards on stage and mix them together to a single stereo output pair. I also use them in my studio to save precious mixer channels. Rats!
Ain't ADAT A Shame For digital interfacing, Kurzweil have given the K2661 a pair of optical interface ports as standard. These are software-configurable as either two-channel S/PDIF or eight-channel ADAT ports at 16, 20 or 24-bit precision. This is welcome in helping to integrate the workstation into a wider studio system without compromising on either the number of available output channels or sound quality. The K2661 must act as the master clock to make use of the input port, and its sample rate is fixed at 48kHz. This is an unfortunate limitation for a system otherwise running at 44.1kHz — as is mine! More rats! It seems a shame there's no electrical (co-axial) S/PDIF port, too. Kurzweil have created 575 programs for the K2661 utilising the capabilities of the expanded sample ROM. I admit that I'm very impressed by many of them. I'm well-used to the capabilities of VAST, but the programmers have The rest of the generous range of really excelled themselves with some connections are visible in this shot. As well as display controls, there's provision for the of this material. The Program material using the included ROM-block samples connection of two continuous control pedals, no less than four footswitches, and dedicated has also been re-edited and enhanced breath and ribbon-controller connectors. by Kurzweil from that provided when There's all this, a headphone jack, and three the blocks were available as add-on assignable sets of stereo analogue outputs, options. During this review I didn't have too! the ability to compare the original Program set with these new ones (my own K2000 doesn't have any of the optional ROM blocks installed), but I can say that the Programs and Setups continue Kurzweil's honourable tradition of being stunningly good. Whatever the K2661's faults, you're extremely unlikely to be left stuck for a decent sound of any genre. There is a wealth of sounds for all styles of music, and I can imagine anyone looking for film-score inspiration will have no problem here. I found the Roland TR808 samples particularly good, especially the kick drum, which usually suffers when sampled — whether this is down to the quality of the source samples or the D-A converters I'm not certain, but whatever it is, it works. Make no mistake; in terms of the quality and breadth of their sounds, I believe that Kurzweil are still at the top of the pile, even after all these years. The sounds just seem to leap out of the speakers and caress your ears. If you get a chance to audition a K2661, try out some of the Groove Setups and imagine weaving your own material into them — a guaranteed mouth-watering experience! General MIDI finally sees itself a permanent, selectable, feature, complete with its own sample set. Previous K2-series workstations did have the ability to work
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Kurzweil K2661
under GM, but required the loading of a specific set of Objects to achieve the trick. As with all of the K2661's sample set, the quality is superb. Confining VAST to the production of GM material would be madness, admittedly, but at least the option is there when needed — and many of us do need it from time to time. Sample library compatibility has always been an important aspect of the K2series. The K2661 follows the norm with its ability to load library material from the Roland 700-series, Akai S1000/S1100/S3000, and Ensoniq EPS/ASR. Of course, WAV and AIFF formats are also supported, and I had no trouble loading in older K2000, K2500 and K2600 material. Kurzweil supply compatibility banks on the supplied CD-ROM, which ensures that older material will sound as close as possible to the originals (albeit with better-quality effects) on this new machine.
Effectively The Same Effects are supplied by Kurzweil's KDFX multi-channel effects processor. This is integrated into the output busses as four insert effects and a single master global effect. I had hoped that the processing power of KDFX would have moved with the times, but we are still stuck with a maximum of four PAUs (Processor Allocation Units) to share between the insert effects — and this is woefully inadequate, particularly as the number of effects using only one PAU are very much in the minority. Come on, Kurzweil! The power of processors has increased dramatically since the KDFX first saw the light of day and these kinds of restrictions seem to hail from a different age. On the plus side, the KDFX effects cover a wide range of useful ground and sound as good, if not better, than those on any workstation I've come across. A hard-copy of the familiar Getting Started manual is supplied with the keyboard, but all other documentation (and the sound library) is on the accompanying CDROM in Adobe Acrobat format. I can accept this as being a sensible option. Searching K2-series manuals has always been daunting — not due to inadequacies in the documentation, but because Kurzweil have a habit of providing extremely thorough coverage, making searching for answers a timeconsuming process. This way, a simple search in Adobe Acrobat Reader helps you get to the item you need in double-quick time — and any errors or omissions can simply be remedied by posting a new manual on the web site. Inevitably, I haven't covered all of the features of the K2661 here, but as I mentioned near the start of this review, you can find many comments relating to the features I haven't covered here in my previous reviews of the K2-series synths.
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Kurzweil K2661
K2661 Specification 61-note non-weighted action keyboard. 64 x 240 backlit LCD. 1.5MB P-RAM. Up to 128MB of sample RAM (128MB is fitted as standard from April 2004). 32MB of base ROM, expandable to 48MB. 48-note polyphonic, 16-part multitimbral. Four analogue outputs, plus a main stereo pair. Stereo headphone output. Eight-channel 24-bit ADAT I/O or two-channel S/PDIF I/O (optical only). Flash OS for software updates via Smart Media, MIDI and SCSI. VAST synth engine. KB3 tonewheel organ emulation. 32-track sequencer. SCSI port. Built-in KDFX effects processing.
Conclusion And there lies the rub. Because I can point you in the direction of those reviews, safe in the knowledge that the K2661 offers very little that I didn't cover in my K2600 review four years ago. The biggest problem I have with the K2661 is simply that it feels like a machine that should have been introduced back in 1998. At that time it would have amazed and impressed, much as the K2000 did six years earlier — but by the standards of 2004, it seems like we're simply going over old ground. Yes, the sample ROM has increased, connectivity has been improved and we've lost the floppy drive in place of a Smart Media card slot, but I can't help thinking 'so what?' Nearly all of the K2661s features were introduced with the K2600 four years ago and even at that time, I said that Kurzweil needed to do a lot more than include previous options as standard and add a few digits to the model number. But that's essentially what has happened here. Heck, I'm even still getting the aliasing noise on the DSP waveforms — this should have been cured years ago. That said, personally I'd love to own a K2661. Why? Because it has all the features of a K2600 (with the sad exclusion of those gorgeous ribbon controllers), but has the footprint and non-weighted keyboard feel of my K2000 (my personal file:///H|/SOS%2004-05/Kurzweil%20K2661.htm (8 of 9)9/22/2005 6:08:06 PM
Kurzweil K2661
preference). So would I stump up that £1999 asking price? No. Sorry Kurzweil. I'm a great champion of the K2-series synths, as anyone who knows me will confirm, but I don't think they've done nearly enough this time around to justify that kind of price tag. A look at what features are available on comparable competing workstations at the same price, or for even less, backs me up here (just check out pages 176 to 186 this month for a prime example). Had the K2661 appeared 12 months after the K2600, I could have been considerably more forgiving, but I don't think I'm completely unjustified in feeling that what I should be reviewing in 2004 is the K2700, or even the K3000, with an appropriately more up-to-date feature-set. Add a bigger screen, up the polyphony, screw in another MIDI port and the story could have been so different. Hang on... that's exactly what I said four years ago! There, told you... Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Latest Sample CDs
In this article:
Amazing Thailand **** LA Riot V ***** WEP ***** White Grand ****
Star Big In Japan
Latest Sample CDs Sample Shop Published in SOS May 2004 Print article : Close window
Reviews : Sound/Song Library
***** Shampoo **** Sigue Sigue Sputnik *** Flock Of Seagulls
Amazing Thailand ****
** The Vapours
MULTI-FORMAT
* Big In Japan
Based in Japan, Discovery Sound have a wide range of sample CDs. Their Ethnic series comprises Amazing Thailand, Bom Dia Brazil, Techno Lama, Legend Of China, Discovery Japan, Shamisen, Digeridoo, Voice Percussion and Okinawa Tradition. The subject of this review comprises an audio CD and an Acid/WAV/REX 2-format CD-ROM. The audio CD kicks of with a demo track, closely followed by a generous selection of short percussion loops, typically just a couple of bars long. This section includes both drums and gongs, as well as some tuned percussion phrases. From there it's on to stringed instruments and wind instruments, before moving on to male and female voices. Interestingly, the stringed instruments contain multisamples and, as most feature a fairly fast decay, it's easy enough to load them into a sampler, as you don't have to worry about looping. Of course there are also string phrases and some suitably raspy sustained bowed-string multisamples, some of which would need looping. Some of these phrases are wonderfully plaintive, while others are more strident, and all capture the genre beautifully. The same recipe carries through to the wind instruments, which include flutes and some reeded pipes. The female vocal phrases are lovely, though I haven't a clue what they're singing about, and there are also some equally inscrutable spoken phrases. The collection finishes with ambient backgrounds from places like markets and restaurants, more often than not with traffic or air-conditioning noise in the background. That's the end of the sound content, but, for those interested in learning more about Thai music, there are 24 short movie clips of the various instruments being played. Overall the recording quality represented here is impressive, though I would have much preferred the multisampled instruments to be available in a broadly file:///H|/SOS%2004-05/Latest%20Sample%20CDs.htm (1 of 6)9/22/2005 6:08:09 PM
Latest Sample CDs
supported sampler format, such as Akai, Gigastudio, Halion, or EXS24. The samples are wide-ranging and well recorded, and the REX 2 option is also particularly welcome, as it enables loops to be used over a range of tempos, so that you can fit the sampled phrases in with existing arrangements. Paul White Audio CD and Acidised WAV/REX 2 CD-ROM set, $91 (around £50). Click here to email www.discoverysound.com
LA Riot V ***** AUDIO The bulk of this new Chronic Music hip-hop collection is made up of nearly 70 construction kits, which do the usual job of giving a burst of the full groove followed by its component loops. Tempos range from 68bpm to 136bpm and the kits are well put together, with lots of layers and interesting one-off touches. All the basics are present and correct, with the drum loops well programmed and the bass good and deep. The melodic elements are a nice mixture of real and synthetic instruments, with some nifty guitar work, smooth keys, and synth washes. Percussion samples and loops are also well utilised, with conga, bongos, and even tabla cropping up. The welcome inclusion of some subtle DJ skills, so often missing in other hip-hop collections, really injects some added interest. In terms of style, the kits cover a huge amount of ground, from smooth laid-back R&B grooves, to more upfront urban styles. The influence of the Neptunes is heavily in evidence, as is the choppy, electronic funk of influential producers like Dr Dre, Timbaland, and others. Of course, with so much material, not everything is going suit any one producer's specific needs, but the sheer number of samples means anyone with an interest in hip-hop/urban sounds would be able to find something useful. The first three CDs also include a selection of pretty good (if a touch noisy in places) scratching, and over 300 guitar riffs and loops. The key/tempo-grouped guitar licks are good, making use of some nice tremolo and funky wah-wah effects to create tasty riffs to cut and paste. CD four bucks the construction-kit trend and instead focuses on a huge assortment of single loops and hits, starting with 350 drum loops grouped by tempo. The live drum kit is varied and the beats sound fresh, played with the right dose of hip-hop sensibility and groove. The programmed loops are equally well put together, with the right amount of booming kick and crisp snare. I also liked the wide use of electronic percussion. Completing this rhythm section is a good selection of bass guitar lines (sorted by file:///H|/SOS%2004-05/Latest%20Sample%20CDs.htm (2 of 6)9/22/2005 6:08:09 PM
Latest Sample CDs
key and tempo). These are fat sounding, as you'd expect, with some nice filter/ wah-wah effects, especially on the slap lines. Although a good chunk of the samples are too busy for my taste, for raw material ready to be chopped up in a sampler they are just the job. After some well-played keyboard riffs, which are a bit too polite (with the exception of the excellent clavinet loops), the remaining chunk of CD four is taken up with vocal samples and a comprehensive collection of single drum hits. The vocals are split up into sections of talkbox phrases (let down by some sloppy editing), female sung hooks, and some male rapped shouts and ad libs. The female vocals are very good, with urban-styled hooks and phrases that really wouldn't be out of place in any of the pop R&B/hip-hop tracks in the charts. The rapped section also works well, with the two vocalists sounding the part and managing to avoid most of the usual hip-hop clichés. Once again, the editing of the samples could have been better. LA Riot V contains a huge amount of source material for anyone wanting to put together any flavour of hip-hop or urban music. Although the kits lean more towards the newer, pop-based hip-hop styles, the sheer amount of material here, especially on CD four, makes LA Riot V a very handy collection of usable, authentic samples. Oli Bell Audio 4-CD set, £59.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com
WEP ***** MULTI-FORMAT Scarbee's Fender Rhodes 73 Stage Piano (reviewed in SOS March 2003) was a great success, and it's a pleasure to report that the company have now turned their attention to another classic vintage keyboard, the Wurlitzer 200A electric piano. Cheaper (and therefore more widespread) than the Fender Rhodes, this instrument was first manufactured in the early 1950s, popularised by Ray Charles and The Beatles, and later featured by Supertramp — but don't hold that against it! As with the Rhodes project, Thomas Skarbye went to the ends of the earth to get the sound of the good old Wurli just right. Prior to recording, the Danish musician instigated an extensive restoration process, which, from the hapless instrument's point of view, must have felt like being experimented on by aliens. Virtually every reed was replaced, the power transformer ripped out to eliminate mains hum (always a problem with Wurlitzers), preamp transistors updated, broken knobs fixed and dampers adjusted. Finally, the piano was tuned by the crude but timehonoured method of adding solder to a reed to flatten its pitch, or filing it off to file:///H|/SOS%2004-05/Latest%20Sample%20CDs.htm (3 of 6)9/22/2005 6:08:09 PM
Latest Sample CDs
make the note sharper! Considering its relatively unsophisticated design, the Wurlitzer is an expressive instrument with a surprisingly wide tonal range. To reproduce this, the producer took samples at 16 different dynamic levels, matched by a corresponding set of 16-dynamic release samples. The piano has 64 notes from A1 to C7 (middle 'C' being C4) — the Wurli's top five notes are undamped and didn't require release samples — so the total sample count comes to 1968. The Gigastudio version (about 1.2GB in size) sounds and feels great. At times you'd swear you were playing the real thing — all its dynamic nuances have been captured, from a subdued, delicate harp-like tinkle to a biting, reedy twang. Anyone who has played a real electric piano will appreciate the utterly realistic and subtle note-off effect created by the release samples. My partner and I compared WEP to her beloved Wurlitzer 200, and found that the Scarbee version sounds practically identical, even down to the characteristic tremolo effect (now assigned to the mod wheel). The real instrument seemed to have slightly more bite, but that's probably a by-product of its rather trebly built-in speakers — play the Scarbee version loud through a stage combo amp, and it would take your head off. In one important respect, the samples are better than the real thing: there's no bloody noise! Why all this trouble over an old electric piano? As Mr Skarbye points out, it's soon going to be impractical to service this fine instrument, let alone take it on the road. By performing such a painstaking sampling job, Scarbee have effectively kept in circulation one of the great sounds of pop music. Its trademark mains hum may now be consigned to history, but thanks to this library all the desirable qualities of the Wurlitzer electric piano live on to fight another day. Dave Stewart 16-bit Gigastudio multi-CD-ROM set, £125; 24-bit Halion, EXS24, or Kontakt multi-CDROM set or DVD-ROM, £125. Prices include VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com
White Grand **** MULTI-FORMAT Talk about pushing the envelope — in an attempt to provide the most dynamically expressive sampled piano ever, the Sampletekk White Grand incorporates 36 different samples per note (16 dynamic layers for pedal up, 16 file:///H|/SOS%2004-05/Latest%20Sample%20CDs.htm (4 of 6)9/22/2005 6:08:09 PM
Latest Sample CDs
for pedal down, and four dynamics of release samples). With over seven octaves (A0 to C8) to cover, that's a scary amount of samples, but the producers made life a little easier for themselves by sampling only the white notes. (Maybe that's why it's called the White Grand?) According to my calculations, a 'grand' total of 1836 samples were performed on a Malmsjo nine-foot grand piano and recorded at 24-bit resolution. It isn't possible for a musician to consistently perform 16 distinct dynamics on fifty-odd piano notes, so my guess is that the player generated a large number of dynamically varied samples which were later analysed and graded accordingly. (The alternative would be to haul in a mechanical arm programmed to strike the keys with a predetermined degree of force, but the slightest miscalibration could see the piano reduced to matchwood.) The 16-bit Gigastudio version (which will only run on Gigastudio 160) requires 3.5GB of disk space, while the Halion and Kontakt versions (available in a choice of 16- or 24-bit resolution) consume 4.8GB. The piano is cleanly recorded with a natural stereo image, and its samples are clear, noise-free, and precise, with plenty of presence. The dynamic range is exceptionally wide — for the first time, we can hear the soft, closed timbre of very quiet piano notes gradually opening up in response to increments of velocity. As the volume grows, the timbre becomes naturally brighter, until one arrives at the brilliant, ringing sound of fortissimo notes. To capture a piano's timbral transition from soft and warm through to hard and steely with no apparent 'jumps' is quite an achievement. Pressing the sustain pedal engages a complete new set of 16dynamic 'pedal down' samples, their glorious sustain benefiting from the additional sympathetic overtones of other strings. The piano's bottom register sounds big, strong, and stately, its samples dying away naturally with no looping. The tuning complies to the industry standard of 'A' at 440Hz, ensuring that this instrument will layer successfully with other keyboards. The White Grand is technically beyond reproach, and musically, its bold, assertive sound will work fine in a pop/rock setting. However, it's hard to imagine jazz and classical pianists warming to its slightly middly tone and rather percussive attack, both perhaps the result of miking too close. But piano sound is an intensely personal thing, and while it's hard to be absolutely objective, what is beyond doubt is that Sampletekk have reproduced the tonal nuances of this Malmsjo grand to an unprecedented and impressive degree. Dave Stewart Gigastudio, Halion, or Kontakt multi-CD-ROM set or DVD-ROM, $195 (around £106). Click here to email www.sampletekk.com Published in SOS May 2004
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Latest Sample CDs
Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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M Audio Firewire Audiophile
In this article:
Overview Drivers And Utilities In Use System Requirements Conclusion
M Audio Firewire Audiophile Firewire Audio & MIDI Interface Published in SOS May 2004 Print article : Close window
Reviews : Computer Recording System
M Audio Firewire Audiophile £229 pros Flexible routing options. Assignable Level Controller. Low-latency drivers.
cons Currently has problems with sleep in OS X. All connections unbalanced.
summary The M Audio Firewire Audiophile uses the power of Firewire to offer more inputs, outputs and routing options than similarly priced USB alternatives, though it's currently held back under OS X by driver issues.
information £229 including VAT. M Audio +44 (0)1442 416590. +44 (0)1442 246832. Click here to email www.maudio.co.uk www.m-audio.com
Test Spec Firewire Audiophile driver build 1016 (Windows) and 1068 (Mac OS X). PC with 2.4GHz Pentium 4 CPU, Belkin three-port PCI Firewire adaptor and 512MB RAM, running Windows XP Home Edition (SP1).
Until recently, the benefits of the high-bandwidth IEEE 1394 interface were available only to musicians on a relatively generous budget, but now M Audio are making Firewire peripherals available at new low prices. Mike Watkinson
Firewire audio interfaces have, until recently, been specified and priced for the pro and semi-pro market, with names such as MOTU and Metric Halo at the forefront. M Audio, with their 410 (reviewed in Sound On Sound March 2004: subscribers can read the review on-line at www.soundonsound.com/sos/mar04/articles/maudio410.htm), have expanded the market downward and continue this trend with the introduction of the Firewire Audiophile, taking what is now a family name from their ever-popular PCI interface and its USB sibling. With an SRP of £229 including VAT, and already just under £200 'on the street', this interface breaks a psychological price barrier and competes with some of the higher-end USB interfaces such as the Tascam US122 and Emagic EMI 6|2 (and 2|6). Where it scores, of course, is in using the high-bandwidth Firewire protocol, which has been proved capable of supporting far more channels than this interface offers. The sense of 'will it work or won't it' engendered by the restrictions of USB, a connection originally intended for printers and scanners, and the resultant negative attitude towards it as a serious protocol for audio are not stigmas from which the Firewire Audiophile will suffer.
Overview
Apple iBook with 600MHz G3 CPU and 640MB RAM,
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M Audio Firewire Audiophile
running Mac OS 10.3.2. Apple G5 dual 2GHz with 1.5GB RAM, running Mac OS 10.3.2. Tested with Steinberg Cubase SX v1.0.6 (on PC and iBook), SX v2.0.1.10 (on G5), Emagic Logic Platinum v6.4.1 (on iBook), v6.3.3 (on G5).
In the box along with the Audiophile, M Audio supply two Firewire cables, one sixpin to six-pin and one four-pin to six-pin. Along with the power-supply unit and a CD containing drivers and manuals, they also include a software bundle comprising Ableton Live Delta, Arkaos VJ Lite, Propellerhead Reason Adapted, Dsound RT Player Express, IK Sampletank Free and Linplug's Free Alpha soft synth. The unit itself has an elegant case intended for the desktop, measuring 184 x 140 x 42mm. It follows the design cues of the larger 410 and the USB Audiophile (albeit with more muted colours) with brushed-aluminium finish and silvercoloured control knobs. The front panel has a reassuring simplicity, with only two buttons and two rotary controls along with a headphone jack and a sprinkling of subtle LEDs. Round the back there are two unbalanced analogue inputs using RCA sockets at -10dBV line level, four analogue unbalanced outputs with identical spec, co-axial RCA connectors for S/PDIF input and output, MIDI In and Out sockets, a power-supply connector and two sixpin Firewire 1394 sockets — the second is useful for connecting further Firewire devices if your host computer The Firewire Audiophile's driver software under Mac OS X. The unit's Assignable has only one port. The S/PDIF output Level Controller can be switched to control should support pass-through of AC3 and DTS-encoded multi-channel digital four different signal levels (right). audio for connection to a suitable decoder, although at the time of writing there was an issue with DTS awaiting a driver revision. There are no mic preamps or associated phantom power options (as there are with the 410) but the software-assignable 'Level Controller' remains. By default it controls the monitor level of the analogue outputs, but it can also be assigned to any group of mixer output faders in the Audiophile's control panel. This selection comprises: Software return — that is, the level of the three stereo pairs running from the host software to the virtual mixer. Output levels of the three stereo pairs of outputs (two analogue pairs and one digital pair). Input levels of the analogue input pair and digital input pair. Auxiliary send buss level.
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M Audio Firewire Audiophile
Next to the headphone socket is a switch that selects the monitor source (in conjunction with the Headphones source settings in the hardware page of the control panel). There are LEDs to indicate headphone source monitor status, S/ PDIF input and output (in the presence of a valid signal) and power, where steady status indicates a valid connection to the host computer.
Drivers And Utilities Checking M Audio's web site for the latest drivers, I downloaded build number 1016 for Windows XP (the supplied CD had version 1011) and 1068 for Mac OS X (1064 on the CD). I'm undecided about this constant driver revision — of course it is reassuring to see that M Audio are committed to revising drivers that iron out bugs and conflicts, but one can't help worrying that early adopters might feel they are being used as unpaid beta testers. Driver installation was completely painless on both XP and OS X, having taken heed of the warning not to have the Audiophile connected during the process. Along with the driver, a control panel is installed, accessible from an icon in the system tray on Windows, and in the Other subsection of System Preferences on the Mac. You could drag this to the dock to create a graphic shortcut similar to XP's System Tray icon. The control panel in both cases has the same four-pane design as that which comes with the 410, reflecting the commonality of functions, but displays fewer channels on each pane, in line with the reduced number of inputs and outputs. The Mixer pane shows level faders for the six software return channels, Amongst other options, you can choose arranged in stereo pairs. These are accompanied by output select switches which output feeds the headphone socket. that, by default, route these channels to their corresponding physical outputs, and give you the opportunity to set up your own routing if you wish. The two pairs of faders controlling input levels have similar monitoring options that are unselected as a default. All the faders can be linked in stereo pairs and each pair carries two auxiliary buss send level controls, one for each side of this extra stereo buss, which allows for the creation of a separate submix if desired. All input channels have pan position controls. The Output pane has faders for the six physical outputs of the Audiophile, arranged in stereo pairs, and stereo channel faders for the auxiliary buss and headphone levels. The Hardware pane shows detected sample rate and allows for the setting of sync source, headphone source and the function of the frontpanel momentary switch. This can be set to switch between the two possible file:///H|/SOS%2004-05/M%20Audio%20Firewire%20Audiophile.htm (3 of 6)9/22/2005 6:08:14 PM
M Audio Firewire Audiophile
headphone source settings (A or B), or to switch between Direct Monitoring Off or On. On this pane the XP version differs from OS X, in that samples per buffer (latency) is selected on this pane of the control panel. In OS X this is chosen in the driver configuration window of the host application. Common to all panes are the Save and Load switches for mixer settings and the buttons for choosing the function of the Assignable Level Controller. Just below this is a graphic indicator of the number of Audiophiles connected, showing which one is being actively affected by the Control Panel. It should be noted that under Mac OS X multiple Audiophiles were not supported at the time of writing, but imminent change was expected with a future driver revision.
In Use With the provision of unbalanced inputs and outputs only, the Audiophile is always going to be susceptible to the rogue hums and buzzes that plague such equipment when connected to other devices in the studio. This can be avoided to some extent by the use of pseudo-balanced cables (see Frequently Asked Questions in Sound On Sound January 2002, amongst other articles). As far as audio quality is concerned I found the unit to be on a par with that of the Tascam US122 I have been using recently for mobile recording and playback, although this latter device does benefit from the advantage of balanced inputs. Tests with Rightmark's Audio Analyser revealed a frequency response of +0.12, -0.87 dB for 20Hz to 20kHz at a sample rate of 44.1kHz. This improved to +0.04, -0.10 dB at 96kHz. Dynamic range improved from 95.6dBA for 16-bit operation to 101.3dBA for 24-bit operation, agreeing with M Audio's quoted figure for input 1/2 — the headline figure of 108dBA quoted on M Audio's web site is for the outputs only. Total harmonic distortion and noise for combined input and output was measured at 0.004 percent.
System Requirements PC Pentium III 500MHz or higher. 128MB RAM. Windows 2000 SP3 or Windows XP SP1. Direct X 8.1. Six-pin Firewire port or adaptor.
Mac (OS 9) G3 500MHz or higher. 128MB RAM. Mac OS 9.2.2.
Mac (OS X) G4 500MHz or higher. 256MB RAM. Mac OS 10.2.6 or greater.
The Audiophile can be buss-powered, although on many laptops the Firewire buss is not able to supply enough power. These are recognisable by their fourpin socket (often called iLink), where a fully powered buss sports the larger sixpin variety. Confusingly, PCMCIA Firewire cards often carry six-pin sockets but file:///H|/SOS%2004-05/M%20Audio%20Firewire%20Audiophile.htm (4 of 6)9/22/2005 6:08:14 PM
M Audio Firewire Audiophile
despite this they also lack sufficient power. In both these cases it is necessary to connect the supplied mains adaptor. The test iBook seemed perfectly at ease powering both the Audiophile and a Lacie Pocket Drive daisy-chained from its single Firewire port (M Audio recommend the Audiophile to be last in the chain) although battery life was considerably foreshortened in this situation. It already tires too quickly for my liking, although as a side note, upgrading to OS X 10.3.3 just before this review went to press seemed to have quite a beneficial effect on battery life, a more thorough test of which will hopefully appear in a future article. On both platforms the audio drivers worked flawlessly down to the smallest buffer size of 64 samples, recording at the maximum resolution of 24-bit and 96kHz sample rate, giving an impressively (almost) negligible 1.5ms of latency. However, where everything worked like a charm in the XP environment, functionality in OS X on the iBook was hindered by a few annoying niggles. Initially, while audio worked without a problem, MIDI communication was notable by its absence; after several launches and quits, attempting to load Cubase or Logic with the Audiophile connected caused the computer to restart. Initially I suspected a MIDI driver conflict, having recently run into problems on another machine connected to a Digi 002 Rack and Midiman 8x8, where the simultaneous presence of both MIDI drivers in the folder System/Library/Audio/ MIDI Drivers stopped the 002 from being available to hosts other than Pro Tools. Simply moving the 002 MIDI driver outside the Drivers folder had allowed Logic and Cubase to use the 002 as an audio interface (without MIDI, of course). An obscure forum posting concerning driver conflicts between Digidesign and Tascam had led me to the above 'solution', and the test machine for this review had Tascam drivers installed, so I tried removing both of the other MIDI drivers in the iBook's driver folder (Emagic's MIDI driver as installed by Logic, and the one for the Tascam US122) and retested. Initially I thought I'd cracked it, as removing both these drivers and leaving only the Audiophile MIDI driver allowed the MIDI In and Out to work on the device. Just to make sure, I then put them back expecting the problem to return, which it didn't. So much for my hunch (as Jean de Florette once famously remarked).
Like many USB interfaces, the Firewire Audiophile is equipped only with unbalanced audio I/O.
My suspicions then turned to Disk Permissions. When odd things happen that don't conform to a pattern it often indicates a problem 'under the hood' of OS X that can be cured by repairing Disk Permissions on the System disk in Disk Utility. Realising I hadn't repaired disk permissions for quite some time, I went to Disk Utility and pressed the magic button. Since when I have not had a problem with the Audiophile's MIDI I/O. However, the story did not end there. Whenever my iBook went to sleep a valid connection to the Audiophile was naturally lost, but it was not re-established on file:///H|/SOS%2004-05/M%20Audio%20Firewire%20Audiophile.htm (5 of 6)9/22/2005 6:08:14 PM
M Audio Firewire Audiophile
waking up. The only way of doing so was to restart the machine (turning the Audiophile on and off was not enough, nor was disconnecting and reconnecting the Firewire cable). I encountered the same difficulty on a dual G5, and checking M Audio's web site revealed that this is a known problem with OS X, and also suggests that the latest driver revision goes some way to curing it. What I did discover was that this is only a problem when an application (such as Logic or Cubase) is open. With no application open, connecting to the Audiophile is reestablished on waking up. How often do you close an application before putting a computer to sleep? Still, it's good to know that M Audio are aware of the problem and, even as I write, are beavering away to provide a solution. Windows XP and 2000 users are unaffected by this anomaly.
Conclusion The market positioning of the Audiophile is intriguing. It offers four-in, six-out operation (albeit that two each of those are digital) and MIDI, along with highly competitive audio quality for the price, flexible software control and the bandwidth headroom and general cachet of Firewire, for the price of a higher-end USB interface. Firewire still has a premium attached to its name, and the extra overhead it affords allows this device to gain an appreciable advantage over its USB stablemate, in terms of simultaneous channels of operation and the ability to carry an independent submix. With unbalanced connections and no preamps or phantom power, its role as the hub of a laptop-based audio recording workstation has a few limitations. M Audio maintain that the Audiophile was created with live performers and DJs in mind, and this device would make an excellent choice for laptop users requiring multiple outputs and some MIDI control in a live situation, with the added flexibility of surround-encoded digital output. And this is just a suggestion, but how about a version that has two pairs of analogue inputs, one pair of analogue outputs and a mix fader (along with software emulation of the RIAA EQ curve) — that really would be a laptop DJ solution! Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Mackie HR626
In this article:
Design Niceties Round The Back Performance
Mackie HR626 £1598
Mackie HR626 Active Studio Monitors Published in SOS May 2004 Print article : Close window
Reviews : Monitors
pros Clear sound with good bass extension. Useful frequency tailoring to suit the environment and the user's taste. Can be used horizontally or vertically. Magnetically shielded.
Mackie continue to expand their range of studio monitors by combining an innovative D'Appolito design with their existing passive radiator technology.
cons Slightly forward high end may not suit those who demand absolute neutrality.
summary These monitors are sensibly priced in the UK and they do the job. They have clarity, they have adequate bass extension, and they tell you what you need to know about a mix. You can buy more refined-sounding speakers if you are prepared to spend a lot more, but in most project studio situations I don't think the mixes would be any the better for it.
information £1598 per pair including VAT. Mackie UK +44 (0)1268 571212. +44 (0)1268 570809. Click here to email www.mackie.com
Paul White
Mackie's HR626 follows on from the Photos: Mark Ewing HR824 and HR624 to further expand the company's range of active monitors. It employs the same design principles as these earlier models in most respects, but also utilises the so-called D'Appolito driver layout, named after Dr Joseph D'Appolito, who championed this particular geometry. As I understand it, if the speakers are arranged so that a single tweeter sits directly between two identical bass/midrange drivers and in line with their centres, the topology qualifies as D'Appolito. This particular design has one 6.7-inch woofer at each end of the baffle with a one-inch dome tweeter set centrally within a cast-aluminium waveguide designed to match the dispersion characteristics of the drivers at the crossover point. The monitors measure 8.8 x 19.6 x 12.4 inches and weigh 31lbs each, so they're fairly compact. Both bass/mid-range drivers are built into cast-aluminium frames, where the 1.25inch voice coil works within a ferrite magnetic structure that employs an additional field-cancelling magnet. This aims to reduce the external magnetic field that might otherwise cause picture distortion in CRT video monitors or TVs. A field-cancelling magnet is also fitted to the tweeter, which is driven by a oneinch voice coil. The use of a waveguide provides symmetrical horizontal and vertical coverage specified as 120 degrees, ±10 degrees.
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Mackie HR626
One benefit of the D'Appolito layout is that the speaker can be used either vertically or horizontally, though I'd instinctively use it vertically where possible, as this should produce the most accurate off-axis response. On the other hand, it looks seriously cool when used horizontally, and the sweet spot still seems plenty wide enough. In addition to its use as a stereo studio monitor, the HR626 is also suitable for use in surround systems, with or without subwoofers.
Design Niceties Like the other monitors in the Mackie HR series, the HR626 is a two-way biamplified design using servo feedback circuitry to minimise low-frequency distortion. The tweeter is driven by an integrated Class-AB amplifier utilising what is described as 'parametric servo feedback'. The conventional bass port is replaced by the now-familiar Mackie passive radiator, which resembles a 6 x 9inch elliptical speaker (a little smaller than the one used in the HR824), but with no voice coil. This takes the place of the usual 'slug' of air in the bass port for the purposes of cabinet tuning and comprises a mass-loaded piston with an aluminium honeycomb-composite diaphragm suspended in a rubber surround. It is mounted on the rear panel behind the vented amplifier heat sink and extends the low-frequency performance almost as far as that of the HR824. The waveguide-loaded tweeter uses a similar aluminium dome design to the other models. Although the drivers and waveguide look very stylish, the black ash cabinet is a no-nonsense design made from 5/8-inch MDF with additional internal bracing to increase its rigidity and reduce resonances. The crossover design is also optimised for the D'Appolito driver configuration, and includes time correction and phase alignment. This is an adaptation of a Butterworth-response, 18dB/octave filter with a crossover frequency of 1.8kHz. As with the other models, extensive driver protection guards against excessive signal peaks, overheating, and radiofrequency interference, while independent compressors come into play to prevent the high- or low-frequency amplifiers from being driven into clipping. As expected, the amplifier has a delayed, soft start to prevent thumps, and it also shuts down gracefully. The power switch and its status LED are mounted on the front panel along with a clip warning LED, but the rest of the controls and switches are on the rear panel.
Round The Back The rear of the cabinet houses the amplifier heat sink and the passive radiator, where the heat sink also doubles as the control and connection panel. Power comes in via an IEC mains lead, and the signal input is on balanced XLR, balanced TRS jack, or phono. Gain controls are fitted for trimming the input levels, and the familiar Acoustic Space switches are provided to optimise the response of the speaker depending on its proximity to walls and other surfaces. file:///H|/SOS%2004-05/Mackie%20HR626.htm (2 of 4)9/22/2005 6:08:18 PM
Mackie HR626
These offer three options: the 'A' position's 4dB of shelving cut at 40Hz; the 'B' position's 2dB of shelving cut at 40 Hz; and the flat 'C' position. The reason these filters are necessary is that, when a loudspeaker is mounted away from walls, very low frequencies radiate in all directions, so the available LF energy propagates over a full 360 degrees. This is known as full-space operation. If, however, you were to place the speaker against a solid wall, all the energy radiating to the rear would be reflected back into the room in phase with the forward-radiating sound, resulting in a doubling of the bass energy at the listening position. To compensate for this, the bass end needs to be dropped by 3dB, but as few home studio walls are completely solid, Mackie have given the switch a 2dB cut in the Half Space position. There's also a -4dB setting (quarter space) for use when the speaker is mounted close to two walls, such as in a corner, though this is not generally recommended, as the bass end can become very uneven. For systems where it is inconvenient to switch the monitors off at the mains, there's an automatic standby mode, selected from the rear panel, that activates the speakers when the input signal rises above -74dBu (0.155mV). A High Frequency switch provides 2dB of HF cut or lift at 10kHz, both to compensate for the acoustic environment and to suit personal preference, and there's also an inbuilt, switchable 80Hz high-pass filter (second-order Butterworth) for when you're using the HR626 with Mackie's optional subwoofer. The optional bass roll-off filters of the other two models have been omitted. The outcome of all this technology is a monitor with a frequency response of 42Hz-20kHz ±1.5dB and a -3dB point at 40Hz. The high end is 3dB down at 22kHz, where the maximum peak SPL (per pair) at one metre is 118dB and the maximum short-term SPL is 109dB.
Performance I've used HR824s in my studio for a long time now and I found the HR626s to have the same family sound. If anything they are slightly crisper in the mid-range because of the smaller bass/mid-range drivers, and in this respect their sound is probably closer to that of the HR624s, but with the bass extension approaching that of the HR824s (which actually go down to 38Hz). When you compare any of the Mackie monitors with really high-end models using soft-dome tweeters, they sound slightly less smooth and a little more 'forward', but in the context of 'real life' monitoring, I find they work extremely well, giving a good account of what's really going on when you're mixing.
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Mackie HR626
The HR626s have a very solid punchy low end which is at the same time well controlled, a clear and detailed mid-range, and plenty of high-end detail without the shrillness some metal-domed tweeters can produce. I'm not usually a fan of metal tweeters, but I think these represent a sensible compromise between neutrality, clarity and the ability to play loud without excessive distortion. The stereo imaging is fine in either orientation, and because the low-end roll-off is slightly higher than that of the HR824s, they are suitable for use in smaller rooms with less-than-optimal acoustic treatment. In my own studio, which is garage-sized (on account of it having originally been built as a garage!), they delivered plenty of kick, but without provoking the room into misbehaving. If you like the other monitors in the Mackie HR series, then you should get on well the HR626s. They are in many ways a blend of the attributes of the two previous models, and they deliver more than adequate level for any sane user at nearfield distances (just a couple of decibels less than the HR824s) while also 'telling it like it is'. Good mixes sound great, bad mixes sound awful, which is just as it should be. In all, a very worthwhile addition to an already well-respected range of monitors. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Presonus Eureka
In this article:
Facilities Overview Internal Circuitry Front-panel Layout EQ & Dynamics Verdict
Presonus Eureka Recording Channel Published in SOS May 2004 Print article : Close window
Reviews : Recording Channel
Presonus Eureka £499 pros High-quality transformercoupled Class-A preamp. Adjustable Saturation control. Flexible three-band fully parametric EQ. Highly controllable VCA compressor with side-chain filter.
This smart new unit offers EQ and compression, plus a transformer-coupled Class-A preamp and an unusual Saturation control. Hugh Robjohns
cons
Despite the rise of the virtual studio, Photos: Mike Cameron some hardware devices will always remain necessary. Microphones and loudspeakers are clearly indispensable, but so too are decent microphone preamps. The 'channel in a box' units — combining preamp, compression, summary and equalisation — are consistently The Eureka is a well-designed popular solutions and, despite the voice channel with the benefit advancing quality of plug-ins, in many cases analogue compression and EQ offer of a transformer-coupled Class-A mic preamp, a useful a sound quality or character that most plug-ins can't match. Control knob size and density makes operation fiddly. No clear centre detents on gain controls. Input metering too simplified.
Saturation feature, and effective and controllable compressor and EQ facilities. The optional high-sample-rate A-D board should enable easy integration with digital recording systems when it becomes available.
information £499 including VAT. Hand In Hand +44 (0) 1579 326155. +44 (0)1579 326157. Click here to email www.handinhand.uk.net
The American manufacturer Presonus have produced a wide variety of costeffective but always well-designed preamps and channel units. Many have been valve-based designs, but there have also been several solid-state devices. One such product was the VXP reviewed by Paul White in SOS March 2001, and the Eureka under review here bears some striking similarities to that unit (although at a significantly lower UK price). Most notable is the porting across of several key technologies, including the transformer-coupled Class-A preamp and the unusual and adjustable saturation distortion control. This bestows the Eureka with a sound character between a traditional valve-based preamp and a solid-state one.
Facilities Overview
www.presonus.com
The facilities provided by the Eureka are fairly conventional for a mono channel
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Presonus Eureka
strip. There's a switchable mic, line and DI input, a flexible solid-state compressor, and a three-band, fully-parametric EQ. There is also a balanced insert point immediately after the preamp section, and the equaliser can be switched into circuit either before or after the compressor. As already mentioned, the mic input is transformer balanced and features a discrete Class-A FET buffer followed by a 'dual servo' gain stage using industrystandard NE5534 op amps. The FET buffer circuitry incorporates the unusual Saturation facility (created by varying the FET drain current) which generates an adjustable degree of even-order harmonic distortion. This provides a variable quasi-valve warmth, yet with the benefit of solid-state precision and reliability. Another facility included in this preamp design — and one which is becoming increasingly popular — is a variable input impedance, spanning 50-2500(omega) in this case. A simple three-LED input meter is provided, showing -20dBu, 0dBu, and clipping (+22dBu). Although better than nothing, I felt that this input metering was a little too simple to be very useful. The compressor section is based around the familiar THAT 4301 VCA chip, and includes a complete set of user controls rather than the preset modes offered in the VXP. A variable high-pass filter is also included in the side-chain to enable simple frequency-dependent dynamic processing — such as crude de-essing — and the compression characteristic can be switched from the default hard knee to a soft-knee mode, enabling both dramatic and more subtle compression effects. The three-band parametric equaliser section is versatile, and avoids the dualconcentric controls that Paul found frustrating in the VXP, although the close proximity of controls still makes it fiddly to adjust for anyone with fingers thicker than a supermodel's. The band gain controls provide a slightly smaller boost and cut gain range than most designs, with just 10dB on offer, but there's usually something seriously wrong if you need more than that from a general-purpose analogue equaliser! The side effect of a restricted control range is that the precision of gain setting has been usefully enhanced. The bandwidth of all three sections can be varied from a narrow two thirds of an octave (Q value of two), to a broad two octaves (Q value of 0.4), and with the widest bandwidth settings and extreme frequency positions the top and bottom bands can be made to serve as reasonable shelf filters too. The frequency ranges of all three bands overlap well to provide seamless coverage of the entire spectrum. The electronically balanced main output is provided on both XLR and TRS sockets, and the nominal level can be controlled between +10dBu and about 80dBu. A small moving-coil VU meter in the centre of the front panel can be switched to show either the output level or the compressor's gain reduction. The insert point is fully balanced, with separate TRS sockets for send and return. An optional 24-bit A-D module can also be installed via the rear panel if required, operating at all the standard sample rates from 44.1kHz to 192kHz, and providing
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Presonus Eureka
both AES-EBU and S/PDIF outputs. Since these are both dual-channel interfaces, an auxiliary balanced TRS socket accepts an external analogue linelevel signal to access the second A-D channel — the Eureka's own output signal occupying the first digital channel. Unfortunately, this A-D converter module was not provided for the review, so I am unable to comment on its performance.
Internal Circuitry The Eureka is certainly well built. The majority of electronics are contained on a main PCB which covers most of the available floor area of the box, with mic preamp circuitry to the left, dynamics and EQ in the centre, and an integral linear power supply to the right. Three power-rail regulator chips are mounted on a metal strip running from front to back inside the unit, and this acts as the only heat sink — but the power consumption is very small so heat isn't really an issue. The main circuit board has a rectangular section cut away in the centre at the rear to accommodate the optional A-D converter module. This can be fitted by the user very easily, as it is simply bolted in place and connected to the main unit via a small multi-pin plug. Most components on the circuit board are surfacemount devices, although there are a few conventional-sized components too, including a pair of socketed NE5534 op amps in the input stage. Two small daughterboards run just below the lid to carry and the top row of front-panel switches and controls, while the toroidal mains transformer and mu-metal screened mic transformer are located at opposite sides of the box. One point worth noting is that the mains operating voltage is factory configured and not user adjustable. The IEC mains inlet incorporates a fuse holder accessible from the outside, and the mains power switch is also located here on the rear panel — which may be a blessing or a hindrance depending on your point of view!
Front-panel Layout The Eureka looks very clean and tidy, with lots of small blue knurled knobs on a brushed-steel front panel. The panel markings are small but clearly legible, and all of the continuous rotary controls have a light detented action traversing about 40 'clicks', which gives good tactile feedback. Unfortunately, though, this detented action feels rather more precise than it really is, and doesn't help in locating the unity positions of the gain controls, for example. So, to set the output level to the calibrated unity gain mark, or to cancel the gain of an EQ section, you are forced to look closely at the panel markings, instead of just feeling for a
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Presonus Eureka
centre detent. It's a small point, but one which some users might find frustrating after using the unit for a while — I know I did! The input stage is easy and logical to use, with three rotary controls along the bottom and five illuminated buttons running in a row above. Microphone input gain is continuously adjustable from +10dB to +54dB, but this can be supplemented by an additional 10dB of Here you can see the printed circuit board make-up gain in the compressor and a which runs the entire width of the rack further 10dB at the output level control, casing. The toroidal mains transformer is at the right-hand side, and the cutout for the giving a maximum overall gain of optional rear-panel A-D converter board is at 74dB. The microphone preamp is the top. specified with an EIN value of -127dB (weighting not given) and distortion is quoted as below 0.005 percent with no saturation (rising to 0.5 percent with full saturation). The buttons provide phantom power, a 20dB pad, an 80Hz high-pass filter, and polarity reversal. One of the other two rotary controls adjusts the Saturation from zero to 100 percent, and increasing the Saturation certainly adds a warmer character to the sound, making it sound thicker and richer. However, this is pretty subtle most of the time, and it's not quite of the same character as the distortion you get in a typical valve preamp. Then again, it isn't as unpredictable as valve preamps, and won't wear out either! I'm sure many users will find it a very useful facility, but don't audition the Eureka with the expectation of it sounding like a traditional valve preamp. The third rotary control is actually a switch to change the input impedance in five steps from 50(omega) to 2500(omega). Many preamps offering variable input impedance provide a higher maximum figure than this — typically 5k(omega) or so — but the range offered here is sufficient to match a wide selection of mics, including most ribbons, and affords useful tone-shaping characteristics which are far more subtle than most EQs can manage. The fifth and final button selects the fixed-level line input, which bypasses the mic input stage altogether and so is unaffected by the Gain, Saturation, and Impedance controls. The front-panel unbalanced instrument input presents a 1M (omega) input impedance and is selected automatically when a plug is inserted. The DI input is processed through the microphone gain stage (with a lower overall gain spanning 0-44dB), and is therefore able to take advantage of the Saturation effect, which is a nice facility. In fact, the DI input sounds particularly good, with a full rich character — I find many preamp DI inputs tend to sound rather too clean, but not the Eureka. I'd still recommend recording guitars and basses through a decent simulation system like the Pod XT, but if you want a simple clean input then this is a good one.
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Presonus Eureka
EQ & Dynamics The front panel places the dynamics control section before the equaliser, and the default signal path follows the same order, but an illuminated button allows that sequence to be reversed should you need to equalise before compressing. The compressor is fully featured, with six rotary controls and a pair of illuminated buttons. The Threshold control spans +20dBu to -40dBu, enabling the compressor to be used effectively as a peak limiter at high threshold values and ratios. The Ratio control is scaled rather oddly from zero to 10, but actually offers a range of 1:1 up to 10:1, and has a hard-knee response by default. A soft-knee curve can be switched in when a more subtle effect is required. The Gain control offers a ±10dB range, and the Attack and Release controls are both marked simply with Fast and Slow at their control extremes. However, the attack-time range is detailed in the specifications as being adjustable from 0.1ms to 200ms, while the release-time covers 0.05s to 3s. The last rotary control determines the turnover frequency of the side-chain's highpass filter. This is adjustable from 10Hz to 10kHz, and is useful for reducing the compressor's sensitivity to bass instruments at the lower end of the scale, and for helping the compressor react only to sibilance at the higher end of the scale. The entire compressor can be bypassed by another button, and the amount of gain reduction can be displayed on the central VU meter. The equaliser occupies most of the panel space to the right of the VU window, with three sets of three rotary controls plus another pair of illuminated buttons. The first button places the EQ before the compressor, as described above, and the second bypasses the entire equaliser circuit. The controls in all three bands follow the same order left to right — Q (bandwidth) followed by Gain then Frequency — but the middle section is reversed vertically compared with the outside two. In other words, the Gain control is above the Q and Frequency controls for the centre section, but below for the other two. This caused some operational confusion initially, but once familiar with the unit I found it fairly easy to navigate the controls. I have already mentioned the ranges afforded by the Gain and Q controls, so to complete the picture the frequencies covered by each band are 20-300Hz, 200-3000Hz, and 2-20kHz respectively. As you would expect, the equaliser section is versatile and sounded pretty good to my ears — far better than the average budget mixing console's EQ, and capable of better high-frequency correction than most software plug-ins. The ability to vary the Q over a reasonably wide range is certainly helpful when trying to tune in to a specific part of the spectrum, allowing each EQ section to affect a broad range for gentle tonal shaping, or a narrow range for more delicate corrective surgery. The final control panel section is the output stage, with the output fader and a
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Presonus Eureka
button to switch the VU meter between output and gain-reduction levels. The output can sustain signals up to +22dBu from a low 51(omega) source impedance, which is the same as the balanced insert send point. The balanced insert return and line input both share a 10k(omega) impedance and can accommodate signals to +22dBu.
Verdict The Presonus Eureka doesn't have the larger-than-life low-frequency quality associated with true high-end preamps, but it remains well controlled at the bottom with a detailed top end and a neutral overall balance. I would liken the Eureka's preamp stage to that of a good-quality, traditional mixing console — there's something about a transformer input stage that just sounds right. The Eureka offers a worthwhile step up in quality from the relatively simple mic input stages of budget mixers and computer interfaces. The Saturation control can be used to smooth any hard edges from the top end, while adding an extra richness and density through the mid-range. With most sources — such as vocals and some instrumental sources like electric or electronic keyboard parts — the subtle distortion sounds musically flattering. However, it's not an effect to overuse: the addition of those extra harmonics not only thickens the sound, but also tends to obscure transient detail, so it can be damaging on complex and delicate sources — such as acoustic guitars (especially 12-strings) or crisp percussion parts. Overall, the Eureka is a very good-sounding unit — especially given the UK price — and easily stands up to comparison with higher-priced channel strips from the other major players in this field. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Preview: Korg Legacy Collection
In this article:
What's In The Legacy? Come In, Control The Sounds Final Thoughts
information Legacy Collection (Wavestation, Polysix and MS20 instruments, plus bundled scale-model MS20 USB controller), £399 including VAT. Korg UK Brochure Line +44 (0)1908 857150. +44 (0)1908 857199. Click here to email
Preview: Korg Legacy Collection Virtual Instrument Suite/Hardware Controller Published in SOS May 2004 Print article : Close window
Reviews : Keyboard workstation
Korg's Legacy Collection recreates some of their bestloved keyboards in software form. It wasn't finished as we went to press at the end of March, but we managed to get this sneak preview as the final touches were being applied...
www.korg.co.uk www.korg.co.jp
Paul White
Few software synth bundles have Original Photo: Mark Ewing created as much advance interest as Korg's Legacy Collection, and though it still wasn't ready to ship at the time of this preview, it was clearly very close and worked well enough for me to form a few impressions. Perhaps one of the reasons for the intense interest in the Legacy Collection is that Korg are modelling their own vintage instruments, but a major contributing factor has to be the highly cute bundled hardware controller, which comprises an 84-percent 'scale model' of the Korg MS20 monosynth, complete with working keyboard, controls and patchbay. This controller connects to the host computer via USB, though it doesn't have to be connected for the software to run. When it's finished, the Legacy Collection will be useable as 'stand-alone' applications, or as Direct X, VST or Audio Units plug-ins, but only for computers running Mac OS X or Windows XP.
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Preview: Korg Legacy Collection
Legacy comprises three instruments plus Legacy Cell, a kind of super virtual instrument that can be made by using one or two synths plus both insert and master effects. In all, 19 different effects types are available and include all the usual modulation effects, delay and reverb, plus a few slightly quirkier offerings, with 127 onboard factory presets based on these effects. All the effects are fully editable and there's an impressive bank of Cell presets that produce very rich, analogue polysynth-style sounds. Two of the included synths model old analogue keyboards, the MS20 monosynth and the Korg Polysix, a fairly simple but rich-sounding six-voice analogue machine with an onboard arpeggiator. I'm told that Legacy came about because a group of Korg designers had a fascination for creating software models of their favourite old instruments, which is probably why the end result is so authentic; Korg have used what they call Component Modelling Technology to recreate the sound and behaviour of the original instruments by modelling individual components. This is more processor-intensive than the alternative method of modelling the approximate behaviour of blocks of circuitry, but it does get a lot closer to the original sound. The only departure from authenticity is that both instruments may be used polyphonically with up to 32 voices (or 16 voices in unison), though you can restrict the voice count to lower numbers if you wish. One surprising result of this added capability is that the MS20 software becomes a real monster in polyphonic mode. If I had to compare it to anything, I'd say the poly MS20 sounded pretty close to an old Oberheim. Legacy Cell can create more complex layered and/or effected sounds using these analogue models either singly, in pairs or by using one of each. Both synths come with their original factory sounds, and of course you can store as many of your own sounds as you like, as well as creating new Legacy Cell patches. For me, the third instrument in the collection is the real prize — the much-loved Korg Wavestation. The Wavestation was an ambitious wave sequencing instrument that suffered from the limited processing power available at the time of its inception, which meant that although it was technically multitimbral, it really only had enough polyphony to be used as a single instrument if complex sounds were being used. It also had onboard effects that had to be deployed in a compromised fashion in multitimbral mode. Legacy models the original Wavestation and its effects exactly, including, apparently, some small design flaws that contributed to its unique sound. In this respect, Korg had the advantage that some of the original engineers were available to help out with the software. It contains the same wavetables as the original, and comes with all 550 preset performances found in the Wavestation SR. What's more, SysEx dumps of existing Wavestation patches can be imported. As on the original, the polyphony is limited to 32 oscillators (which equates to between one and 32 voices depending on the complexity of the Performance), but as the Multi mode has been dropped (it makes no sense in a software environment where you can open as many tracks of each instrument as you like), each instance has full polyphony rather than all the parts having to share.
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Preview: Korg Legacy Collection
Come In, Control We've come to expect software instruments with photo-realistic front panels, and the Legacy instruments are no exception, but all three instruments can also be controlled via the hardware controller. Obviously MS20 maps to the controller exactly; the patchbay, assignable wheel and assignable button work just as they did originally. Even the external input for filter and envelope processing has been recreated (though signals go in and out via your own audio interface rather than the jacks on the controller). The settings of Polysix and Wavestation can also be adjusted via the MS20 USB controller, but the controller mapping is inevitably a compromise and sometimes you just have to turn a knob to see what it controls. When the controller is being used to control MS20, inserting any of the included patch cables into the front panel jacks causes a duplicate cable to appear onscreen, though you can simply drag the virtual cables around to create connections if real cables are too untidy for you. In creating the hardware controller, the designers have been real sticklers for authenticity, but in some ways they may have gone too far, as there's only one control wheel, assignable to pitchbend or mod depth — but no way to control both at once. Having said that, I guess most users will already have a full-size controller keyboard with the necessary levers or wheels somewhere in their studio.
Legacy Cell.
Both of the 'analogue' instruments can be edited perfectly adequately via the on-screen representations of their front panels, as the originals came from the era when there was one knob or switch for each function. Wavestation is a different case, though. The original hardware keyboard was a menu-driven digital instrument that was notoriously hard to program, so there was little sense in recreating that interface. Instead, Korg have produced a dedicated graphical editor that makes creating new wavetables or patches very easy. Wave sequences can be set up using draggable envelopes for duration, amplitude and crossfade rate, but as with the original Wavestation, you need to be aware that changing an existing wave sequence will in effect edit every preset that uses the wave sequence. Because the Wavestation was a digital instrument, and because Korg still have the code for the original hardware version, the task of recreating the sound was perhaps easier (and less processor intensive) than modelling the apparently simpler analogue machines.
The Sounds
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Preview: Korg Legacy Collection
It's strange how instruments which are now regarded as classics never seemed quite so classic at the time (with the possible exception of the Minimoog). The MS20 always struck me as the budget choice for the player who really wanted a Minimoog but didn't have the money, but when you revisit these early Korg analogue synths, you realise that they had very rich, fruity-sounding filters and they were capable of some great sounds. Making the MS20 software polyphonic has really raised its game and it stands up well alongside some of the classic analogue polysynths, but things notch up another gear when you use Legacy Cell to either layer two together or to layer MS20 and Polysix, both with independent insert effects plus overall master effects. Some of the presets show off these hybrid instruments very effectively, and the end result often seems to be far greater than the sum of the parts. What's particularly impressive about Korg's modelling is the way the filters still sound analogue when the resonance is turned up close to self-oscillation. I always had a soft spot for the original Polysix, because it sounded far richer than the simplicity of its synthesis engine should have ever allowed. I also liked its built-in arpeggiator and the fact that it had proper knobs to control everything. The character of the original has been captured superbly in this software version, and the only slight disappointment is that the provided presets cover only the original 32 factory sounds, leaving you to make up the rest. As a current owner of three hardware Wavestations (two SRs and an AD), I was extremely keen to see if Legacy would allow me to move these to the 'store cupboard of eternal darkness' and replace them with their software counterparts, thereby eliminating hum and hiss from my studio as well as freeing up inputs on my system. What I heard from Legacy was in fact a revelation, because listening to those familiar Wavestation sounds played through a modern soundcard with decent converters and low-noise electronics was like peeling clingfilm from one of the great masterworks of art. Apparently the sample rate of the hardware Wavestation is only 32kHz, giving it an upper frequency limit of between 15 and 16kHz, but in the Legacy version, you can hear everything with a new clarity. Of course you could use the built-in top-cut filters to restore the 'warm', somewhat vague sound of the original, but I for one relish the chance to use these sounds in all their glory. This is one instance where the recreation is definitely better than the original! I also spent around an hour tinkering with the editor and found it really simple to modify patches. The same tweaks would have driven me to distraction if I'd had to do them via the SR's hardware interface.
Final Thoughts During this preview, I was only able to use the instruments in stand-alone mode, but that was enough to convince me that I had to have one — for me, Legacy would be worth the money just for the Wavestation alone, even without the rather lovely hardware controller. I can't fault what Korg have done within their design brief, but sometimes you can be too purist. For example, I'd have liked the ability to choose a Wavestation preset as the oscillator for one of the analogue synths,
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Preview: Korg Legacy Collection
which would more than make up for the fact that the Wavestation has no resonant filter of its own and that even envelope control is clumsy. Given that we now have very powerful computers, this type of mix-and-match approach should be technically possible. Perhaps these are options for future versions, but even as it stands, Legacy promises to set new standards in sonic authenticity and value. My credit card is primed and ready for the release version!
Look out for our full review of the completed Legacy Collection, which starts next month. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Rode NT2000 & NT1A
In this article:
NT2000 Multi-pattern Mic NT1A Fixed-cardioid Mic In The Studio Overall Impressions
Rode NT2000 £399
Rode NT2000 & NT1A Capacitor Microphones Published in SOS May 2004 Print article : Close window
Reviews : Microphone
pros Fully variable polar-pattern, roll-off and pad controls. Clear, well-balanced sound. Comes with rigid case and shockmount.
cons Weight may cause cheap mic stands to droop.
summary Rode's new NT2000 proves is a cost-effective, high-quality mic which not only sounds good, but which is also extremely quiet.
Rode NT1A £179 pros Transparent, classy sound. Extremely low noise. Nicely engineered. Included shockmount.
Two new large-diaphragm mics from Rode improve on their previous technology for a classy sound with exceptionally low noise. Paul White
Ever since the NT2, Rode have been stirring up the mic market, and their range now includes a variety of respectable valve and solid-state models priced for the project studio owner. It's been a while since the last time I reviewed anything from Rode — I looked at the NT4 and NT5 back in August 2002 — so when the NT2000 and NT1A turned up for review I was keen to see how their technology had advanced.
Photos: Mark Ewing
cons May lack the 'pose value' of some of the more chunkylooking capacitor mics. No hard case.
NT2000 Multi-pattern Mic
There's no mistaking the NT2000 — its distinctive three-dial control panel is quite unlike anything else The Rode NT2000. summary on the market! This solid-state microphone uses a The NT1A may have an entrylow-noise JFET design with bipolar transistor output level price, but it's buffers, so there's no transformer in the signal path. Rather than having performance is anything but entry level. It has a great switchable polar patterns, the uppermost knob allows the pattern to be varied sound, it is extremely quiet continuously from omni, through the centre-detented cardioid position, to figure of and it comes with a good eight. The mic can run from 24-48V phantom power. shockmount. information NT2000, £399; NT1A, £179. Prices include VAT. HHB Communications +44 (0)20 8962 5000.
The JFET preamp, which makes use of surfacemount technology, is extremely quiet, with just 7dBA equivalent input noise, and the mic's frequency graph shows a full 20Hz-20kHz frequency range with a gentle presence peak at around 12kHz. In cardioid mode there's also a lesser secondary presence lift at around
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Rode NT2000 & NT1A
+44 (0)20 8962 5050. Click here to email www.hhb.co.uk www.rode.com.au
5kHz and, where the low end needs to be rolled off to compensate for the proximity effect (which affects cardioid and figure-of-eight patterns only), the centre knob is used to adjust the low-cut turnover frequency from 20Hz (effectively bypassed) to 150Hz. This 'continuously variable' theme also carries over to the pad knob, which goes from zero to 10dB of cut. With the pad switched out, the maximum SPL is an impressive 147dB, increasing to a massive 157dB with the pad in. Overall, the dynamic range of the mic is 136dB and the signal-tonoise ratio works out at a very healthy 84dB. The mic is very solidly built and chunky, with a machined all-metal body finished in satin nickel plate. The controls make the body a little longer than usual, which in turn makes the mic quite weighty, so you'll need a good mic stand to avoid getting the droops! The capsule is protected and shielded by the usual dual mesh structure, and the capsule itself is shockmounted to further improve the isolation from stand-borne vibrations. A locking ring at the bottom secures the mic to the included SM2 shockmount, and the whole kit comes in a bespoke moulded plastic case. Dismantling the mic reveals a very tidy surfacemount circuit board with the Rode manufacturing logo, while the internal standard of metalwork finish is also good.
NT1A Fixed-cardioid Mic Rode's NT1A looks physically similar to the original NT1 — other than the finish, which is now satin nickel rather than grey lacquer. It is supplied in a cardboard box rather than a fancy case, and looks less macho than some of its competitors, but it comes with a good metal shockmount and a soft zip-up storage pouch. Like the original NT1, the NT1A is a cardioid-pattern, large-diaphragm capacitor mic with a gold-sputtered one-inch diaphragm. Despite the similarities with its precursor, the NT1A is no minor update — the circuitry is completely new, and the capsule has a much-extended frequency response compared with the original, giving it full 20Hz20kHz coverage. By contrast, the NT1 started to roll off above 16kHz. The response is also flatter than that of most large-diaphragm mics, as the presence peak is less pronounced and quite broad, but there's also a hint of low-end lift at around 120Hz, which gives the mic a very subtle 'smile EQ' characteristic, the practical outcome of which is that the sound is mildly flattering but still very natural sounding. Transformerless JFET circuitry is used to amplify the signal from the capsule, and the circuit board employs The Rode NT1A. surfacemount components. Niceties such as goldplated XLR pins and a heavy machined locking ring for fixing the mic to the shockmount have been retained, even though this is still file:///H|/SOS%2004-05/Rode%20NT2000%20&%20NT1A.htm (2 of 5)9/22/2005 6:08:32 PM
Rode NT2000 & NT1A
Rode's entry-level model. A very tough stainless-steel grill protects the capsule, with a second much finer layer beneath to help reduce popping and to help protect the capsule from contamination and RF interference. However, a separate pop shield is essential for studio vocal recording (as it is with all such microphones). There are no pad or low-cut switches on the mic body, as this is the baby of the Rode range, but it's also amongst the quietest mics I've ever used, with an equivalent input noise of just 5dB, which translates to a signal-to-noise ratio of 88dB. By comparison, many large-diaphragm mics have a signal-to-noise ratio of between 75-80dB. The sensitivity of the mic is quoted as -31.9dB (with a 1V/Pa reference) and the sensitivity was subjectively very similar to other capacitor mics in my collection. The maximum SPL is 137dB, which is about as good as it gets for a capacitor mic with no pad switch, and the NT1A will work on 24-48V phantom power.
In The Studio I tested both these mics side by side and found that the sound in cardioid mode was quite similar — there was a wonderful sense of high-end detail, but without the sound becoming thin or harsh. In fact I'd say both models produce what you'd expect from a well-designed transformerless FET mic, but with rather lower background noise than most of the competition. The somewhat constricted character that most cardioid mics exhibit to some degree seems less noticeable in these new Rode designs, but when you switch the NT2000 to its omni pattern, you quickly realise that omni-pattern mics still sound more open and natural than cardioids. The same is true of the figure-of-eight mode, which sounds very clean and pure. Of course many people buy large-diaphragm mics because of their character, and some models have much more of a 'sound' than others. To me, these new Rode designs are characterised by a very open and detailed high end, which is nevertheless properly balanced by the low end. There's nothing thin or edgy about the sound and, because there's no excessive inherent coloration, they respond well to EQ — you're not constantly trying to fight the mic's in-built character. On most voices, the sound comes over as gently flattering and well focused, and of course this attribute bodes well when you come to mic acoustic guitars and other instruments. I like the idea of the continually variable controls on the NT2000, the only slight downside being that settings become less repeatable if you pick up a session months after starting it. However, the manual does include a blank settings page you can photocopy to write down your custom settings. There's also a lot of practical advice in the manual about using the mic, including a reminder that you still need to buy a separate pop shield for studio vocal recording. As an all-round instrument mic, the NT1A has a brighter, more articulate high end file:///H|/SOS%2004-05/Rode%20NT2000%20&%20NT1A.htm (3 of 5)9/22/2005 6:08:32 PM
Rode NT2000 & NT1A
than the original NT1, but the low end remains well balanced, resulting in a clear, believable sound. However, I suspect most users will buy the NT1A predominantly for use as a vocal mic, where it delivers a very modern sound, balancing transparency and airiness with a supportive lower mid-range. There's certainly nothing budget or second class about the sound of this microphone, and where you do need to emulate the somewhat warmer, more middly sound of the original NT1, a little high-end EQ cut gets you pretty close. From a practical viewpoint, the NT1A is a little lighter than many of its contemporaries, which makes mic-stand sag less of a problem, and the very low noise means it won't struggle when recording quieter instruments or when working at a moderate distance from the sound source. It's transparent high end also enables it to do a great job on acoustic guitar and hand percussion.
Overall Impressions Rode's new designs have really increased expectations of what can be achieved in microphones at this UK price, particularly when it comes to circuit noise — they are typically 6-10dB quieter than most of the competition. You don't get the fudgy warmth of a tube mic or even a transformer-coupled model, but to my ears this makes it easier to place the vocals at the front of a mix without using excessive volume. Despite its low cost, the NT1A delivers professional performance, both for vocal recording and for general instrument use. The lack of any heavy-handed presence boost makes the sound well suited for use with a range of singers and vocal styles and makes it's also easy to fine-tune using modest amounts of EQ. At the same time, the high end is as open and detailed as you could wish for, so if you like a vocal sound with a modern breathy quality, you can achieve it using little or no EQ. If you need a good all-round, multi-pattern mic that will do justice to just about anything you ask of it, in addition to delivering pristine vocals, then the NT2000 is one mic you shouldn't miss checking out, but if you're one of the many people for whom a cardioid-only mic is perfectly adequate, then the cheaper Rode NT1A has a very similar character, with extended bandwidth, extremely low noise and a classy subjective sound. Published in SOS May 2004
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Rode NT2000 & NT1A
Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Tascam 2488
In this article:
Tascam 2488
25 Years Of The Portastudio 24-track Digital Portastudio Onboard Preset Effects Published in SOS May 2004 Interfacing Hardware Control Surface Print article : Close window Initial Impressions Reviews : Software Channel Facilities Monitoring Options Disk Management & USB 2 File Transfer Flexible Effects Tascam's new Digital Portastudio is now the most The 2488 In Use affordable 24-track workstation on the market. In our Previous Tascam exclusive hands-on review, we find out how easy it is Multitracker Reviews In SOS to use, and whether its sound is as impressive as its Writing Mixes To CD track count. Portastudio Power
Tascam 2488 £999 pros Versatile, immediate, and easy to use. Multi-effects processor for use with the DI input. Built-in MIDI tone generator. Onboard CD-RW drive. Flexible import/export facilities and USB connectivity.
cons No moving faders. Effects and analogue circuitry adequate, but not sparkling. Difficult to see rear-panel connectors when leaning over from the front.
summary Tascam have redefined the Portastudio by combining 24track hard disk technology with a versatile digital mixer, an array of internal effects, a built-in CD-RW burner, and a General MIDI tone generator.
Hugh Robjohns
To home recordists of a certain age, Tascam will always be associated with the Portastudio. Many of us cut our recording teeth on the original fourtrack cassette Portastudio in the early '80s, before moving on to four- or eighttrack reel-to-reel recorders — usually still bearing the Tascam logo. Tascam also maintained an interest in the professional side of the industry too, and were influential in the standardisation of the 24-track two-inch analogue multitrack.
Photos: Mark Ewing
However, times have changed and technology has moved on in leaps and bounds, but Tascam have neither forgotten nor ignored the needs of the home recordist. Their latest product is intended to bring to that market a simple-to-use, yet professional-quality, 24-track hard disk Portastudio, with some unusual extra facilities. With many modern musical styles comprising complex layers of loops, guitar overdubs, synth lines, and effects, 24 tracks is rapidly becoming a common requirement even in home studios, and while computers have their advocates, for many musicians a dedicated hardware recorder is a simpler and less frustrating way into music recording.
information £999 including VAT. Tascam +44 (0)1923
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Tascam 2488
438880. +44 (0)1923 236290. Click here to email www.tascam.co.uk www.teac.co.jp
Test Spec Tascam 2488 OS v1.00, build 0060. 2MHz Pentium IV PC with 1GB RAM, running Windows XP Home.
25 Years Of The Portastudio The new Tascam 2488 has been launched exactly 25 years after the original cassette Portastudio and, while the basic concepts of multitrack recording are similar, the detailed operation of the latest model is a revolution in comparison. The 2488 enters the market as the most affordable integrated and transportable 24-track workstation, combining a 24-track hard disk recorder, operating at 24-bit resolution and 44.1kHz sampling rate (although there is a 16-bit mode, if required); a 36-input digital mixer, with four phantom-powered mic/line inputs and four additional mic/line inputs, one of which has a dedicated DI input; versatile built-in effects; and a CD-RW burner. In addition to the 24 real tracks, the system also provides up to 250 virtual tracks to enable multiple takes and comping of vocals, for example. Among the more unusual features for a machine of this type, the 2488 also incorporates a 16-part, 64-voice General MIDI tone generator, complete with an onboard Standard MIDI File player and metronome. This enables MIDI files to be loaded and used as backing tracks, for example. Like most products of this type and price, there are inherent compromises and constraints in its design and operation that make it a little less flexible than the high-end equivalents, but for the vast majority of practical purposes in its intended market these will largely be irrelevant. The most pertinent factors, which are immediately apparent from a cursory examination of the control surface, are that the 2488 can record only up to eight sources simultaneously, and it has just twenty physical (and non-motorised) faders for mixdown — the last six each control stereo channel pairs. The configuration options for the internal effects processors are also rather convoluted, and the effects — although adequate — do not really compete with the best multi-effects and reverb processors available. Having said that, the 2488 does have provision for two external effects sends, and the eight input channels can be used as effects returns on mixdown, so there's nothing to prevent the user from adding high-quality outboard effects if required. For most of the intended users though — novice recordists — these limitations will rarely be restrictive in practice. The internal digital mixer's 36 channels are sufficient to enable mixing of the 24 replay tracks, the eight 'live' inputs, the onboard MIDI tone generator, and the internal effects processor. Each mixer channel is equipped with phase reversal, pad, basic three-band EQ (high and low shelves, plus a fully parametric mid-band), three aux sends, and an insert point (to accommodate an in-line effect or dynamics processor). I'll describe the effects configurations in more depth later, but basically the processors can be set up in three ways: either as eight separate channel insert effects; as four channel inserts plus a multi-effect (intended for combination guitar effects); and as a
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Onboard Preset Effects CHANNEL INSERT EFFECTS: Compression Exciter De-Esser
MULTI-EFFECTS: Distortion-Flanger Distortion-Phaser Distortion-Chorus Distortion-Exciter
Tascam 2488
single high-quality send/return loop effect.
Distortion-Pitch Distortion-Tremolo
Currently, the 2488 is shipping with a 40GB hard drive which is claimed to be sufficient for over twenty full 24-track Songs. However, a very welcome feature is that the unit also incorporates a USB 2 port specifically to enable data back-up or audiofile import/export via a computer workstation — either for archiving or for further production work. The CD-RW drive installed in the centre of the front panel below the faders is for archiving and for burning audio CDs.
Distortion-Vibrato Distortion-Wah Compressor-Flanger Compressor-Phaser Compressor-Chorus Compressor-Exciter Compressor-Pitch Compressor-Tremolo Compressor-Vibrato Compressor-Wah
LOOP EFFECTS:
Interfacing Hardware
Reverb Delay
When first unpacking the 2488 it looks a little 'plasticky', but it quickly becomes apparent that it is actually put together rather well, and it feels fairly solid. It weighs about 8kg and measures a compact 145 x 545 x 355mm (hwd). The only supplied accessory is an IEC mains cord, so you'll need a USB cable if you want to be able to export data directly to a computer, and some CD-R or CD-RW blanks for the CD mastering and archiving functions, respectively.
Chorus Pitch-shift Flanger Phaser Gate+Reverb Compressor-Wah Tuner
As always, my eye was drawn first to the rear panel, since the connectivity is what defines a machine of this type. There are four combi-XLR inputs (XLR and TRS jacks, the former with globally switchable phantom power) and four more TRS inputs. All eight can accommodate both mic and line signals, but the XLR inputs are slightly more sensitive. All eight jack sockets present an 8k(omega) input impedance and expect nominal signal levels between -43dBu (with the toppanel gain knob at the Mic end of its range) to +4dBu (at the Line end). The maximum input level is a fairly healthy +20dBu. The XLR mic input presents a pretty typical 2k(omega) impedance and offers a sensitivity range from -57dBu to -10dBu, with a maximum input of +6dBu. The preamps are adequate for typical close-mic recording techniques, with reasonable headroom and low noise. At the end of the same row as these input connectors is the USB 2 socket mentioned earlier, and all of these sockets are protected by the protruding top panel. This protection could be useful in some circumstances, I suppose, but it also makes it very hard to plug things up from the front, because you can't see the sockets or read the labels. It all has to be done by feel. The outputs are file:///H|/SOS%2004-05/Tascam%202488.htm (3 of 13)9/22/2005 6:08:41 PM
Tascam 2488
below the input connectors, but this part of the rear panel is recessed even further, by about another half inch, making it completely impossible to see the sockets when peering over the back from the front. Again, not a problem if the machine is installed and left alone, but a source of annoyance if you need to be able to access and change the connections regularly. On this lower row are two unbalanced quarter-inch sockets providing effects sends (at a nominal -10dBu level), two phono sockets for the main stereo mixer output (again at a nominal -10dBu to match domestic or semi-pro recorders), a pair of balanced TRS sockets for the monitoring output (nominally at -2dBu), and another pair of phono sockets providing S/PDIF digital input and output. The digital input can be allocated to replace any odd-even pair of the eight analogue inputs, but oddly it is not made available to the monitoring as a two-track return, which might have been a useful facility. Finally, there are two MIDI sockets (In and Out), the IEC mains inlet, and the power switch. The power supply is factory configured to operate on the appropriate mains voltage for the territory in which it is sold, and is not useradjustable. The machine consumes 41W of power. The front edge of the 2488 contains a few more quarter-inch sockets. A brightred TRS socket on the left provides a headphone output (generating a maximum of 55mW per channel into 30(omega)), while on the right below the transport controls are a dedicated guitar DI input and a pair of footpedal sockets — one for punching in and out, and the other for an expression pedal. The DI socket overrides the eighth input channel and has a 1M(omega) impedance, with a gain range from -55dBu to -8dBu. The expression-pedal socket is intended to allow a foot controller to alter some of the effects parameters in real time — such as the wah-wah effect. A thoughtful and useful facility.
Control Surface The control surface is uncluttered and has clearly defined functional areas with very legible labels. At the top left of the panel are eight input level controls, plus a button to activate phantom power on the four XLR inputs. Immediately below are eight illuminated buttons (complete with small overload LEDs) to select, link, and route the input sources, while a further three buttons are used to access and configure the effects section. Below these are three large rows of 19 illuminated buttons, making this unit look more like a TV vision mixer than an audio recorder! The top row deals with record enabling; the middle row is used to select mixer channels and configure track routing; and the bottom row provides the mute and solo functions. The buttons mute by default, and the solo function must be activated with a button to the right of the stereo output fader. Below these button rows are twenty closely spaced and short-throw (45mm travel) faders, which are not motorised.
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Tascam 2488
Above the button array is what I would describe as a small (roughly two and a half inches square) backlit LCD screen — although Tascam seem to think it 'large' — with fairly restricted contrast adjustment and viewing angle. It's fine for single-person operation, but if you have a friend sitting next to you they will have trouble reading the screen. Above the main stereo output fader are three small buttons to select the various mixer-channel function displays (EQ, Send, and Fader/Pan) plus the button for activating the solo function. Many of the workstation's buttons have dual functions, and these are marked with secondary reversed-blue legends. They are accessed by pressing and holding a dedicated Shift button just below the screen. To the right-hand side of the machine are the usual selection of transport controls, menu navigation buttons, and data-entry/jog wheel, as well as the monitoring section. Again, there's plenty of space around everything, Here the three editing screens for the 2488's which makes it easy to find and use internal GM sound module have been laid the facilities. The monitor section end to end so that you can see all the sports a large rotary level control with available parameters— parts 12-16 can be edited by scrolling the display using the three associated buttons. The first selects the monitoring source, and also cursor keys. provides a mono check facility as a secondary function. The Record Source Monitor button allows you to audition the dry input tracks when recording, and there's also a monitor Mute button. A row of four buttons below the monitoring section provide various system functions, with the all-important Shutdown button tucked up in the right-hand corner — pressing and holding this ensures that everything is safely written back to the hard drive before the power is turned off. Just below the Shutdown button is the Home key, and this takes the display back to the basic mixer metering screen. Below the jog wheel, cursor and Yes/No buttons are various facilities to set locators for automatic drop-ins and basic track editing functions (including delete, insert, trim, move and so on). A Tap button lets you adjust the timing parameters of appropriate effects intuitively, and there's a built-in metronome. It's all fairly familiar and generally intuitive stuff, and after a few minutes of prodding around I was able to start making recordings and overdubs without any problems at all.
Initial Impressions Powering the machine up takes about 25 seconds. During that process the software version is displayed — in this case v1.00, build 0060 — and at the time of writing there were no OS updates posted on Tascam's web site, which bodes well. The 2488 has no cooling fan, and the internal hard drive is remarkably quiet, even when recording or playing back. This feature immediately gives the workstation a distinct advantage when compared to many of the earlier generation of comparable workstations — and I had no problems at all recording
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Tascam 2488
in the same room with the 2488. The unit becomes barely warm after prolonged use, so no problems there either. I have to say that the internal signal path is not entirely intuitive, but there is a useful block diagram in the back of the handbook that helps. Essentially, the eight analogue inputs (and the internal MIDI tone generator source) are each connected directly to their own input channels, each with phase reverse, pad, effect insert, and EQ facilities. At this point the input signals can either be routed directly to the appropriate record tracks, or passed on to a submixer to be used as live inputs at mixdown. The latter route adds level and pan controls before contributing to the stereo submix buss. There are also facilities here for the three aux sends: the first routes to the internal stereo effects loop send bus, while the second and third route to the two external effects sends outputs, each with pre/ post switching. The twenty-four track-replay channels have similar facilities to those provided in the submixer. The main output can be sent to either the stereo output or submix busses, and the latter is used when bouncing down recorded tracks, should 24 not be enough. Although the main stereo output normally appears at the main rearpanel output connections (and the digital output), there is also a routing facility which enables any odd-even pair of tracks to be presented at these connectors instead, as a direct output. The eight analogue input channels and the tone generator are freely assignable to any of the 24 recording tracks by the simple process of holding the relevant input source button until it flashes, and then pressing the required channel selection button. A source can be routed to more than one track if required, and the function works equally well if you hold the mixer button and then select a source instead. For stereo signals, sources and tracks can be linked in odd-even pairs by holding down one button and pressing the adjacent one. Input signal routing can be interrogated by holding down either a source or mixer channel button, and observing which other buttons flash, or by calling up a Map display, a two-screen depiction of the entire routing status. There are also three pre-configured routing assignments, which can be overwritten if required and which can be recalled using the Quick Routing button. The first of the three modes is called Recording and simply assigns inputs one to eight directly to tracks one to eight, respectively. The other modes are Bounce, which picks up the submix buss and routes that to the appropriate record tracks; and Mixdown which routes the input channels to the main stereo output, via the submix buss, along with the replay tracks.
Channel Facilities The function of the channel phase reverse is self evident, but the pad facility is worthy of a brief mention. Accessed from the EQ display page, this setting provides coarse gain adjustment in simple 6dB steps from +6dB to -42dB. When file:///H|/SOS%2004-05/Tascam%202488.htm (6 of 13)9/22/2005 6:08:41 PM
Tascam 2488
applying significant amounts of EQ boost, it is often necessary to introduce some signal attenuation to avoid overloading the output converters and internal mix busses. The maximum of 6dB of gain usefully enables slightly low-level recordings to be remedied too. The equaliser section for the selected channel is accessed by a dedicated button to the right of the main output fader strip, and boasts high and low shelving plus a parametric mid-band. All of these bands have a ±12dB gain range, which can be adjusted in 1dB increments. The frequency ranges of the three bands are 32Hz1.6kHz, 32Hz-18kHz, and 1.7kHz-18kHz for the low, mid-range, and high bands respectively. The mid-band's Q control offers settings from an extremely narrow 16 to a very broad 0.25. Although relatively simple, the EQ is usable and flexible, allowing both gentle tonal tweaking and fairly fine spectral surgery, and it sounds quite respectable. The entire EQ section can be bypassed, either by moving the cursor to the on-screen bypass button and turning the data wheel, or by holding the Shift button and clicking the Yes or No buttons — although this only works while the relevant EQ page is displayed on the screen. The channel's EQ settings can be completely flattened by holding the Shift button and pressing the EQ button, which is handy too. A library of EQ presets gives you access to factory settings and you can add your own favourite settings to these. The Send screen for the selected channel is accessed from its own dedicated button, and reveals the three virtual aux send controls, each with channel send, pre/post switching, and aux master level control. The loop effects send is actually stereo, but there is no separate pan control, as this follows the channel's own panning. The third channel screen is accessed with the dedicated Fader/Pan button, and reveals the fader and pan positions, plus the phase reverse selection, for four channels at a time. Since the 2488 has scene memories that store fader positions (among all the usual other things: input routing, EQ and effects settings, pan, and so on), but the machine doesn't have motorised faders, indicators on the screen show any disparities between the physical fader and the virtual fader positions. Needless to say, the two then have to be matched or 'nulled' by hand. This process is helped by a dedicated menu screen that allows the faders to be nulled without changing the audio levels, but it is a process which is tedious and fiddly in the extreme! Obviously, motorised faders are expensive and the target price of the 2488 precludes their use, but it is a frustrating compromise all the same. I also found that having nulled one fader and moved on to the next, the null error indicators often reappeared on previously adjusted channels! Fortunately, there are also alternative fader modes which may make life easier in certain situations. The first is simply to allow the virtual faders to jump to the physical fader positions as soon as a fader is moved, and the second is to enable the virtual faders to be 'caught' when the physical faders pass through their stored positions.
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Tascam 2488
Monitoring Options The monitor section is fairly straightforward, with facilities to audition the main stereo output, the internal submix buss (useful when bouncing tracks), and the effects sends (both to the internal send effect and to the two external sends). A fifth option is to deselect all the monitoring sources, muting the monitoring completely — which would appear to make the dedicated Mute button redundant! In fact, though, the Mute button serves to kill the rear-panel output (normally used to feed loudspeakers) while leaving the headphone output functioning normally. The idea of the Record Source Monitor button, which lets you monitor the input source channels rather than the track replay channels, is that you may want to record dry, but introduce reverb or some other effect on the track replay channels (to help the vocalist with tuning, for example). By switching to the Record Source Monitor mode, you can still check the quality of the input signal, free from any distracting effects. If any channel is soloed, it automatically breaks into the monitoring chain as you would expect. This is a true 'solo in place' mode, allowing the panning and fader levels to be assessed.
Disk Management & USB 2 File Transfer The 2488 is equipped with various disk-management functions to optimise and format the internal drive. There are actually two disk formats used simultaneously. A FAT32 partition is used for the external file transfers via the USB port, and this is restricted to 4GB in size and cannot be changed. File names must conform with the 8.3 style ('filename.txt'). A number of Tascam native-format partitions take up the remainder of the disk capacity and are used to store audio and Song data. These may be any combination of four, eight, 16 or 32GB in size, but a maximum of only four partitions may be allocated. The native partitions cannot be seen or accessed by a host computer via the USB port. The disk may be reformatted in its entirety, or the FAT32 and native partitions formatted separately. The USB port is normally disabled, and must be switched on and off specifically at the start and end of file-transfer operations. While in use, the drive is effectively connected as an external drive to the computer, and no recording or playback operations can be performed on the 2488 at all. The USB import/ export functions can handle standard WAV audio files (16-bit or 24-bit resolutions) and standard MIDI files. Once imported to the FAT32 partition, the
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Tascam 2488
USB port is closed, the machine restarted, and the files transferred to the native partition. A similar process is required for transferring MIDI files to the internal player and tone generator. Backup and restore functions operate in a similar way through a Data Backup menu. You can copy material to or from the CD-RW drive, or via the USB port. The required Song files are identified and transferred (via the FAT32 partition in the case of USB exports).
Flexible Effects Effects processors can be inserted into the appropriate channels simply by holding the required effect mode button and pressing the required channel selection button. When the effects are configured as eight channel insert effects, the processor can provide compression, de-essing or exciter functions (selectable independently in each of the eight installations). The compressor mode is equipped with Threshold, Ratio, Attack and Gain parameters — the last three simply numbered from zero to 100. Sadly, there is no display of the amount of gain reduction either, which can make fine-tuning the settings more difficult than it really needs to be. Again, there is a library of factory preset configurations to which you can add your own setups. The de-esser and vocal exciter modes both provide simple frequency and depth controls. If you only need four such insert effects, then you can substitute a single multieffect processor for the other four. This multi-effect is most likely going to find its way into the guitar DI input, and offers a chain of effects starting with a noisegate followed by either distortion or compression. Then comes a simple amp modeller and one of several modulation treatments, with a delay rounding things off. All these effects are usable, and will turn a very dry guitar DI signal into something with rather more body and interest, but we aren't talking Line 6 PodXT standards here. If you are happy to abandon channel insert effects entirely, then you can set up the effects block as a single send/return processor which provides somewhat higher-quality effects. Algorithms include reverb, pitch-shifting, delays, and so on, and are accessed by dialling in an amount of Loop Send from any appropriate mixer channels. There is no dedicated effects return channel as such, but the level of the chosen effect can be adjusted in the relevant control screen. The quality of the reverbs and other effects are, once again, adequate, and sufficient for the intended 'novice' level of user, but fall rather short of the standards provided by some of the more up-market digital effects processors — including those in Tascam's own mixers, like the DM24. However, as the requirements and demands of the 2488 user grow, it is a trivial process to integrate higher-quality outboard effects using the two external sends and the input channels. Although not an effect as such, there is also a built-in guitar tuner facility which is accessed through the effects section buttons. In a similarly non-related vein, it is file:///H|/SOS%2004-05/Tascam%202488.htm (9 of 13)9/22/2005 6:08:41 PM
Tascam 2488
worth pointing out that Tascam have chosen wisely to retain a separate and permanent stereo compressor in the main output channel.
The 2488 In Use As with many other multitrack workstations, before starting a recording you first have to define a Song. With a Song created, the task of recording can begin by allocating input channels to tracks, arming the appropriate tracks, and pressing play and record. It's all pretty logical and painless, and I enjoyed using it largely because it just gets on with the job. There are none of the inherent distractions and foibles of most computer workstations, and as a result I found I simply got on with the business of making and recording music, which was a nice change! The 2488 follows the common path of allowing up to 250 virtual tracks per Song, which means that multiple takes can be recorded and retained, combined or discarded as required. A dedicated menu page is used to control the virtual tracks, allocating each selected virtual track to one of the 24 playback tracks. Finding your way around the recording is made easier thanks to quick locator The CD-RW drive lets you burn your mixes functions like 'return to zero' and 'last to audio CD in disc-at-once or track-at-once recording position'. You are also able modes, and also lets you back up your Song to set and name markers (up to 999 files. per Song), and I particularly liked the facility to recall pre-defined marker names from a list (eg. chorus, verse, middle eight, and so on), which made this tedious but essential task quick and easy. The time display can be switched between normal timecode (hours, minutes, seconds, frames and sub-frames), or bars/beats and tempo. In and Out points can be defined for editing and punch-in operations (with userdefinable pre-roll and post-roll times), and there is a fair waveform display and jog facility which enables slow-speed scrolling and auditioning to identify precise locations in the selected track. The waveform display can be zoomed both vertically (to reveal quiet recordings) and horizontally (to improve time resolution). Related functions are the varispeed and Slow Speed Audition (SSA) facilities. The varispeed control offers real-time adjustment by up to ±6 percent, with the corresponding pitch-shift. The SSA facility retains the correct pitch, but still allows the speed to be adjusted, and is intended to assist analysing or rehearsing particularly tricky solos! The range is in 10 percent steps down to 50 percent of normal speed. This facility only works on selected odd-even track pairs, and sounds as metallic and fluttery as most pitch-shifters do, but it is a file:///H|/SOS%2004-05/Tascam%202488.htm (10 of 13)9/22/2005 6:08:41 PM
Tascam 2488
useful feature nonetheless. The 2488's editing functions are as intuitive as everything else, and few people are likely to have trouble understanding the concepts of copy, move, paste, insert, open, cut, silence, or clone. The editing operations are controlled by just three points: In and Out to identify the required source audio, and To, which sets the destination start point for the copy and move functions. Relocating verses or solos is very simple and surprisingly quick — and it's all non-destructive, with a comprehensive multiple undo facility in case it all goes horribly wrong!
Previous Tascam Multitracker Reviews In SOS Studio 8 reel-to-reel multitracker, January 1986. Porta 2 Portastudio cassette multitracker, December 1986. Porta 05 Portastudio cassette multitracker, March 1988. 238 Syncaset reel-to-reel multitracker, August 1988. 644 Midistudio cassette multitracker, October 1989. 488 Portastudio cassette multitracker, March 1991. 464 Portastudio cassette multitracker, June 1992. Porta 07 Portastudio cassette multitracker, June 1993. 488 MkII Portastudio cassette multitracker, March 1995. 424 MkII Portastudio cassette multitracker, September 1996. 564 Portastudio cassette multitracker, December 1996. 414 Portastudio cassette multitracker, January 1997. 788 Portastudio digital multitracker, February 2001. 424 MkIII Portastudio & 414 MkII Portastudio cassette multitrackers, March 2001. Pocketstudio 5 digital multitracker, November 2002.
Writing Mixes To CD With a complete mix built up, the next stage is mastering to the onboard CD-RW drive. The system assumes that the start will be at 00:00:00:00 and the end at the defined Out point, although it would have been more flexible if the start point could have been defined by the In function. A degree of editing and gap removal from the start of your recording may therefore be required before mixing and mastering. The first stage is to create a stereo mix on the hard drive in a single pass, using
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Tascam 2488
the Premaster menu function. Once created, it is then possible to trim the start and end in the usual way, before committing the track to a CD-R or CD-RW. The 2488 supports both track-at-once and disc-at-once operations, the latter being required to create a master for a duplication factory. The CD-RW drive is a slowspeed type, and the handbook recommends CD-R media optimised for use at 12x speed or lower (and 4x or lower for CD-RW discs). The disc can be finalised automatically at the end of a disc-at-once process or manually, and there are facilities to playback the completed CD, if required. The whole process of burning a CD is straightforward and logical, and makes the process of recording, mixing and mastering music remarkably easy and pleasurable! The internal CD-RW drive can read normal audio CDs, as well as CD-ROMs — but only Mode-1 discs, which may preclude loading audio from certain kinds of sample libraries. An addendum for the handbook (found on the web site) recommends not using the front-panel CD-RW eject button, but only the software CD Eject menu function. Apparently manual ejection has been found to upset the operating system under certain conditions.
Portastudio Power All in all, I found the 2488 to be a well-designed machine, ideally suited to its intended market. Some aspects may cause a little frustration — the hidden rearpanel connectors and the fader nulling, for example — but at this UK price point these are niggly points and can be accommodated without much effort. This machine is capable of making some fine-quality recordings, with plenty of real and virtual tracks for even the most complex of productions. The internal effects processors are competent and versatile, if not exactly of class-leading quality, and are certainly sufficient for most people's requirements, especially as there is the ability to add higher-quality outboard facilities if needed. The overall operation is very simple, intuitive, and quick, allowing the user to concentrate on making music rather than grappling with recording technology, and the ability to mix internally and burn the result to a CD-R, all in the one highly portable package, will be greatly appreciated by anyone starting out in home recording. With the 2488, Tascam have further refined the Portastudio format, and that can only be a good thing for recording musicians. Published in SOS May 2004
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Tascam 2488
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All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Waves IR1
In this article:
Impulse Responses Choose Your Space Usability Taking It Further In Action
Waves IR1 Convolution Reverb Plug-In: Mac & PC Published in SOS May 2004 Print article : Close window
Reviews : Software
Waves IR1 £699 pros Easy to use. Flexible controls offer more range than the majority of convolution processors. Great-sounding set of impulse responses covering just about every size and type of space.
If you want reverb 'sampled' from real spaces, there are now quite a few options available. As you'd expect, however, Waves have managed to come up with something a little special — a convolving reverb that offers detailed control over the sound.
cons Like all convolution processors, IR1 can be quite CPU-hungry.
Paul White
Now that we all have processing power to burn (at least until new and more Waves have introduced some powerful software eats it up again!), it genuine innovations into convolution reverb, giving the seems as though the process of user almost as much control convolution is the answer to getting as with a conventional seriously high-quality reverb running on synthetic reverb unit. a desktop studio system. Convolution information makes available the sampled acoustics HTDM version £1051.63; of real spaces, and also makes it other versions £699.13. possible to clone the sound of hardware Prices include VAT. reverb presets, but its down side is that there's a limited amount of adjustment Sonic Distribution +44 (0) available. That's because convolution works by 'multiplying' your audio file by an 1525 840400. impulse response recorded in a real reverberant space, and as the impulse +44 (0)1582 843901. response is, in effect, just another audio file, the amount of adjustment available Click here to email to the user is often limited to shortening the reverb decay and adding EQ. summary
www.sonicdistribution.com
www.waves.com
Test Spec IR1 version 1.0. Apple G5 dual 2GHz with 2GB RAM, running OS 10.2.7. Tested with Emagic Logic Pro v6.4.1.
Waves seem to have overcome at least some of these limitations in their new IR1 convolution reverb plug-in, which is available for Mac OS X and Windows in RTAS, Audiosuite and VST formats and also in Windows Direct X and Mac OS HTDM, MAS and Audio Units versions. IR1 allows the user to both shorten and lengthen the reverb RT60 decay time, adjust the room size, apply complex decay envelopes, emphasise room mode resonances, adjust the damping and change the relative levels of the early reflections and the decay tail. I can understand how some of these things could be achieved by applying resynthesis methods to stretch or squash the impulse response, but the IR1 designers have clearly gone
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Waves IR1
much further than this. However, it is clear that virtually all these things are achieved by processing the impulse response itself, as a new impulse response is calculated after every adjustment — a process which takes a second or so. Not only is IR1's reverb engine rather special, then, but a lot of innovation has also gone in to creating the impulse responses themselves, many of which were recorded in world-famous buildings — the full list is reproduced elsewhere in this review. Arranging access to these buildings is no trivial matter, so a process was devised that would produce the very best results, and which would also translate to future multi-channel surround formats. The IR recordings were made by Professor Angelo Farina of Parma University, who developed an improved method of capturing the impulse responses to minimise noise contamination and harmonic distortion, presumably by using slower sweep signals and/or correlating multiple sweeps rather than the simplistic 'bursting a balloon' approach. All the IRs were sampled at 96kHz and a comprehensive set of 360degree surround measurements was taken (one measurement every 10 degrees), which is why his impulse responses can be used for surround as well as the present stereo system. Professor Farina's system is also claimed to capture the early reflections in a way that preserves stereo directional information more effectively, though no additional information is available on how this was achieved.
Impulse Responses The full list of buildings and ambiences 'sampled' for the IR1 impulse library is as follows: Concert halls Kirishima. Uhara. Rome Santa Cecilia Auditorium (small, medium and large halls). Clairmont Hall. Herbert Zipper Hall. Ingram Hall. Sydney Opera House Concert Hall.
Auditoriums Atherton. Parma Auditorium. Ryman Auditorium (Grand Ole Opry).
Theatres Noh Drama Theater. Rishon Letzion Heichal Hatarbut. Wells Fargo Pavilion.
Small rooms Knights Halls (small hall).
Choose Your Space
Hamon Studios (small concrete room). JamSync Bathroom.
The IR1 library comes on two CD-ROMs and includes over 60 room types, plus 60 impulses from classic hardware reverb devices hardware presets. If you exhaust the possibilities of the library, you can also import third-party impulse responses in WAV form. As with any convolution reverb processor, you can also do lots of weird things by using short snatches of audio file:///H|/SOS%2004-05/Waves%20IR1.htm (2 of 6)9/22/2005 6:08:47 PM
JamSync Florida Room.
Medium halls Azrieli Hall. Knights Halls (medium hall and Refectorium). Sydney Opera House — The Studio.
Churches
Waves IR1
instead of impulse responses. When I first heard how the recordings had been made, I asked about the possibility of surround reverb, but was told this was a project for the future. However, some of the halls are sampled both from close to the front and from seats further back, so it may be possible to cook up something convincing in surround by setting up two stereo reverbs on two sends, one providing the front-of-room reverb and one the rear. If positioned correctly in the surround mix, this could be quite convincing, even though it isn't true surround reverb.
Belle Meade Methodist Church. Presidio Chapel. St. John Church. St. Paul's Chapel. Stanford Memorial Church. Trinity Church.
Recording studios Mishkenot Shaananim Music Center. Hamon Studios. Cello Studio 1. Masterfonics Studio tracking room. Schnee Studios. Signet Sound Studio A.
Usability
Sound Emporium Studio B Stockton Studio C.
IR1 has a straightforward user interface that offers controls very similar to those you'd fine on a traditional synthetic reverb unit, the main difference being that you have to load an impulse response of the type you want to use before you can make any adjustments. A 'light' version is also available for where CPU power is limited, and still sounds very sweet. The manual doesn't make it clear where corners have been cut — some competing models achieve this by reducing the reverb bandwidth, which for most 'real space' simulations is quite acceptable. The IR1 decay time can be adjusted both positively and negatively over a 4:1 range and there are four faders for adjusting room size, reverb density, mode resonance and decorrelation, the latter being a slightly unconventional parameter that provides a useful degree of control over stereo spaciousness. A familiar wet/ dry fader sets the reverb mix and there are separate windows for adjusting the direct sound, early reflections and tail timing and the relative ER/tail level. This works by analysing the impulse response to
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Westlake Audio Studio D.
Outdoors Nekarot Cavern. Luzit Cave (medium and large). Knights Halls (courtyard).
Opera houses Teatro Valli. Sydney Opera House Opera Theatre.
Amptitheatres Siracusa Greek Theater. Taormina Greek Theater.
Scoring stages Todd-AO.
Clubs Birdland. Bluebird Cafe. The Bottom Line. The Kitchen. The Knitting Factory. TLV.
Stairwells
Waves IR1
separate it into early reflections and reverb tail components, while damping is achieved by applying filtering to the high, middle and low-frequency bands of the IR. The overall colour of the reverb sound may also be tweaked using a familiar four-band paragraphic equaliser.
Skyscraper stairwell.
Car interiors Ford Econoline 150. Lincoln Navigator.
Stadiums Gaylord Entertainment Center.
Running some tests confirmed that most of these adjustments sound natural if used over a sensible range, and the decay time can be adjusted quite significantly without giving the game away. However, over-adjustment of the resonance parameter definitely sounds a little odd, though it could be useful as a special effect. The provided impulse responses cover a good range and there are some exceptionally nice ambiences (check out some of the studios) which sound very natural on drums. The ability to fine-tune the settings is very welcome, and is particularly useful with the sampled hardware units where you might want a little more control rather than using the preset as is. Something that seems to be true of many convolution reverb IRs is that because of the way the measurements are taken, the original sound and early reflections are much stronger than with artifical reverbs, the practical outcome of which is that you often have to add way more reverb than you would if you were using synthetic reverb. The main benefit is that you do get a very strong sense of a sound sitting in a real space rather than a dry sound with a layer of reverb stuck on top, and in this respect, Waves are to be commended in providing some excellent impulse responses that capture the magic of a whole range of musically useful halls, rooms and spaces.
Taking It Further Most convolution reverbs sound similar when fed with the same impulse responses — the process is well known and mathematically well documented, even though it is very processor-intensive. What sets IR1 apart from most of its competitors is that it offers a great deal more adjustment once you've loaded the impulse response, and while excessive adjustment of some of the parameters can detract from the naturalness of the sound, the early reflections balance works well as does the control over decay time. Where the decay change involves altering the length of the impulse response, lengthening the decay time uses correspondingly more CPU power, but there's also a clever trick (again with no explanation of how it works), where you can shorten the impulse response and still increase the RT60 without hogging extra CPU power. However it's done, it sounds good. Being realistic, you need one of the faster computers to use convolution reverb effectively without eating up a disproportionate amount of your CPU power (though the 'efficient' version helps in this respect), and if you want to work at 24file:///H|/SOS%2004-05/Waves%20IR1.htm (4 of 6)9/22/2005 6:08:47 PM
Waves IR1
bit, 96kHz, you're going to need a Mac G5 or a fast Pentium 4 computer. You can run convolution on a G4 at more sensible sample rates, but one decentlength vocal reverb can still eat up half your processing power. I also found that I needed to keep my Logic buffer setting at 128 samples or above, or my audio broke up.
In Action I found IR1 easy to use, and providing you don't try to stretch the original reverb too far from its natural state, the adjustments are also very flexible. Having a wide decay time range is particularly good and it sounded natural even with fairly big changes, whereas I found the resonance control had to be used very sparingly to avoid an unnatural sound. The ability to separately control the early reflection level and tail level is also immensely useful. The more ambitious user can also modify the decay envelope shape, but for me, the main attraction of this type of reverb processor is that the results sound right without too much messing around. The included reverb impulse responses are extremely good and cover the whole range, from intimate spaces to caves and cathedrals. They also encompass some very useful hardware reverb presets. You don't get nearly as many as with something like Emagic's Space Designer, but then you have a little more in the way of control to help make up for that. Also, the 'real' studio rooms are absolutely superb for processing drums and other acoustic instruments to give them space and to help them sit in a mix without washing them out in reverb. Until you've heard the effect of a good convolution reverb on a drum kit, you might never realise how good a close-miked kit can be made to sound! Now that Logic users automatically get Space Designer with Logic Pro, they may not feel the need to add IR1 to their collection, but the extra tweakability might still win some of them over. Some Mac owners might even be tempted to move to Logic Pro instead of buying IR1, as they're the same price! Whatever your host software, however, you should look at it very closely, as the ability to manipulate the acoustic environment in such a realistic way can make a world of difference to the quality of your recordings. Published in SOS May 2004
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Waves IR1
Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Q. Are there different types of MP3?
Q. Are there different types of MP3? Published in SOS May 2004 Print article : Close window
Sound Advice
I have downloaded various MP3s from the Internet. I also buy vinyl, which I record into Steinberg Wavelab and convert to MP3. The Wavelab MP3s sound different to the downloaded ones and are generally quieter. Are there different MP3 types and how can they differ sonically? Eddie Howell
Features Editor Sam Inglis replies: The answer to your question is yes. Firstly, the MPEG Layer 3 format (MP3 for short) supports different levels of data compression. Perhaps the most common is 128kbps (kilobits per second), but 48, 56, 64, 96, 192 and more are all in use. The lower the bit-rate, the more extreme the compression, and the more obvious its audible consequences. Secondly, there are two types of MP3 coding — constant and variable bit-rate. In the former, the data compression is applied 'evenly' to an entire audio file, so the compressed version will use the same amount of data to encode each 'frame' of the file. In the latter, the limited amount of data available is used more efficiently. Some parts of an audio file will be more complex than others and will require more data to encode without audible side-effects, so variable bit-rate encoding 'saves up' data from less demanding passages to code these more accurately. As a result, a variable bit-rate MP3 usually sounds better than a constant bit-rate one for a given amount of data reduction. Thirdly, and most fundamentally, different encoders can produce different results. The basic function of an MP3 encoder is to take an audio file and output a data file that conforms to certain requirements. A decoder does the reverse — it takes a data file and 'reconstitutes' it as audio. However, the MP3 format doesn't specify exactly how the encoding should take place, and programmers have developed a number of different encoders, which make different decisions about what parts of the audio to discard when creating an MP3 file. The original 'Fraunhofer codec' is one of the most widely used, but there are numerous others, and you will certainly notice the difference between them even on MP3s coded at the same bit-rate. For more detail, take a look at www.mp3-converter.com/mp3codec/implementation.htm.
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Q. Are there different types of MP3?
Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Q. How should I set my compressor for recording rock guitar?
Q. How should I set my compressor for recording rock guitar? Published in SOS May 2004 Print article : Close window
Sound Advice
Can you give me some compressor tips to make heavy guitars sound more full and even? For example, I want single-note melodies to sound as full as chords or power chords. Also I want to give palm-muted power chords more pump or bite. I'm using the built-in dynamic processors on a Yamaha 03D mixer. Graham Day Editor In Chief Paul White replies: Part of your problem is that the Yamaha 03D compressors are wonderfully transparent, so getting a pumpy sound isn't that easy! I'd recommend you try a fairly high ratio in hard-knee mode (around 8:1) and set the threshold so that the If you're looking for an 'external analogue gain-reduction meter is showing just a dB or two of reduction on the compressor with a reputation for attitude', single notes. Chords should then be brought down to a similar level. you could do worse than the Dbx 566 dualHowever, don't expect a huge improvement in evenness as channel valve compressor. overdriven guitar tends to to reach similar peak levels for chord and single notes anyway — with the waveform peaks all clipped flat, the average level becomes almost the same as the peak level, as with heavy limiting. Any remaining disparity can be smoothed out using your mixer automation. You may be able to coax a bit of pumping from the 03D's compressors by using a longer attack than normal (100 to 250ms) and a short release time (50 to 100ms) but an external analogue compressor with a reputation for attitude may be the best solution. Check out the Dbx range, as their units are often favoured for beefing up rock sounds. Publisher Dave Lockwood adds: If what you really want is for your single-note lines to achieve a similar subjective impression of power as chordal parts, one tried and tested method is to hit the front end of your amp or Pod-U-Like processor a bit harder using a compressor pedal in front of it. Use the minimum squash necessary to achieve a noticeable increase in sustain and then set the pedal's output level to something just above the level of the signal when the pedal is bypassed. The downside is that you will have less touchsensitivity when playing, but there will certainly be more 'pump and bite' in palm-muted chords and low-string riffs, and a quite different subjective effect to applying compression after the distortion stage. Single-note highstring parts can often sound a bit thin when using a lot of distortion, due to the added harmonics, but when there is little or no fundamental in the signal, low-end EQ can't help much. It is room ambience and early
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Q. How should I set my compressor for recording rock guitar?
reflections that tend to give any sound its sense of scale, so using a fairly dark, short ambience program, or adding a room mic if you are using a real speaker, can bring about a dramatic change in the apparent fullness of your single-note lines. Where EQ can help sometimes is in selectively notching out the more aggressive upper mid-range, 'presence' frequencies — start somewhere around 2kHz, cut no more than 3 to 4.5dB, with a bandwidth of only about an octave or so. This will significantly reduce the perceived level, allowing you to then turn up the signal a little to restore the same subjective loudness, thereby allowing a little more of the 'body' of the note to come through. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Q. Should I build a vocal booth?
Q. Should I build a vocal booth? Published in SOS May 2004 Print article : Close window
Sound Advice
I've recently upgraded my recording system. Unfortunately the improvements in the signal chain have highlighted more and more the sound of the room I'm in. The other night I tried singing under a couple of futons with really good results. With a touch of reverb the vocal sits in the mix much better and the fans on the computer are no longer heard. Alas, I can't play the guitar under there, so I was thinking of building a small booth, say 3 x 5 x 7 feet, and hanging the futons on the walls to create a deader environment in which to track. Some degree of soundproofing is a bonus, but my main aim is a more neutral acoustic to take and treat with effects. Will I just be swapping one duff sound for another? SOS Forum Post Technical Editor Hugh Robjohns replies: I'd advise against building another small boxy room, and instead, sort out your existing room to provide a better acoustic for recording and monitoring. Building a good-sounding booth is far from trivial, especially when the booth will have fairly small dimensions, and you are more likely to end up with a worse-sounding recording space than you started with! If you feel the need to do some woodwork, I'd recommend making a soundreducing box to put the computer in to help control the fan and drive noise. You will then be free to apply suitable acoustic treatment to the room as a whole, to make it sound much better, both for recording and monitoring. This approach would be far more cost-effective overall, and a lot more pleasant to work in! For the computer box, you'll need to design something that allows access to the CD tray, power switch and so on — perhaps through a front door — and has an orifice through which the various cables can exit. You'll also have to ensure that a sufficient volume of cooling air can flow in and out, but by designing the case so that air flows around a labyrinth which is well damped acoustically, you should be able to reduce the computer noise substantially. As for the room, the ideal is to treat it so that reverberation is well controlled and even at all frequencies, and standing waves are minimised. This will require a
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Placing a piece of hardboard
Q. Should I build a vocal booth?
combination of bass trapping, absorption and possibly even some diffusers — check out the web sites of the usual suppliers, as well as archived articles on the SOS web site for ideas. Most suppliers offer a free planning service too. Beware, though, that inappropriate or excessive foam treatment on its own will soak up the high and mid-frequencies, but will do little to control the lower frequencies, resulting in a boxy-sounding space.
on the floor between the guitar and microphone can improve the sound of your guitar recordings.
A handy, quick and cheap solution for recording vocals is to temporarily hang duvets (or futons) close to a rear and side wall, to form a well damped corner. Stand with your back to the corner, a couple of feet out, and place your (cardioid) vocal microphone so that it is 'looking' into the corner. That way, the dead side of the mic is facing the room and will therefore tend to ignore most of the reflections, room ambience and hopefully even the computer noise. The duvets or futons in the corner will prevent the room sound from being reflected back into the front of the mic, and the result should be a reasonably good-quality, dry vocal track. You can try the same basic technique for recording your acoustic guitar, although in this case it is often useful to deliberately encourage some early reflections by placing a sheet of hardboard (or similar) on the floor between mic and guitar. Arrange the duvets or futons in a semicircle behind you to prevent room reflections from heading back into the front of the mic again. These kinds of problems and solutions tend to feature heavily in our Studio SOS series, so keep an eye out for ideas that you might be able to apply to your own situation. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Q. What can I do to improve my stereo recording setup?
Q. What can I do to improve my stereo recording setup? Published in SOS May 2004 Print article : Close window
Sound Advice
The Microtech Gefell M930's ORTF
I record brass bands regularly using a stereo mic mounting bar allows precise mic placement. setup (X/Y or spaced pair). The people I record for are always happy with the results, but I feel I can do better. The sound still doesn't come close enough to a commercial brass band recording. My current setup consists of two AKG C1000s mics, a Behringer 1804 mixer and compressor, and a Sony Minidisc. I know that some parts of my setup are not state-of-the-art, but I'm sure the results can still be better. Which mic placement should give the best results? How can I find out if I'm suffering from phase problems? I only use very subtle compression to cut off some peaks. At normal levels the compressor doesn't have to work at all. What could I do to make the sound more bright? Would better preamps contribute to a bettersounding record? Ief Sels Technical Editor Hugh Robjohns replies: The first thing I'd say is that the C1000s doesn't have a particularly good high-frequency response. Brass instruments have a very strong harmonic content which is critical to their sound, so I fear that your mics are not ideal for the task. The other thing to mention straightaway is that the Minidisc format uses a data reduction system which discards a lot of subtle information, and this will affect the perceived quality of your recordings as well. In terms of mic placement, coincident and spaced mic arrays produce different kinds of stereo imaging. Choose the configuration that you feel sounds best in each location. Personally, given a pair of cardioid mics I'd probably start with an ORTF arrangement (named after the French broadcasting network, Office de Radiodiffusion Television Francaise), which largely combines the best of both coincident and spaced configurations. Angle the mics outwards at 110 degrees to each other (in other words, 55 degrees each side of the centre axis), and space apart by about 17cm (see diagram). It depends on the room to a degree, but I find this arrangement usually gives good imaging with a nice sense of spaciousness. You'll always risk encountering phase problems using spaced microphones, since the two mics capture sound from a given source at different times, and hence with different phases. However, I've never had a problem of this kind with the ORTF configuration, although whether or not you'll suffer phase problems in a specific venue file:///H|/SOS%2004-05/Q.%20What%20can%20I%20do%20to%20improve%20my%20stereo%20recording%20setup.htm (1 of 2)9/22/2005 6:09:21 PM
Q. What can I do to improve my stereo recording setup?
is hard to predict. To find out, simply listen in mono. If the mono sound is noticeably coloured compared to the stereo sound then you have phase problems and the only practical solution is to revert to a coincident (X/Y) mic arrangement. Coincident mics don't suffer phase problems at all since the two mics capture sounds from any direction at exactly the same time (and hence phase). Finally, I wouldn't record with a compressor at all. Most compressors reduce brightness of the material when driven and that's the last thing you need here. It sounds like you are only using the compressor as a security blanket anyway, so instead, why not simply leave a few more decibels of headroom in your recording. That way you won't have to worry about peak overloads while recording, and you can adjust the overall dynamic range and levels as a post-production treatment, when you know exactly what you have to play with, and can make such critical decisions without committing them forever to the original recording. To answer your last question, yes, better preamps will contribute to a better-sounding recording, but it's a case of degree. Using more appropriate mics and recording with an uncompressed format will probably have a more significant effect on the quality of your recordings and I would suggest that that should be your first approach. Almost any of the small-diaphragm condenser mics on the market would probably give you better results. I'd suggest the Rode NT5s or Sennheiser e664 (superb value for money but soon to be discontinued), or maybe the SE Electronic SE1s if your budget is tight. After that, maybe think about changing the recorder for something with better converters, decent preamps and a linear recording format. The new Fostex FR2 would be an excellent (if still relatively expensive) choice for a hardware recorder, or you could think about using a USB or Firewire interface box and recording directly to a laptop computer. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Q. What should I look for in a PC laptop?
Q. What should I look for in a PC laptop? Published in SOS May 2004 Print article : Close window
Sound Advice
I'm considering moving over from my current Mac setup to a PC laptop for mobile recording, using Nuendo 2 and either an RME Cardbus interface or one of MOTU's Firewire systems. However, with so many different manufacturers and models, I don't really know where to start. Can you point me in the right direction? Is a Centrino laptop the only sensible choice for this kind of application? Diane Reeves PC music specialist Martin Walker replies: When choosing a PC laptop for music it's even more important to get the most suitable model for your requirements than when buying a desktop machine. After all, it's not only likely to be a hefty one-off investment, but also incapable of having any of its chips changed over if you come across an incompatibility with your choice of audio hardware. Audio recording hardware is often picky about which PCs it works with, whether desktop or PC, so make sure you can get a refund if you run into problems with your proposed laptop. RME's Cardbus interface, for instance, is known to be picky about certain Cardbus controller chips, and similar cautions apply to MOTU gear and any piece of high-performance audio hardware. If you want to buy a specific mainstream model such as one from IBM or Sony then the safest way to proceed is to choose your audio interface, and check with its manufacturer that the appropriate (Cardbus, Firewire, or USB 2.0) laptop controller chip is compatible with it, depending on whether it requires a PCMCIA, Firewire, or USB connection. However, this assumes that you can get hold of details of the chips in question, and while Centrino laptops do all use the same Intel chip for USB 2.0 support, Firewire and Cardbus chips can vary from laptop model to model, as can USB controller chips on non-Centrino models. A Centrino laptop, like this IBM Thinkpad, will Chip details rarely appear in the spec list, but a helpful dealer or a have a long battery life and produce less fan fellow musician who has already bought the same model (such noise than a standard Pentium 4 machine. information is often asked for on the SOS forums) might take a peep inside Device Manager to find out the chip makes and numbers for you. However, the safest way to proceed is to buy your laptop from a specialist music retailer who has already
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Q. What should I look for in a PC laptop?
checked out these aspects for you, and can guarantee that it will work with a wide variety of music hardware and software. As for the specification, this depends on what sort of mobile recording you want to do with your PC laptop. If you want the most powerful solution, a desktop replacement model featuring a standard Pentium 4 processor may prove most suitable, since these offer clock speeds above 3GHz, as well as screen options up to 17-inch. However, such models will almost certainly require mains power, except for short emergency sessions of an hour or less, and will soon get hot and require fan cooling, which may be noisier than you'd like if you want to use microphones a short distance away. Given your two audio interface suggestions I suspect you're intending to record four or more audio tracks simultaneously, and probably require mains power anyway for the interface and mic preamps, so don't rule out this approach. However, if on the other hand you want to venture into the wilds on battery power with one or two mics then the only real choice is a Centrino, since this will generally provide double the battery life compared with most desktop replacement models (typically two to three hours, depending on how much processor power you're consuming), not get hot, and therefore be almost silent, and generally be half the weight (handy if you're lugging it up a hill in a backpack). The choice is up to you, but a 1.4GHz Centrino will provide similar processor clout to a 2.1GHz standard desktop PC, which should be sufficient for the majority of mobile recording duties. This presupposes that you're recording live music, but if you ever intend to use your laptop on stage playing back songs with plenty of soft synths and existing audio tracks then go for the fastest processor you can afford in each range. A sensible amount of RAM to partner either laptop is 512MB (unless you intend to run loads of soft samplers, in which case 1GB might be useful), while a typical 40GB hard drive will probably be sufficient for most purposes. However, if you think you may eventually benefit from a larger drive, it's far more convenient to go for a larger internal drive at the time you buy your laptop than to buy an external USB 2.0 or Firewire drive later on — when recording on the move, the less you have to carry about the better. This highlights another advantage of buying from a specialist music retailer: you can specify the processor speed, amount of RAM, hard drive capacity, and so on at the time of purchase, rather than opt for the (mostly) fixed specification models offered by mainstream outlets. Remember also that if you have problems running Nuendo or a MOTU/RME interface on a mainstream laptop, and you ask for advice on the laptop manufacturer's tech support line, you're likely to get the reply "Never heard of them". I rest my case. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Q. Where should I put my monitors?
Q. Where should I put my monitors? Published in SOS May 2004 Print article : Close window
Sound Advice
How much distance should there be between my monitors and should they face straight forward or be angled toward the listener? Also, as my monitors will be placed against a wall, should some acoustic foam be placed directly behind or between them? SOS Forum Post Technical Editor Hugh Robjohns replies: You ideally want yourself and your two speakers to sit at the corners of an equilateral triangle, with your head the same distance from each speaker as the distance they are spaced apart (see diagram) — something between 1 and 2.5 metres (3-8 feet) should be about right, depending on the size of the room and its acoustics. The speakers should ideally be mounted symmetrically in the room, and on rigid stands tall enough to place the tweeters roughly level with your ears. Most monitors are designed to be used in a particular orientation (usually with the tweeter above the woofer), so make sure you place them the right way up. Turning speakers upside-down or on their sides can have disastrous effects on the stereo imaging and frequency response! Most speakers are designed to be used well away from both side and rear walls, but if you have a choice, it's usually better to put the speakers closer to a back wall, rather than side walls. If placement near walls is unavoidable try to use a speaker with appropriate compensation built in. Most active speakers include some switchable provision for low-frequency correction (such as a specific LF roll-off curve) to suit placement near a back wall or — horror of horrors — in a corner! If you are using passive speakers with a reflex port, you may be able to gain some useful improvement in the low-frequency balance by plugging the port with acoustic foam, although the results are unpredictable and could make the overall sound worse rather than better. If the speakers have rear-firing ports, you'll have to leave a reasonable air gap behind them anyway. Read the manufacturer's handbook for specific advice. You also need to think about where the speakers are pointing. Although a lot of designs (Genelec monitors, for example) are intended to be aimed directly at the listener — often referred to as a degree of 'toe-in' — many others are designed to face directly forwards into the room so that the listener is effectively placed slightly off-axis to each speaker. Most PMC monitors are designed to be used this way, for example, and if pointed directly at the listener they will sound slightly brighter than intended. This aspect of loudspeaker placement is influenced by the designer's approach to the speaker's file:///H|/SOS%2004-05/Q.%20Where%20should%20I%20put%20my%20monitors.htm (1 of 2)9/22/2005 6:09:35 PM
Q. Where should I put my monitors?
horizontal polar pattern or dispersion, and the speaker's off-axis frequency response. Bear in mind that as well as the direct sound reaching your ears, the speaker puts a lot of energy into the room, and that all comes from the speaker's off-axis response. The degree of toe-in or toe-out can also have a significant effect on the accuracy of the stereo imaging and the stability of the central image, so it's worth experimenting with small changes of angle to try to optimise the precision of the imaging and the width and stability of the listening 'sweet spot'. Beware of early reflections from the walls to either side of the monitors too (and possibly from the ceiling if it's fairly low), as these can also mess up stereo imaging, particularly at mid- and high-frequencies. If you can't avoid placing your monitors near side walls, consider putting acoustic foam tiles or even the ubiquitous duvets on the side walls between the speaker and listening position to absorb those reflections. You can determine the best place for the absorbers by getting someone to move a mirror around on the wall (or ceiling!) While you sit in the listening position. When you can see the speaker's tweeter in the mirror, that's the place to centre the acoustic absorber. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
file:///H|/SOS%2004-05/Q.%20Where%20should%20I%20put%20my%20monitors.htm (2 of 2)9/22/2005 6:09:35 PM
Business End
In this article:
Business End
Ubiquity Readers' Recordings Ed Dowthwaite Published in SOS May 2004 Amalthea Print article : Close window This People : Miscellaneous Month's MPG Panel
Business End enables you to have your demo reviewed by a panel of producers, songwriters, musicians and managers. If you want your demo to be heard by them, please mark it 'Business End'. This month's industry panel is drawn from the MPG (Music Producer's Guild).
Ubiquity Sam Stubbings (SS): "I think they're really good — a really good pop-rock band. I think they could write a huge hit actually. I'm not sure there are any on this demo but it seems like the potential is there.
Track 1 2.1Mb
"Track three is easily the best and that's when I started to think that they were going to write a big hit. The production's there and the ability's there, the guitar sound is superb. The kit list they've included doesn't even look that advanced considering the sound they've achieved. I think that, for a record that's not on sale in the shops, the production is excellent . "The singer's got a really good voice but I think she should maybe stay away from the higher notes. Her vocals are strongest in the choruses of some of the more rocky tracks where she really lets go." Barry Sage (BS): "All of these tracks are very good and I really don't have anything to say about how they could do it better technically because I think they're doing it amazingly well already. "These are all quite poppy songs but at the same time I think that's intentional and not a compromise. The tracks have definitely got a certain integrity about them. "The only problem with this is that although the songs all sound like they would make great album tracks I'm not sure that there's a single there. With some good singles and the right management I think they could do really well." Nikolaj Bloch (NB): "If you're trying to make commercial-sounding music like this, you need to have hit songs, and I don't think there are any on this demo. I think these tracks have a really nice sound to them but that's all there is to it. They need something to catch your attention. I liked the verses of the songs but as soon as you get to the chorus where you expect to get this big payback, it doesn't happen. I do like the sound of it though and the way it's been recorded, it's just the songwriting which they need to work on, if they can get that sorted out then there's no reason why they shouldn't be successful." Andy Rogers (AR): "I don't disagree that they could write a big hit, track three is very strong and quite catchy for example but I don't think the voice is all that good. The production isn't bad but it sounds a bit woolly in places.
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Business End
"I think it has quite a European sound and I could see it being popular there, I'm not sure it would be so successful in the UK or America though. I agree with what Barry says about these all being good album tracks but they're going to need singles to get an album deal."
Ed Dowthwaite BS: "I like this. I think maybe he could do with a producer though, someone who understands the genre of music he's going for and can help him to get it sounding right.
Track 1 2.8Mb
"I think with this sort of singer-songwriter style you need a bit more clarity, the instruments don't seem that true to themselves at the moment. He mentions in his letter that he's used an open tuning on the guitar to give it more of a sense of space but the tracks don't really have that feel at all. The reverb doesn't help, it's just clouding everything at the moment. I like the songs though, it just needs to be mixed a little more delicately to make it feel less claustrophobic." NB: "This is more like the sort of music I'd listen to myself, but at the same time I'm not sure that I really like it. I like the style of it and it sounds OK technically but somehow I don't think he's getting his point across very well. "I think the singing is a bit weak and, I found it quite hard to focus on his voice. The mix is very cluttered, the vocals and the guitar seem to be getting in each others way a lot and that makes it hard to listen to. "He's written a very nice letter though, it's very sweet and it makes me want to like his music." Karen Murphy (KM): "This feels very dreamy — the way it sort of takes off and takes you on a little journey while it modulates and flows. I think it could work better if it had some stronger lyrics though. At the moment this seems a bit wishy-washy and it doesn't really go anywhere. It needs a stronger melody to give it some more focus. The instrumentation is OK, but again it's not particularly inspired and it doesn't feel like it's got that much direction." SS: "I think this sounds like someone who enjoys playing and recording their own music, and from that point of view it's great but it's certainly not something you could release. "I really admire the fact that this guy is over 45 and is still writing music and sending CDs out. I think the songs are quite nice but there's nothing too original about this and it's terribly recorded. I don't even know how Karen can tell that the lyrics are bad because I can't even hear them. The guitar just completely dominates the mix and the vocals aren't nearly strong enough to compete with it. There's far too much reverb, it sounds really murky and dark." AR: "I think this is absolutely terrible. I'm sorry but I think you have to be cruel to be kind in this sort of situation. I think that a lot of people write music at home and ask their friends what they think — and they're never going to be completely honest with you, so maybe we ought to be. "This isn't really original in any way — all the lyrics are about love and that sort of thing, it's all been done before. I also think this is very badly recorded — the acoustic guitar sounds quite nice but the vocals are set too far back and are swamped with reverb. Maybe that's because he feels unsure about his voice. Reverb can mask a lot of things but the vocals still sound out of tune to me and there's no hiding that. "I think the whole thing just sounds rather dated and I can't see how he's going to get anywhere with this."
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Business End
Amalthea KM: "I think this starts well, I like the instrumental track at the start, it almost reminds me of Radiohead, with the slightly discordant part at the end. I like the second track too but it feels like they've got a bit lazy after that; the last three tracks seem more like a jam session than anything else.
Track 1 2.5Mb
"The biggest problem I have with this band is the singer's voice. The music sounds like it's trying to be meaty and hard and when the vocal comes in it seems really weak and thin. Maybe it's just the contrast with the music that makes it sound like that, but it does seem like there's a bit of a mismatch between the music and vocals. Her voice doesn't have any strength to it, I think she should have a much more forceful, growly delivery. It seems too timid and I think that really lets it down." NB: "I think Karen's being a bit harsh on the singer. I don't think her voice is that bad, it just doesn't go very well with the music. With a different band she could be quite a good singer. I noticed that all of the tracks seem to be in the same key. Maybe that's because the vocalist is comfortable with it though, maybe they are. I think they could do with a producer, somebody who could rein them in a bit and make it a bit less indulgent." AR: "This is all right, it's very well recorded and they've obviously spent a lot of time on it. They sound very well rehearsed as well, they say in their letter that they do a lot of live gigs and you can tell — they're very tight musically. None of the songs particularly stuck out though, none of them really grabbed me. I also think it seems a bit dated, the vocal reminded me of the singer from All About Eve. "The worst thing about this demo is the letter they sent with it. There's nothing worse than a new band telling you that they're the greatest thing in the world because you're instantly going to think that they're not. There's just no way that you can live up to that sort of statement." BS: "The letter is not a good idea. As much I'd like to see Pop Idol and Simon Cowell and the like disappear from this planet, I don't think Amalthea are going to be the ones to do it. I can't really see them being the phoenix rising from the ashes of pop music. "There are five tracks on this CD, I really like the classical part at the start, it gets your attention and makes you wonder what you're about to be led into, and then the first song after that is great. The rest of the tracks just sound like them jamming though, I think that would be good in a live gig but it's a bit much for three songs one after the other on a CD." SS: "I think this band, especially the drummer, are fantastic musicians, easily the best we've heard so far tonight. I think they need to concentrate more on the composition and structure of the songs though. They seem to be doing it the wrong way round at the moment. I've been in a lot of jams with bands where you're all playing along together and it all sounds great to you, and it might sound good live but it doesn't really work on record. If they want a fivealbum deal then they're going to need at least one song. They should sit down with a guitar or a piano and write a melody first, they should write the song and then let all these great musicians take it and develop it from there. "In their letter they're saying that pop music is crap and they're the antidote. Why are they sending that to record companies? Who do they think the music industry is? It's record companies! Do they think that A&R people are all struggling musical activists or something?"
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Business End
This Month's MPG Panel
Barry Sage is a freelance producer and engineer. He specialises in Latin music and is well established in Spain and South America for his work with pop acts La Oreja de Van Gogh and Melon Diesel. As an engineer he has worked with a wide variety of artists, including New Order and the Rolling Stones. Recently he has been involved in the creation of a sample library of Cuban percussion (Beats Working — In Cuba) for Zero-G and Native Instruments. A full list of Barry's work can be seen on his web site at www.barrysage.co.uk. Sam Stubbings is the Senior Producer for the DVD division of Metropolis. He began his career five years ago at Abbey Road and has since worked with artists ranging from Paul McCartney to Muse. More recently he has produced both the first DVD single (Bjork's 'All Is Full Of Love') and the first commercial DVD-Audio disc (Holst's The Planets). He also has his own act, Redstar, who are currently recording an album and gigging in London. Nikolaj Bloch is a freelance engineer, writer and programmer with almost 20 years' experience in the music business. As the guitarist in the band Subcircus, he played all over the world until their split in 2000. Since then he has worked as a programmer and soloist on several major Hollywood films. He has also written for a range of artists varied enough to include American country singers and Jimmy Somerville. He enjoys spending time writing and collaborating in Nashville throughout the year. Andy Rogers is a producer of production music for BMG/ Zomba. After graduating with a degree in music from Kingston University he spent a year as an engineer in a Hammersmith studio. Following a six-month stint at Abbey Road he joined BMG as an office junior. A year later he was producing albums of production music for TV, film and radio. Having left BMG to join Zomba, he was reunited with his former colleagues two years later when the companies merged. Karen Murphy began her musical career by training to become an opera singer. She later worked as a professional rock, pop and jazz singer in Australia, Japan and the UK, performing with original bands, cover bands, and in commercials. Since moving to the UK she has been employed as a Post Production Co-ordinator at Abbey Road Studios and currently works as a Project Co-ordinator for film and TV specialists Videosonics. Many thanks to The Firebird Suite who hosted the session. Their web site is at www.thefirebirdsuite.com. The MPG's web site is at www.mpg.org.uk. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Educate Your Ears
Educate Your Ears Leader Published in SOS May 2004 Print article : Close window
People : Industry/Music Biz
One of the most frequent types of question we get here at the SOS office concerns the choice of equipment, and it's evident that a great many people don't have enough experience of the available products to judge what is good and what is less so. Perhaps more seriously, they don't have the experience to know what's good enough to meet their needs. Obviously it's part of our job to check out equipment and give our honest opinions, but interpreting them can still be difficult if you haven't already listened to a wide range of alternatives in that same area. Synthesizers are probably the easiest instruments to evaluate, because you can take a trip to your local music store, cruise a few presets and then compare the sounds with other machines they have in stock. However, it's much more difficult in the case of, for example, microphones, where you need quiet surroundings, reasonable acoustics and plenty of time to make any kind of meaningful comparison. If you've only ever used a dynamic mic before, you will probably hear a noticeable difference whatever capacitor mic you try, but comparing one capacitor mic to another is less easy in a shop environment. And if you don't have the chance to try out quality gear at all, you may not even realise what you are missing — you might decide that your dynamic mic is fine as it is, simply because you've never heard anything better. Another area where it pays to be aware of how equipment sounds at different ends of the price range is reverb. Hugh Robjohns and I have done quite a few Studio SOS visits now, and while many of the people we've met seem happy to spend a lot of money on computers, monitors and instruments, they usually end up mixing using a host-powered reverb plug-in or a budget hardware box. When you're in a position to hear all the different reverbs available, you soon realise that the native reverbs that come bundled with the majority of sequencers simply don't cut it. Cheap reverbs sound gritty, they clog up your mix and they don't create a natural sense of space. Compare a cheap plug-in with even a mid-priced hardware reverb or a DSP-assisted reverb running on a plug-in card, and you'll hear a huge difference. And now that we have a new generation of convolution-based reverbs, both as hardware and plug-ins, we are experiencing another quantum leap in quality. However, if you never hear 'the good stuff', you're less likely to be aware of the shortcomings of your own equipment or software, just as you never realise how bad your eyesight is until you put on your new spectacles! You don't need to buy the best of everything to make good-sounding records, but if you can identify the areas where you need to lift your game, and by how much, it will help enormously. You may also discover that in some areas, once you get past a certain price point, you have to pay a lot more money just to get a small increase in quality, so a mid-price solution could make more sense than buying the very best.
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Educate Your Ears
The reason I'm banging on about this is that I've spoken to a few colleges and music stores, and suggested they host the musical equivalent of a wine-tasting evening, where you can go and listen to the difference between a dynamic mic, some budget Chinese capacitor models, a few mid-priced mics and perhaps a highend Neumann. The same goes for reverb and EQ. If this seems like a good idea, then why not suggest it to your local music shop? Paul White Editor In Chief Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Emulating Brian May's Guitar Amp Sound
In this article:
Carry On Amping Games For May From '30 to Thirty
Emulating Brian May's Guitar Amp Sound TC Electronic's Stefan Moeller Published in SOS May 2004 Print article : Close window
People : Artists/Engineers/Producers/Programmers
For many years, German designer Stefan Moeller has been on a quest to perfectly emulate Brian May's guitar amp sound. Now, with TC's new TC Thirty plugin for Powercore, he believes he's cracked it... Paul White
Amazingly, Powercore, TC Electronic's standalone DSP platform for processing and instrument plug-ins, is already nearly three years old. During that time, the plug-ins available for it have expanded from the original TC Works effects software to include voice-modelling instruments from TC-Helicon. The last year has seen third-party companies joining in, with Sony porting their Oxford TDM plug-ins to Powercore, and now synth company Access have also got involved, announcing a software version of their Virus analogue-modelling synth for the platform, Virus Powercore.
Carry On Amping Recently, TC took Powercore in a new direction when they announced two new amp-modelling plug-ins for the system. The entry-level Tubifex (originally announced back in December last year in the news pages of SOS as Voltronics prior to a name change) is to be bundled with all Powercore systems from now on, and runs in a hybrid mode utilising both host-powered Native power and the Powercore's DSP chips; apparently the amp modelling runs native and the speaker emulation runs on Powercore. Rather than copying specific amplifiers, this model comprises three virtual 12AX7 tube stages, complete with variable
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Emulating Brian May's Guitar Amp Sound
EQ, bias, HT voltage and distortion symmetry. It also includes speaker emulation and comes with a range of presets. The other amp-modelling plug-in, TC Thirty, is a software emulation of Vox's timehonoured AC30 amplifier, and was developed in conjunction with Stefan Moeller. In contrast with Tubifex, this model is very specific in modelling the sound of the Vox AC30 as used by Brian May and includes a treble-boost emulation based on Brian May's own setup. Stefan, a self-confessed Brian May obsessive, has worked for over 15 years on this project; he started with analogue emulations, and then learned programming to that he could port his work into software. As finished in the form of TC Thirty, the emulation runs entirely on Powercore. With any guitar-amp model, latency is a concern and Powercore has a processing latency that depends on the I/O buffers set for the soundcard, though using buffer sizes of 128 or even 256, you don't notice the latency. However, you can activate the No Latency mode inside the plug-in to remove Powercore's latency altogether while playing (at the cost of using more CPU power) and then turn this function off again after you have made the recording.
Games For May I spoke to Stefan Moeller about his fascination with the Brian May guitar sound and he explained the background to the project. "I have studied electronics for more than 20 years, and have always played guitar, but I was particularly interested in the playing and guitar sound of Queen's Brian May. I tried lots of different amps and guitars, but after reading interviews with Brian, I learned that he used the Vox AC30 along with a guitar that his father built for him. So I bought myself an old 1961 AC30 and built myself a guitar copied from Brian's own, fitted with the original Burns pickups. I used the Brilliant channel and the sound, while quite good, was not quite right, which is when I realised he also used a treble booster before the amp. "My next step was to try to simulate the exact sound of the vox AC30 using analogue, solid-state circuitry based around op-amps, diodes, FETs and so on. I finally settled on using op-amps and diodes, as that gave me the most design flexibility. During this work, I realised that the Vox is a very unusual amplifier that almost sings! It's both smooth and wild at the same time. There's a lot of information on the Internet about this amplifier where most people will tell you that it's a Class-A amplifier, but it's not really — it's actually a Class-AB amplifier. I tried to measure the behavour of every stage of the amplifier, and then copied it tube for tube, filter for filter, using my circuitry. It worked well and when I played through it, I could feel the same transient response as the real AC30, which is very important. You can feel the 'sagging' effect when you play hard and the noise drops away, then fades back in after a short delay. This turned out to occur naturally in my analogue version after copying each stage carefully, but the work was quite demanding and took me around two years. file:///H|/SOS%2004-05/Emulating%20Brian%20May%27s%20Guitar%20Amp%20Sound.htm (2 of 4)9/22/2005 6:10:22 PM
Emulating Brian May's Guitar Amp Sound
"A student who turned out to be good with DSP algorithms helped me to program a software version of the amp using assembler language, and he's now working for TC, so he introduced me to the company. I'd originally done some work modelling the circuitry in the C language, and that was much easier than DSP code. It was also difficult to get rid of the quantisation Part of Stefan's test rig, featuring a vintage artifacts that occur in non-linear Vox AC30 amp with his analogue treblesystems such as distorted amplifiers. boost emulation on top. TC were impressed with my work, and so we decided that they would develop my algorithms for use on the Powercore platform."
From '30 to Thirty I was curious as to how accurately Stefan had modelled the real amplifier, as it clearly isn't practical to model every single component. "I looked at the input stage and modelled that as one block. Then I modelled the phase-splitter stage, which is based around two tubes, and modelled that as one further stage. Then I modelled the power amplifier and output transformer as a further block, so this is a three-stage simulation. There's filtering between each stage: a low-pass filter before each stage and high-pass filtering afterwards. The splitter stage has some 'secrets' inside, such as the grid simulation where the bias offsets slightly under conditions of heavy distortion. The splitter stage was a complicated thing to model! "The loudspeaker was modelled as a separate stage from the power amplifier. When you load the amplifier with a resistor, it works differently to a speaker which has a complex impedance. So I measured the loudspeaker and then modelled its characteristic so that it interacted correctly with my modelled transformer's characteristics. "Brian May did a demonstration on the Internet where he used only the treble booster and the amp with no effects. I don't know how he miked it, but I listened to it and it was a very typical Brian May sound, so I tried different microphone settings with my original AC30 setup and a replica of Brian's treble boost circuit, and experimented with EQ settings to get much closer to this sound. That took me a further two weeks, then when I was happy that the sound was right, I used that combined speaker and mic response for my model. "The completed plug-in includes an emulation of Brian May's treble booster, but users have a choice of four resonant frequency values: 4kHz, 2kHz, 1kHz or 500Hz, so that people who are using different guitars can still get close to the file:///H|/SOS%2004-05/Emulating%20Brian%20May%27s%20Guitar%20Amp%20Sound.htm (3 of 4)9/22/2005 6:10:22 PM
Emulating Brian May's Guitar Amp Sound
sound. And of course it means the user can get more sounds — not just Brian May's sound. I've also included three gain settings to enable the user to simulate by how much the treble booster overdrives the amplifier. Brian uses his guitar's volume control to move from a rhythm to a solo sound and the amplifier just responds to it. The model behaves the same way, so at low volumes it's clean and crisp, whereas at high levels it's like a wild animal! "So far I have only modelled the Normal channel of the amplifier, which has no tremolo or top boost, though top boost will probably be added to the model during a future revision. I didn't model the Vox's Accutronix spring reverb because I didn't think it was particularly good, and in any event, that wasn't on the original models. I've also left off the Vox amplifier's one tone control, as this was simply a top-cut control and made the amplifier sound rather dull. My model still includes the effect of this control, but it is fixed in its brightest position. The programming took a lot of time to get right, and I've learnt a lot about DSP programming in the process. As the TC engineers developed the code, I did listening tests at every stage. Now I think the result from the miked-up AC30 we have set up here sounds virtually identical to the sound of the digital model in an A/B listening test." Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Graham Fellows (aka John Shuttleworth)
In this article:
Looking At Sound The Organ Master Playing With Time Tricking The BBC Heavy Weather In Wardrobe The Very Best
Graham Fellows (aka John Shuttleworth) Making Engineering Funny Published in SOS May 2004 Print article : Close window
People : Artists/Engineers/Producers/Programmers
Comic actor Graham Fellows, better known as John Shuttleworth, has been recording and editing his own radio series for years — and now he's putting his engineering skills centre stage. Sam Inglis
The days when an engineer's life involved hours spent cutting up and reassembling quarter-inch tapes are, for better or worse, behind us. Some will look back with fond memories, others will be glad to see the back of a fiddly and repetitive task, but there's probably only one person who could turn tape splicing into a comedy routine.
Photos: Richard Ecclestone
Whether or not the name Graham Fellows is familiar, the chances are you've encountered some of his alter egos. As Jilted John, he had a top five hit in 1978 with an eponymous Martin Hannett-produced single, while his middle-aged wannabe pop star John Shuttleworth has enjoyed a more enduring career. A sell-out show at 1992's Edinburgh Fringe spawned numerous radio and TV series, national tours, videos and CDs, and John has been joined through the years by a growing cast of Fellows-voiced ancillary characters such as manager Ken and wife Mary. But if the humour in The Shuttleworths arises from the gulf between John's musicbusiness ambitions and his mundane reality, Fellows's more recent creation will strike SOS readers even closer to home.
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Graham Fellows (aka John Shuttleworth)
The character of Brian Appleton is closely based on the numerous sound engineers that Fellows has encountered over the years. "He started off as a rock historian and multimedia lecturer, who's been dumped upon from a very great height, as he likes to say, by certain people in the rock industry," explains Fellows. "The first show, Brian Appleton's History Of Rock & Roll, which was also a series on Radio Four, was really just a series of shaggy dog stories where Brian crowbars himself into the true history of rock. I read a lot of rock history — Brian's live CD from the Let's like Howard Jones used to work in a cling-film Look At Sound tour, featuring 'How to edit audio tape' — see factory, and Rod Stewart was a gravedigger — www.brianappleton.co.uk for more things like that I found quite amusing. But that, details. by its very nature was a limited idea, and there was only really one show's worth there. So for the second show I did, Let's Look At Sound, I decided to make him explore sound, doing an extension of his college lecture on recording technique." And that, by way of some character research, would eventually lead to the idea of taking a Revox B77 ("a workhorse of a machine," as Brian describes it) on stage. "'How to edit audio tape' is a sketch that was totally inspired by one of the sound engineers that I interviewed," says Fellows. "I interviewed about three sound engineers, just to get the way they talk about sound. Invariably, they'd try to come up with funny anecdotes, and I didn't really want that, but then this guy called Keith Brakes said that when he was about 14, he used to go to a Tape Club. I don't think they exist any more, but he used to go and make sound effects, and one of the exercises they used to do to improve their editing skills on splicing tape was to record 'God save the Queen' into a mic and then edit it as 'Sod the Queen'. As soon as he said that, I thought 'That's going to work really well on stage.' "To do it in front of a live audience, and very quickly as well, I practised until I knew exactly where to cut and didn't have to try too hard. But for an audience to see me actually record someone's voice, cut up the tape, stick it together again and then hear the result — and they knew I'd done it because they recognised the voice — was a bit magical. When it worked..." The magic of sound is complemented by an obvious delight in technical jargon, as Brian warns his uncomprehending audience about the perils of 'bass tip-up' and dispenses hints on editing technique ("You'll notice I've got a built-in splicer to cut the tape. You can use a razor blade — I prefer not to, because I suffer from depression."). "It was a way of discovering a little bit more about the theory of sound myself. I started to swot up on the way frequencies behave, and I was able then to start writing gags about it. The opening gag in the show was about how sound travels a lot slower than light, and you might therefore think that sound is inferior to light, but actually it's superior, because the fact that sound travels slowly allows us to pinpoint the position of things. If sound travelled at the file:///H|/SOS%2004-05/Graham%20Fellows%20%28aka%20John%20Shuttleworth%29.htm (2 of 7)9/22/2005 6:10:26 PM
Graham Fellows (aka John Shuttleworth)
speed of light, we might as well have one ear. "One of the high points was a recording of a nose-blow, which Brian didn't even know he was recording — he was just clearing his sinuses in preparation for a heavy rock vocal — and then accidentally he hits the halfspeed button, and he plays it back and The Yamaha Portasound PSS680 and hears his sneeze slowing down and PSS51 keyboards, so fundamental to the decides it sounds rather attractive. So John Shuttleworth sound. he starts slowing it down and bouncing it backwards and forwards between two Revoxes. Which I did for real at home."
The Organ Master "Perhaps Yamaha's finest moment was the Portasound PSS680," insists Graham Fellows. "It's got so many permutations, and 100 styles. They really pushed the boat out. Unfortunately it's let down by a high-pitched electronic whine. I tend to use the PSS51 more live. I've got two of each." Fellows's best-known character, John Shuttleworth, has remained loyal to the Yamaha brand over the years, favouring the instruments towards the budget end of their home-keyboard range. "I have listened to all of them, and I still check the new models out regularly. The new models fail abysmally because they all sound like records, they just don't have that organ sound. If I think too hard about the equipment I've got on stage I would worry, because a lot of it is a bit Mickey Mouse. I take a perverse delight from going on stage with lots of adults in the audience and playing on a little kids' keyboard." Shuttleworth classics such as 'Pigeons In Flight' and 'Shopkeepers In The North' are all powered by PSS-series auto-accompaniment patterns, with liberal use of the novelty features. "I'm of a certain standard on the keyboard, I know how to get a good sound out of it with three fingers on my left hand and creative use of the Fill button. But because they're mini-keys and I've got quite big fingers, you cannot help but hit the wrong note, so I rely on natural accidents. For early John Shuttleworth gigs I used to have a big shell of a keyboard, a big wooden thing, and I used to have the backing tracks of John's songs on a cassette player. I'd give the impression that I was playing the track, but it was all on tape. And that's partly because I couldn't play very well, and the big wooden things didn't have any auto-accompaniment, it was just for visual effect. But occasionally I prerecorded a mistake."
Playing With Time He may be mocking the engineering geek inside all of us, but Brian Appleton is
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Graham Fellows (aka John Shuttleworth)
also an extension of his creator's own love of recording and sonic experimentation. "When I was a kid I used to have a little Telefunken recorder, and I used to do an impression of a baby by going 'wah wah' and speeding it up. What interests me about sound is the romance of it. I just love that control you have, where you're stopping time and you're starting to play around with it." Not only does Fellows do all the voices for radio series like The Shuttleworths, but he has also recorded all of them himself. "If I get a good line that I really like then I'll write that down, but I try not to use a script. This is why I started doing The Shuttleworths, years ago — I used to listen to radio plays and think how false they sounded. You could see the people standing around the microphone with the scripts. And then I John Aldred's vintage Manual Of Sound Recording provided plenty of inspiration for heard a Mike Leigh radio play, and it the character of Brian Appleton. The Aiwa just blew me away. I thought 'This is portable DAT recorder, right, sees a lot of incredible,' because you were just use for gathering sound effects and drawn in. You could see everything so ambiences. much more clearly. There were no scripts, they'd devised it in the same way he does his stage and TV plays and films, and they were actually moving around in a real environment. They recorded it where it happened. There was a scene in a Chinese restaurant and they stuck the mics on the table — none of these spot-effects people rustling knives and forks. And at times you couldn't quite hear what was going on, and that actually drew you in even more. Which went against the received wisdom of radio recording, as I understand it, where everything should be crystal clear. I think that's patronising to an audience. Sometimes it's quite nice not to quite know what's happening. "That's partly why I wanted to find a new way of recording, and I must have just stumbled on this multitrack thing with my Portastudio. I try to build it up like real conversation, which is totally organic. Basically you've got two people, one of them's talking and the other one's listening, and then at some point will either take over or interject, and then the other person has to listen. And when you're talking, you don't know what you're going to say. So as far as possible, I try to replicate that. I go to track one and I'll be John, and I'll say 'Ah, morning Ken. How are you today? Oof! You've got a bit of egg on your jumper.' Then I'll go on to Ken's track and I'll listen back to that, and when he says 'You've got a bit of egg on your jumper,' Ken might say 'No it isn't,' and that might collide with John's next bit. And if it collides in an attractive way I'll keep it and move on. If I think 'On no, it didn't sound right,' or you can't hear what either of them is saying, I might slip John's line forwards or backwards, or redo the whole thing. The way I get the women's voices is that I slow the voice down a bit, and just get a little bit of sibilance. I slow it down about 10 or 15 percent, and then when you play it back, hopefully it sounds like a woman."
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Graham Fellows (aka John Shuttleworth)
Tricking The BBC These days, series like The Shuttleworths are created on a Roland VS1680 digital multitracker, but it wasn't always so hi-tech. "The first series of The Shuttleworths, which some say is still the best one, was recorded on a Portastudio. I did the pilot on a Portastudio, and then I got the series, and they said 'That was fine for the pilot, but you must have better quality.' So I bought an ADAT, but I was struggling to get the spontaneity. The Varispeed took about a minute to increment, and I needed instant Varispeed — which I've got now, with my Roland VS1680, you just press a button and it's there, you've preset it. To have to wait 30 seconds to slow it down when I want to do Mary's voice is no good, because you've just killed the creative moment.
'How to edit audio tape' brings Graham Fellows's Revox reelto-reel recorder into the limelight.
"So I played a trick on the BBC. I mastered up a bit of four-track recording made on my Portastudio on cassette, and also mastered it up on a quarter-inch Revox at 15ips. Then I put both of them on quarter-inch. They had both basically come from the four-track cassette, but I pretended that one of them was done on the ADAT and one on the Portastudio. A top engineer at the BBC auditioned my tapes. So I played the Portastudio one first, and he said 'Oh yes, it's a bit hissy.' I said 'It was the best I could do,' and played him the other one. I said 'This was done on the ADAT,' and he said 'Oh yes, that's much better, can you hear the difference? That's fine!' So I went back and used the four-track. "But then I got a bit greedy and wanted more tracks, because I had to do a bit of bouncing, and because of the inability to cut and paste with a linear analogue system — it's like mobile phones, how did we live without them? I couldn't possibly be creative without cut and paste, which I've got now. For me, there was a steep learning curve on this, and it still does far more than I need it to, so I'm sticking with it."
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Graham Fellows (aka John Shuttleworth)
Heavy Weather "The interesting thing about doing sound effects for radio — I've spoken to quite a few people about this, and they all seem to agree — is that these blinking sample CDs with effects on are more or less rubbish," says Graham Fellows. "You have to really grab something there and then. All these CDs that have been specially made with sound effects very rarely work, I find. They're just awful. For instance, door slams, or atmosphere, or chatter in a pub, or the sound of wind — I think the BBC wind and weather CD has got 10 different types of wind on, and half of them sound like they're done on a synthesizer. "So part of the fun of making radio, I find, is to grab effects there and then and whack 'em in. That's when I use my stereo mic. I'll just go out with a portable DAT recorder, which was the first good bit of kit I bought. It's an Aiwa HHB1 Pro. The design of the rechargable batteries is awful — they get stuck in there, and I tend to use just the normal AA ones. "The other enjoyable thing about making sound effects, and the beauty of radio, is that you can fool the listener into believing something's not what it is. I had to get the sound of a cassette player being switched on, but the cassette player itself didn't sound like anything, so I got a toasted sandwich maker and generated the noise that way. That's where radio scores over TV. It's more creative, really."
In Wardrobe Graham now has his own studio room, in a converted vet's surgery — "Acoustically, it's pretty dead, because the brick wall's got bad pointing on it," he laughs — but he's put together previous radio series in a variety of unlikely locations. "I used to record The Shuttleworths in a shed when I lived in London, and the last series was John Shuttleworth strikes a Brian Appleton, part-time done in a cupboard in a pose. lecturer in Media Studies hotel off the A1. The (suspended). reason for that is that I was acting in a kids' TV series in Elstree, and I had a deadline for the radio show, so I had to do the radio recording in the evenings when I wasn't filming. I was originally going to have a room in the studios, but this room they put me in had a massive double file:///H|/SOS%2004-05/Graham%20Fellows%20%28aka%20John%20Shuttleworth%29.htm (6 of 7)9/22/2005 6:10:26 PM
Graham Fellows (aka John Shuttleworth)
wardrobe, and I realised that if I put a duvet on the door, and sheets and duvets and blankets up the side, it created quite a nice dead atmosphere, and I just recorded away. I did about four of the six episodes like that. "I tend to just stick a mic in the back of the VS and away I go. My bit of luxury is an AKG C414, which I got partly because I was doing a BBC series called Radio Shuttleworth. It involved recording in the studio, interviewing studio guests, but then I had to pretend they were in my house, which involved adding new lines here — so I got one of these because it's what the BBC had. For ages I used to use an SM58, or the one that was sold to me on the strength of it being Elkie Brooks's preferred live mic, the AKG C535. I've got a few SM58s, and I've got an SM57 for if I'm doing drums or anything. I've got quite a nice stereo mic that I use for recording sound effects."
The Very Best As well as perfecting his tape splicing, Graham Fellows has also been experimenting with video editing, and is now putting the finishing touches to a forthcoming John Shuttleworth film, a collaboration with photographer Martin Parr. Meanwhile, Brian Appleton is heading out on the road again in May, in a new show entitled The Very Best Of The Doors — In My Home. Cunningly disguised as a tribute to Jim Morrison and his cohorts, the title actually refers to a new series of sonic experiments, based on recordings of the interior doors of Brian's house. It's certainly a unique basis for a comedy show — so what can we expect? "I haven't really started writing it yet," admits Fellows. "I just like the title."
Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Hardware design paradigms: why use them?
In this article:
About The Author
Hardware design paradigms: why use them? Sounding Off Published in SOS May 2004 Print article : Close window
People : Sounding Off
When studios are increasingly dominated by software, why are we still hanging on to hardware design paradigms? Simon Price
If you've ever used Apple's OS X operating system, you'll probably have enjoyed the way that the icons in the 'Dock' (the strip of shortcuts at the bottom of the screen) grow and shrink as your mouse rolls over them in a pleasing ripple of smooth expansions and contractions. Just as in real life, as the graphical objects approach, new pieces of information are revealed. Whoever came up with the OS X Dock understood that we're generally only interested in a limited amount of information at any one time. And with a mouse or trackpad, we can certainly only alter one thing at a time. The Dock exaggerates the visual version of the 'cocktail party effect', where your hearing can only focus on one conversation at a time in a crowd of people. In fact, the human visual system is extremely centre-weighted. Try looking directly at the full-stop at the end of this sentence while trying to read what's in the next column. Unless you let your eyes flick across, the most you can say is that there is some text there, but that's about it! On a page, all the information has to be there all of the time, even when it's not being 'used' (read): it's a fixed format. A computer screen is a fluid format: information stored in the computer can be displayed in any way and at any time. I'm stating the obvious, right? Then why does so much of my
About The Author Simon Price is a freelance engineer and Pro Tools editor, and is part of the Flatpack production team. He also writes Pro Tools Notes
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Hardware design paradigms: why use them?
for SOS. music software present me with a static screen crammed with tiny dials? I sometimes find using Native Instruments' Reaktor (for example) a physically painful experience. Trying to program a step sequence by making sub-millimetre mouse movements on a row of 16 knobs, each a few pixels across, feels like some kind of bizarre torture instead of a fun creative exercise. The SAS should use it as one of the 'stress positions' they put captured enemies in during interrogation. Why, in any case, does a 'virtual' row of knobs have to look anything like a real row of knobs?
There are, in fact, some good answers to this last question. Knobs and faders are very efficient interfaces because they both control a parameter and display its current value. A row of faders is in fact a decent information design solution, translating smoothly from the physical world to the on-screen GUI world. Furthermore, by mimicking existing physical instruments and equipment, you make the software instantly usable by anyone who already knows how to use the hardware. Besides, sometimes you wouldn't want it any other way: for Arturia's Minimoog V plug-in to not look like a Minimoog would be just plain disappointing. But how far can this go? The cabling on the back of the virtual rack in Propellerhead Reason makes it easy to figure out how to connect devices together, as well as being an awesome marketing meme. But look what happens when you get loads of cables packed in there: you can't see where anything is going. In the real world, you'd grab hold of the cable and feed it through your hand until you found the end. In Reason, you can grab the end of a jack and the other cables become transparent so you can see what's happening. Here the need for an unreal, non-real-world solution was identified and met. It's interesting to look at circumstances where the software world surpasses the hardware interface that's being replaced. Two examples are audio editors and samplers. Chopping up and manipulating waveforms is far easier than editing tape, and a computer screen beats an LCD display hands down when creating a complex multi-zone sample patch. What's significant, I'd suggest, is that both these tasks were previously handled by devices with difficult and non-intuitive interfaces. A modular analogue synth, on the other hand, has a great user interface, but when reproduced on-screen, it's no longer so great. Sooner or later the usefulness of simulating a hardware synth will dwindle, as the majority of users will have started out on software studios. What's needed is a complete re-think of how we interact with these devices. You can start by removing all the unnecessary graphics and making efficient use of the screen. Ditch that realistically-drawn oscillator waveform selector knob that sits next to four LEDs and little squiggly wave pictures, and give me a big honking square button that steps through the options and displays the current shape. Let me just click on something and then take over with the cursor keys — if I have to struggle with fine mouse adjustments, the interface has failed. Show me one thing at a time and make it big! Take a look at Native Instruments' Battery: you click on the drum pad you want to edit and half the screen gets devoted to it. Or, try Pro Tools' simple editing system with its extreme zoom key, and context-sensitive
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Hardware design paradigms: why use them?
cursor. Finally, consider Ableton's Live: someone has really thought about that bare-bones pots and single-track focus display. Try it out on someone who's under 10, and they're not worrying about the lack of little drop-shadowed fader caps: they're banging out tunes in next to no time. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Matmos
In this article:
Grammar & Vocabulary Far From The Cutting Edge Transparent Tools Making Contact Body Popping Suck It And See Alarm Calls Shuffle Beat Children And Animals
Matmos The Art Of Cut & Paste Published in SOS May 2004 Print article : Close window
People : Artists/Engineers/Producers/Programmers
With an emphasis on cut-and-paste collages and bizarre field recordings, San Franciscan duo Matmos have created some unique and often disturbing electronic music. Tom Doyle
Cast even a cursory glance at the list of sound sources on a Matmos album and straight away you know you're not dealing with any ordinary electronic duo. Alongside the more typical programming and playing credits, the San Francisco-based twosome have taken the notion of 'found sound' to its ultimate, quite possibly ridiculous conclusion. A typical Matmos track can feature contact microphones brushing human hair, the pages of a Bible turning, rock salt being crunched underfoot, aspirin tablets being thrown at a drum kit from the other side of a room, "amplified crayfish nerve tissue" and — oh yes — the very real sound of human fat being sucked through tubes during liposuction surgery. With both their own releases and their studio/live work with Björk — they programmed much of 2001's Vespertine and toured the world with the Icelandic chanteuse in 2003 — the pair appear to be on an almost compulsive mission to seek out the most extreme and unusual audio elements to meld with their artful, angular electronica. Their latest album, The Civil War, boasts an unarguably unique concept: fusing the sounds of acoustic folk instruments like hurdy-gurdy and dobro with modular synths and recordings of human blood flow, made using an ultrasonic Doppler flow detector borrowed from a science museum.
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Matmos
Grammar & Vocabulary Martin Schmidt and Drew Daniel met in the early '90s when the latter relocated to the Bay Area from Louisville, Kentucky. While Schmidt's musical grounding was in industrial music and Daniel's was in hardcore US punk and leftfield electronica, the two initially bonded over an avant-garde '60s recording by Pierre Henry called Variations On A Door And A Sigh. "It's an incredibly thorough exploration of a very stripped-down set of sounds," Drew explains. "Just the door of a country house in France and human breathing. He spent months and months cutting tape and creating this insane musique concrète piece out of this very restricted palette." "We were both turned on by this," Martin continues, "so we wanted to do stuff like that. This was 1991 or something and I'd just been introduced to Sound Edit — at the time it was mono and eight-bit. So we spent hundreds of hours cutting together every stupid record and weird sound that we could think of and made this 60-minute musique concrète piece, which was probably 35 minutes of garbage and maybe 25 minutes of pretty good stuff." "But it taught us to have faith in the process," stresses Drew. "Even though I don't think what we made was amazing, it was so fun to do. Then we started to make dance music patterns but have them play only these noisy, difficult, strange sounds. It was almost like the vocabulary was from industrial music and musique concrète, but the grammar was the cut-up of breakbeat techno. That's when we started sounding like Matmos."
Far From The Cutting Edge Considering the reputation that Matmos are building as a unit operating on the more progressive fringes of electronic music, it's perhaps surprising that the pair freely acknowledge that they've very little interest in following developments in music technology, to the point where they don't even bother keeping up with software updates. "We aren't the most cutting-edge band as far as technology's concerned," Drew admits. "We're a funny sort of mixture of 1989 and 1993." "No, it's later than that — we're up to '97," argues Martin. "We're not cuttingedge, but it's not like the equipment doesn't matter. We've always been more believers in really getting comfortable with what we've got rather than constantly updating and having to the learn the new damn way of doing things. Especially file:///H|/SOS%2004-05/Matmos.htm (2 of 10)9/22/2005 6:10:35 PM
Matmos
old people like me, y'know, I'm 40. I want all my components in separate steel boxes." "We're very traditionally a cut-and-paste band," Drew states. "The cut-up is the core of what makes a Matmos song Matmos, whether the sound source is surgery or an acoustic guitar. It's always something that's getting put onto the screen and chopped into many fragments and then probably put into a sampler and then addressed through sequencing. So there's always an electronic component, but it's not all that different from the way that, y'know, the Art Of Noise worked."
Transparent Tools Having grown from a cottage industry-like operation where Schmidt and Daniel released limited-edition runs of 1000 CDs on their own 'vanity label' Vague Terrain, Matmos continued mixing and recording everything in their home studio in San Francisco even after signing to US indie Matador. Their setup is based around a G3 Apple Mac with a MOTU 2408 interface, and they still run Mac OS 9, primarily to support the older versions of their key software tools. Their main sequencer has for years been MOTU's Digital Performer. "Honestly, what I like about Digital Performer is that it's what I started with in 1987," declares Martin. "As far as MIDI sequencing and so on, there's a lot of parts of the interface that we don't even think about. It's the ideal thing where your tool is largely transparent to you. We literally aren't aware of upgrades because we're too busy working with it. When I heard that it was onto 4.0 or something, I was like 'Really?' One of these days, when I have an extra month to learn where they've put Copy and Paste..." "Yeah, we're using 2.7 still, we're totally living in the past," laughs Drew. "I run Logic 4.7 on my G3 laptop. When we were on the Björk tour, Logic was a great sketchpad for me. It's not the environment where I make Matmos songs, but it's fun when you're on the road, putting together sequences in the hotel room. Frankly, the last time they updated Digital Performer, a lot of the changes were definitely for the worse. Like, for example, writing volume curves in audio, it's slower now, because you've got to go through the automation." "It's probably much better for people who have those breakout digital mixing boards where it's all slaved to an external thing," Martin points out. "But that stuff kind of boggles my mind because I've got a mixing board. So now I'm gonna get a computer program that simulates a mixing board, and then get a mixing board to mix it? It's like wheels within wheels. The way we end up doing it is by writing the volume curves manually anyway, and where dynamic hands-on mixing is needed, I'm fine with just breaking everything out to individual channels in an actual mixer and riding the gains when we mix it down."
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Matmos
Drew adds: "The other thing is, for us, philosophically, because much of the point of our music is that you're able to recognise something about the object that's the source of the music, we're not particularly interested in massive amounts of signal processing or plug-ins." "We almost never use a plug-in," explains Martin. "We have an outboard reverb — a Lexicon MPX500 — and that's about the only thing we'll ever add." As far as other software is concerned, Drew in particular sings the praises of Macromedia's Sound Edit 16, Matmos's main tool when it Matmos recently rebuilt their studio in a San Francisco arts comes to sound transformation. "It's a very, very centre, creating an installation simple but very dynamic program in which I can with which members of the make all of the kind of pitch alterations and audience could interact. Drew subtle stereo manipulations that I want. Then I Daniel (foreground) tends to their just sample them flatly and they're triggered Powerbook and Emu sampler, while Martin Schmidt vocalises. through MIDI. So we're not using a lot of audio files that are being run though truckloads of DSPs, because every single sound has been hot-rodded and tweaked and plucked in Sound Edit. That's often where we're making EQ changes as well. I mean the majority of what we do to transform samples is simply to crop them, pitch them and alter them, layer after layer after layer. I'm not putting it into some plug-in and wiggling the mouse until it sounds good. It's something that's much slower, more labour-intensive, but it's also maybe why our records have more character. You're not gonna hear something and recognise the plug-in." "What was that VST one that was so on everything?" wonders Martin. "Shuffler. You'd hear something and think 'Ah, it's another Shuffler track.'" The pair are similarly unfussy when it comes to their choice of hardware. Their main sampler is an Emu E6400, though they admit they still use their old Roland W30 in every session. "Drew's been using that thing since 1989," says Martin, "so his hand goes into an animated blur when he's working with it. It's only 12-bit, but obviously we're not into audiophile recording because these things are tweaked a lot." As far as synths are concerned, they use a Korg Mono/Poly, a Roland SH101 and the newest addition to their stripped-down studio environment, the Korg MS2000B. "It has this remarkable feature of being able to remember your sounds, which I found amazing," Martin enthuses, mock-sarcastically. "You make a really cool sound and, damn, you press a few buttons and you can get it back later, which is sort of a new thing for me. We used that on the Björk tour. It was the main bass-line machine. Maybe not the best bass-line machine, to be honest, but it has a certain quality."
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Making Contact Given the extent to which Matmos's music relies on field recordings, you'd imagine that their choice of microphones might play a big part in their working methods. The answer is sometimes yes, sometimes no. "Our first couple of records were made with not even a Shure 57," remembers Martin, "but a Shure 57 knock-off. I don't even know what brand it was. Then I started working at the San Francisco Art Institute, and I was in charge of the equipment check-out cage, so fortunately we then had Tascam DAP1 and Sony DA10 portable DAT machines and a bunch of Sennheisers — a couple of ME80s, an ME40 — and an Audio-Technica AT822, the stereo mic, we got to borrow all this stuff. When we made a little money we bought an AKG 414. "Our initial strategy with any song is to either field-record or home record with some combination of microphones, straight into the DAT machine, whether it's playing a skull with chopsticks, or taking a mixdown of a song we had that sounded sort of like ducks and playing it to some ducks and then recording that using a Sennheiser binarual microphone, and then cutting that up and including it in the sequence." The duo's most prized purchase of late is a Barcus-Berry 4000XL Planar Wave Transducer, the latest in a line of various contact microphones Matmos have bought or borrowed. "For years I used a Schaller Oyster which I think was made to stick on acoustic guitars," Martin says. "Now we've got this Barcus-Berry thing, which you're supposed to put in the sound hole of a piano. It comes with this special gooey tape stuff that you put in the piano and jam the thing down into. It takes phantom power and it's a real badass contact microphone, I gotta say."
Body Popping Matmos's most strictly conceptual album to date, 2001's A Chance To Cut Is A Chance To Cure, used the sounds of plastic surgery to create techno patterns. All of these field recordings were made in California by Drew with patients who'd volunteered to have their operations committed to audio. "I recorded three nose jobs, a chin implant, two laser eye surgeries and two liposuctions," he recalls. "And I'm a real scaredy cat. I mean, I've passed out giving blood. So for me to be in the room was very disturbing. "We had to really play the art card to get everyone to agree to it. Even then some surgeons really didn't trust me, for obvious reasons. It's a very litigious society in America and you're in a sterile environment and if anything bad happened to the patient, there'd be a terrible lawsuit. For example, I went to record a laser eye surgery and the surgeon said 'OK, but you can't be in the room, so I'm gonna tape your microphone over here and record the room sound.' Obviously I couldn't push him around. So what I got was this amazing recording of horrible electrical interference because they'd taped my microphone to a high-powered laser. You kind of have to work with what you get. But luckily
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that person had complications and had to go back for even more surgery! So this time I was able to put the microphone in their lap." Drew admits that often, the more dramatic the surgical procedure, the less interesting the sound was when he got the tapes back home. "When they do the osteotomy in nose jobs, they slide a chisel inside the nose and they break the bones of the face with a hammer. But it kinds of sounds like a pencil hitting a desk or something, it's really boring. Whereas other things that are pretty humble are actually really great — some of the brushes that they use to clean wound areas are cool, and the inflatable life support blanket is really great. They lay it over the patient and there's air streaming into it and sometimes it sounded really nice. It's not dramatic, y'know, it's a blanket. But as audio, it had this constant whirring rush that was really soft and beautiful and if you moved the microphone across it, you could actually play it. You're in this environment that's so charged on the level of your sympathy as a human being, but you're also there to gather it as audio and so you're just thinking about it in terms of sound. "The most disturbing operation I watched was this one forehead lift where they were severing the muscles in this woman's head through these incisions in her hairline and all of a sudden I noticed smoke coming out of the wounds as they cauterised the muscles. The sound was like bacon frying, which was her muscles burning inside her head. But a lot of the time what was more surprising is that a surgery room is a workplace like any other. Normally the surgeons would listen to music. They listen to the Beach Boys and stuff."
Suck It And See One important element in the creation of Matmos's music appears to be trying to match wildly different sonic elements, following idiosyncratic trains of thought. For instance, on 'Lipostudio' from A Chance To Cut Is A Chance To Cure, Drew says "We sort of thought, OK, liposuction is the sound of sucking through tubes, so we brought in a clarinet player and then we got Martin breathing and sucking through a pen in a bowl of water, so we're moving back and forth between the field recordings and the music." A similarly skewed process led to them blending a recording of their sometime collaborator Keenan Lawler playing a steel guitar in a sewer pipe with the sound of the blood rushing through Martin's veins recorded by the ultrasonic Doppler flow detector. "It was the same thing, we just felt the two things resembled each other," Drew adds. Another favourite gizmo is an acupuncture point detector. "It's a little bit like a lie detector in that it measures, very crudely, the conductivity of your skin," Martin explains. "It's basically like a 9 Volt battery and a buzzer. You hold one end of the circuit in one hand and you press the other end against your skin. You can rub it along your arm and when you hit one of these acupuncture points, it buzzes, file:///H|/SOS%2004-05/Matmos.htm (6 of 10)9/22/2005 6:10:35 PM
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because it's more conductive there." Having taken a distinct leftfield turn with A Chance To Cut Is A Chance To Cure (see box, above) Matmos developed their fourth album, The Civil War, from an aborted project concentrating on pianos. "We bought a piano down in LA," Martin recalls, "and did this whole performance piece out in the desert where we filled with it with video cameras and microphones and dragged it behind a truck. But the piano songs weren't as easy to finish. So we took all the songs that weren't piano-centred and realised that they all hung together in this weird, folksy, war-like way." The opening track of The Civil War, 'Regicide', has to be the first instance of a hurdy-gurdy being used on an electronic record, with Martin's recorded parts retooled by Drew in Sound Edit 16. "Our hurdy-gurdy is just two drone strings and then a string that you can play the 'melody' on," elaborates Drew. "It doesn't have the thing which baroque hurdy-gurdys have which is called the dog, the buzzing thing that you can use to play bass patterns. So we had to construct the artificial texture of the buzzing of the dog, which I did in Sound Edit by taking the sound of Martin's fingers tapping the keys and then distorting them and stripping out EQ until I had the right buzzes. Then I played with the phase of the sample. Martin had played a melodic line that I liked and I took each note and I bent the right and left channels slightly off of each other. Almost all the hurdy-gurdy sounds in 'Regicide', if you listen on headphones, the right and left channel are always swimming in and out of line with each other. It's like floor in the room is tilted or the tape is getting chewed up." Two other tracks on The Civil War — 'Reconstruction' and 'Y.T.T.E.' — feature improvised experiments with Buchla and Serge modular synths recorded at Harvard University's Studio for Electro-Acoustic Composition. "We were doing a seminar there," Drew remembers, "and we knew that if we asked nicely we might get access to their really amazing modular synths. There was one feverish night when Keith Fullerton Whitman and Martin got to play them together. The recording was actually a blind date duet thing, where Keith was playing the Buchla and Martin was playing the Serge. They were just running patches and trying to coax the most eerie squeals out of these modular synthesizers. And you know that you're probably never going to be able to reprogram these sounds. It's like photographing a cloud or something." One other key piece of equipment, used by Drew to create the skipping guitar solo in 'Y.T.T.E.' is the Numark Axis 8 DJ CD player. "It's a CDJ like the others but with a few more effects inside it. We started with a really over-the-top screaming psychedelic guitar solo and that was burned onto a CD that I scratched with a razor. So the CD of the solo is skipping and it's being skipped again and then it was recorded as audio into Digital Performer and then we cut it file:///H|/SOS%2004-05/Matmos.htm (7 of 10)9/22/2005 6:10:35 PM
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up again. So it had three levels of interference, and sometimes it's very obviously a guitar line and then at other times it's very obviously a CD skipping."
Alarm Calls Perhaps it was inevitable that at some point Matmos's sonic experiments would come to the attention of the queen of otherworldly techno, Björk. After a friend had given the singer a copy of the duo's 1998 album Quasi-Objects, she apparently ordered another 25 copies to hand out to her friends. Soon after, she called the pair in San Francisco and asked them to do a remix of 'Alarm Call'. "It was great for us," Drew admits, "because we weren't really sure of how to work with a vocal that hogs the mid-range and a lot of the frequencies that we normally fill with some freaked-out cut-up. So I spent a very long time with her lead vocal, chopping all the gasps out, 'cause she's such a gasper. I came up with this family of 36 different gasps and made the rhythm out of that for the first third of the song. We must have spent two weeks on that one remix. She called up and said [in a thick Icelandic accent] 'I am flabbergasted!'" "So then," Martin continues, "it was like 'So I want you to be my band for live.' We were like Martin Schmidt generates 'What? I don't know if you've noticed, but we another sample to receive the don't really make music.' She was like, 'No no it'll Matmos cut-and-paste treatment. be fine... so it's all settled then?' She had a faith that was well-founded, I guess. People clapped and shouted." Drew has his own theory about why Björk was so attracted to Matmos. "She has very pro environments around her, and so she likes to bring in a couple of people that are not like that at all. If she needs to get a string arrangement for the LSO then she'll get Vic Mendoza, but I think the appeal with us is that we're not professional in any way. We're kind of self-taught and everything is done on this really humble at-home level. She kind of likes to mix together people who know what they're doing with people who're just kind of oddballs." The result was Matmos providing the electronic rhythm section for Björk's threemonth world jaunt in 2003, in a touring band that also featured the Icelandic String Octet and harpist Zeena Parkins. The pair spent four months reprogramming 40 tracks from the singer's back catalogue, lifting elements from the original Logic tracks and transferring them into Digital Performer in the G3. And while their Emu 6400 was obviously flightcased and taken on the road, their faithful Roland W30 came too. "I use it to make live samples of Martin while he's playing an instrument because the turnaround time is so much faster than the file:///H|/SOS%2004-05/Matmos.htm (8 of 10)9/22/2005 6:10:35 PM
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Emu," says Drew. "It's just a really quick way to be in the moment and yet using a sampler. So why sacrifice it for something that's going to be so complicated that you've got to jump through five sub-windows before you can perform a task?"
Shuffle Beat Drew admits that they perhaps 'over-bullet-proofed' their setup for the tour, but points out that they still left enough room for on-stage improvisation of a typically Matmos-like nature. "On 'Hidden Place', some of the rhythms that were programmed are the sound of cards shuffling, so when we played it on stage, Martin would shuffle cards — it was like doing sound effects for a radio play. Then for the song 'Aurora', Martin would walk on rock salt to create the rhythm. "What I do is kind of a different thing. For each song I have a family of 10, 15, 20 different patterns in Digital Performer and I sit and choose which ones come in when, just clicking on loops. I have a Peavey PC1600X with different parameters of the banks of sounds mapped to each fader controller." Even though the system was supposedly bullet-proof, as you might imagine, the proceedings weren't entirely hitch-free. "Oh God," Drew squirms. "The worst crash was at the Coachella Festival in Palm Springs in front of 30,000 people. My system completely crashed five minutes before the show. We had all the fans going 'Björk! Björk!' and everyone was shitting themselves because we couldn't figure out what the hell was wrong. We thought we'd fixed the problem, we started the show and a few minutes into the first song, we realised that it still wasn't receiving MIDI. So we had this weird skeletal half-arsed thing going on with some of the bass lines and kick drums playing off the Logic rack and nothing at all coming off my rack. I was pretending to play a keyboard with my left hand and sort of smiling at the audience, and with my right hand I was restarting my computer over and over and over. It was your worst nightmare. Björk thought it was really funny, which is why she's a pretty good boss."
Children And Animals Looking to the near future, Matmos will be busy programming with Björk for her next album and getting ready to release a limited-edition EP of tracks created from the sounds of a squealing rat that they caught in a humane rodent trap in their house, as well as searching out ideas for their next full album. Recently the pair rebuilt their entire studio in the Yerba Buena Center For The Arts in San file:///H|/SOS%2004-05/Matmos.htm (9 of 10)9/22/2005 6:10:35 PM
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Francisco for a three-week installation, where they made music interacting with the public. "We did improvs with five-year-old children every Wednesday and Friday and they were so fun," Drew notes, wryly. "So, I dunno, we might do an album called Matmos Loves The Little Children." Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Studio SOS
In this article:
Studio SOS
Quick Mastering Tips Jiang Li Noise Remedies & Published in SOS May 2004 Monitoring Tweaks Processing Submixes Print article : Close window Global Musical Collaboration People : Studio SOS Network Rebalancing The Mix Li's Comments On The Session Recording Traditional The SOS team apply themselves to the task of mixing Chinese Instruments Chinese traditional instruments with mainstream
Western sounds at reader Jiang Li's home studio. Paul White
Jiang Li not only records himself and other musicians at his small home studio, but also collaborates with musicians from different parts of the world over the Internet using a system similar to that pioneered by Rocket Networks. However, he was having a few problems with his recording and mixing and so invited us to his studio, which is packed into a fairly small ground-floor room in his Northamptonshire home.
A little Auralex foam put paid to some troublesome wall reflections in Li's room, but Hugh also felt that the speakers were not angled optimally for a seated engineer, so he wedged them up at a slight angle in order to aim the tweeters more appropriately.
The studio is a little under 3.5m wide and about 2.5m deep, with a ceiling height that is typical of modern houses. The entrance door is in the rear wall of the studio and, once all the gear is in place, the engineer's seat is well under one metre from the door. The monitoring is courtesy of a pair of Mackie HR824s set up on a rigid shelf above the Behringer Eurodesk mixing console. His latest acquisition was a Mac G5 running Pro Tools LE using a Digi 002 interface, but along with many other G5 owners Li had discovered an unacceptable rhythmic background noise problem when using a Firewire audio interface. This isn't something we could deal with, as it seems like an Apple hardware-related problem affecting some G5 machines — for more details about this, check out Mark Wherry's Apple Notes column this month. Because of this
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noise problem, Li has kept his Cubase project on his older 450MHz Mac G4, which works perfectly but tends to run up against its processing limits when lots of plug-ins are being used. He also has a fast PC laptop (with an external 200GB Maxtor drive) which is used to run Steinberg's Wavelab for editing, and also Tascam Gigastudio.
Quick Mastering Tips It's always a concern when clients complain their mixes sound quiet next to commercial recordings, but the truth is that many records end up being overcompressed purely to meet this need for loudness. Li was interested to know what treatment was appropriate, given that much of his material would not be commercially mastered. So we suggested cautious post-EQ limiting (just a few decibels off the loudest peaks) and gentle EQ to clarify the mid-range and to bring out high-end detail. Using an analogue EQ, this usually means applying around 2dB of cut in the 180-200Hz region and adding a wide, gentle boost of no more than a couple of decibels at 12-14kHz with a Q value of not higher than 0.7, though this has to be done while listening to see what the track needs — it shouldn't be used as a blanket sweetener for everything. Like most digital equalisers, I found that I had to apply more cut and boost with the Pro Tools LE parametric EQ plug-in to get the same subjective result, but operating the bypass button showed that the difference was suitably subtle. Again, comparing our processed result with Li's original mix showed that our version sounded noticeably louder and it had more clarity, so these simple tricks clearly work, even though there's a huge amount more to 'real' mastering than this.
Noise Remedies & Monitoring Tweaks There was a noticeable hum on the system at high monitoring levels, but because of the way everything was packed into the room we couldn't do any impromptu rewiring. We did however suggest that Li tidy up his mains connections and route them all from the same wall socket, because the overall power requirement was low enough for this not to be a problem. Further improvements can be made when connecting unbalanced sources, such as synths and samplers, to a mixer with balanced line inputs by using the unbalanced-to-balanced cable connection system whereby the balanced end of the cable is wired as normal (using twin-cored screened cable), but at the unbalanced end the screen is left disconnected and the cold (usually black or blue) cable core is connected to the tag where the screen normally goes. file:///H|/SOS%2004-05/Studio%20SOS.htm (2 of 8)9/22/2005 6:10:40 PM
The Digi 002 rackmount interface went dead during
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the session, but started up fine again after a lunch break. Hugh suspected that it might have been overheating, so he partially removed it from the rack to increase ventilation for the remainder of the session.
Before listening to Li's mixes in detail, we wanted to evaluate the monitor system's performance, so we auditioned some commercial mixes through the system and felt the result was rather bass heavy. Checking out the switch settings on the rear of the HR824s revealed that Li had set them to the correct half-space mode, but with their full bass extension. I found this setting was too low for my own studio, which is more than twice the size of Li's. I have my bass switches set to their middle positions (47Hz), so suggested Li tried the same. The other issue was that the shelf supporting the monitors was quite high, resulting in the tweeters being well above ear level when Li sat in the chair to mix. Therefore we placed a wedge under the rear of each speaker so that the tweeters were pointing toward where the engineer's head would be when seated. Repeating the CD playing tests indicated that the balance was now more manageable and, surprisingly, the bass end was fairly even, probably due to the thin rear door acting as a bass trap (or bass leak) and the window in the side wall also tending to do the same. However, the room was not acoustically symmetrical, as the window was to the right and fitted with slatted blinds (helping to break up reflections) while the left-hand wall was bare painted plaster. To help even this up, we brought in some Auralex foam oddments left over from a previous Studio SOS visit and leant one up against the left-hand wall, put one behind the monitors, and propped one up against the hard surface of the door. We left it to Li to find a way to fix these more securely without actually gluing them to the walls — it's easy enough to fix a piece of string across the back and hang them up on picture hooks. A re-test with the commercial CDs showed that the image had clarified to a worthwhile extent, and at this point we felt we had got the monitoring about as right as we could, given the physical constraints of the studio.
Processing Submixes Next we turned our attention to a mix that was typical of the ones Li was having problems with. This one comprised a mixture of western and Chinese instruments, where the orchestral sounds and some of the percussion sounds were samples, the acoustic bass sound came from a Proteus 2000, and the voice and acoustic Chinese instruments/percussion had been recorded by Li. The problem Hugh and I identified straight away was that the recorded acoustic instruments didn't sit very well alongside the samples — the sampled orchestral sounds had a concert-hall feel, while the acoustic instruments sounded as though they were still in a fairly small room, even though Li had added
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some reverb.
Once the monitoring system was in order, Paul set about tweaking Li's mix to get the sampled instruments sitting alongside the recorded ones.
The vocal also sounded a little thin and uneven in level, and where Li had submixed a number of Chinese instruments to a single stereo track we found that some of the instruments really jumped out while others were getting buried. It seems that the limitations of the G4 had forced Li into working with stereo submixes of sets of the original tracks. It would have been preferable and more flexible had he kept the individual instruments separate and controlled them with fader groups instead. However, this is something he can explore at leisure using his G5 once he's sorted out the noise problems. To make the best of the existing job, we first compressed the vocal track using Li's Focusrite Penta compressor, which has presets for various applications. Although I don't normally like using presets, the Penta allows further changes to be made once a preset has been selected, so we used the Vocal preset as the obvious starting point and then adjusted the threshold to give five or six decibels of gain reduction on the loudest peaks — which seemed enough to even out the sound. We also added just a small amount of a Chamber reverb patch from Li's Lexicon Alex processor to make the vocal sit in the same kind of space as the sampled backing parts. Good reverb is very important when mixing, but the Lexicon Alex is now rather long in the tooth and probably isn't much of an improvement over run-of-the-mill native reverb plug-ins. Once Li's G5 is behaving properly, he'll have power to spare to run a convolution-based reverb plug-in (such as Audio Ease Altiverb or the new Waves plug-in), which should make a vast improvement in this area.
Global Musical Collaboration Network Jiang Li has been bringing music from the East and the West together for the past ten years as a musician and composer, but he has also recently been awarded a fellowship programme from NESTA (National Endowment for Science, Technology and the Arts) to develop the web-based Global Musical Collaboration Network. The idea of this is both to educate people about music from different cultures (via audio and video examples of music from different cultures) and to facilitate co-operative projects by allowing musicians from different parts of the world to collaborate. Although this free-membership community is still in its infancy it already includes a live chat room, a forum, a technical FAQ section, collaboration projects, and an on-line radio station. It also includes a showcase section for members to publish themselves and to search for collaborators worldwide. As a registered member, you can take part in projects and also start up your own. The different levels of collaborations are suitable for amateur musicians up to professional studio audio engineers. www.globalmusicalcollaboration.net
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Rebalancing The Mix That left us with the Chinese instruments mix to sort out, but compressing this with the dynamics section in Cubase happily made a huge difference. Given the levels Li had recorded, the threshold was set to -20dB and the ratio to 5:1, with the attack time fast and the release time on automatic. These settings worked well to control the loud instruments and to lift out the parts that were previously getting lost. Before rebalancing the track, we thinned out the string samples by cutting back the low end, and also took off a little top, as they were just starting to sound strident. A further supporting pad was brought down in level. We tried to EQ the string bass sound to get more definition, but found that we needed huge amounts of EQ boost to do this, as there was very little presence in the original sample. The problem with using this amount of EQ was that the string squeak sound Li had added really leapt out, so we used a compromise setting and suggested that Li find some good acoustic bass samples to use for his next project, rather than trying to re-shape the bass sound from his Proteus 2000. To mix the track, we started with all the faders down and then balanced the bass, drums/percussion, and vocal so they worked together. After that it was fairly straightforward to bring in the strings and brass samples, the Chinese instrument mix, and the individual Chinese instruments to achieve a cohesive balance. Where a little reverb was needed to add a sense of space, the same Lexicon Alex Chamber preset was used. When we compared this remix with Li's original version, it transpired that our mix had a higher level of drums, but what was most noticeable was that the individually recorded instruments no longer sounded out of place, and the overall feel of the mix was less crowded because we'd allowed some of the parts, specifically the strings and pad, to play more of a supporting role.
The bright ambience of Li's kitchen was ideal for recording Li's traditional yangqin.
Li asked if we had any special techniques that could be used to create more of a sense of front-to-back perspective, and in some ways we'd already applied these when mixing. As a rule, upfront sounds tend to be less reverberant and brighter than those at the rear, so if you want to keep a vocal at the front, use a shorter, brighter reverb or ambience program, and perhaps use EQ to add a little 'air' into the sound above 10-12kHz using a wide parametric boost. Sounds to the rear can afford to lose a little high end and can be made more reverberant. On this particular track, the vocals were more of a supporting effect than a lead line, so we could afford to use more reverb. At around this point, the Digi 002 decided to die on us, and after trying to reboot it and checking the connections, we gave up and took a break for lunch — file:///H|/SOS%2004-05/Studio%20SOS.htm (5 of 8)9/22/2005 6:10:40 PM
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traditional Chinese dim sum presented by Li's wife. It made a welcome change from chocolate biscuits, but don't let that put anyone off offering them on future Studio SOS projects — after all, we have a tradition to uphold! After lunch, we re-powered the Digi 002 and it sprang back into life, so we wondered if its earlier failure was heat related, even though it was placed at the bottom of the open rack below the Penta, and wasn't exposed to any obvious major heat source. Just in case heat was the problem, though, Hugh temporarily removed the Digi 002 from the rack and we continued to use it this way for a while. However, it suffered a brief relapse later in the afternoon, so we felt it should be returned to base for a checkup, as something was clearly not quite right. Li has since reported that he has rearranged the rack and upgraded his Mac OS to 10.3.1 (which is approved by Digidesign) and the problem seems to have disappeared. Anyhow, it limped on gamely for the rest of our visit, allowing us to experiment with some overall 'mastering' processing. For this part of the process, Li moved our new mix of the song into Pro Tools so that we could use its plug-ins.
Li's Comments On The Session "I work with Chinese musicians a lot of the time, but have never been quite satisfied with my mixes. Also, some Western instrument sounds, such as bass, strings and so on, are difficult to integrate. Now, by adjusting a few settings on my existing gear, making some simple setup changes to the monitor speakers, and adding a little acoustic foam, I can already hear a great difference. Paul set up different EQs for different instruments on my Eurodesk, and a lot of unwanted frequencies disappeared, making the mix clearer. The compressor also made parts more audible without them being too loud. "The Studio SOS visit made me more confident and positive, in particular when I'm working with Chinese musicians who generally know very little about the technology. Paul and Hugh also looked at other rooms in my house and discovered that they were suitable for different kinds of live recordings as well, which literally expanded my small studio to include the kitchen and living room. I can now put off my plans for renting a studio elsewhere and use some money for the next items on my gear shopping list, in particular a couple of mics similar to those that Paul and Hugh brought along."
Recording Traditional Chinese Instruments Once we had shown Li our approach to mixing, the conversation turned to the file:///H|/SOS%2004-05/Studio%20SOS.htm (6 of 8)9/22/2005 6:10:40 PM
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recording of traditional Chinese instruments, of which Li has a fine collection. Although Li had a couple of decent mics, including a nice Audio-Technica 3035 and an ATM31A, he wanted to be able to record up to four musicians at a time and so was in the market for a couple more instrument mics that wouldn't break the budget. I had a couple of SE Electronics SE1s in the car, which provide a decent example of what you might expect from an entry-level small-diaphragm capacitor instrument mic. Li had been recording instruments in his hallways, which sounded boxy to the ear and gave, I felt, a rather coloured and enclosed tone to some instruments, especially those that couldn't be miked very closely. As Li can use any room on the ground floor of the house for siting the instruments, we checked out the lounge and the kitchen to evaluate their suitability. The kitchen has a hard floor and lots of hard surfaces and so, predictably, sounded quite bright, which we felt could sound good on acoustic guitars and other stringed instruments. To confirm this, we miked up a hammer dulcimer (called a yangqin) with a single SE1 mic via Li's Digi 002 preamps and got a very acceptable sound right away. For a 'real' recording however, I suggested that it would be a missed opportunity not to mic this instrument in stereo where possible, so whatever new mics Li bought, he should get two of the same so that he could use them as a stereo pair.
I've heard of playing 'Chopsticks', but this is ridiculous...
By contrast, the lounge was carpeted and contained a large three-piece suite covered in heavy fabric. The result was a much drier sound that would be ideal for vocal recording. We demonstrated this by me walking between rooms, holding a mic and reading a manual, while Li and Hugh listened (and sniggered!) in the control room. The change in sound when moving from the kitchen to the lounge was massive so, unbeknownst to him, Li already had two very different and very useful recording rooms at his disposal. Even the coloured-sounding hallway could be useful for some sounds, and where good isolation was needed three instruments could be recorded simultaneously in the three spaces, plus a fourth in the studio, with very little spill. Each Studio SOS visit throws up some common themes and some unexpected questions. In this instance, I think we were able to show Li a slightly different approach to mixing and sound shaping, and exploring the acoustic spaces in his house was also a worthwhile exercise. However, all this is pointless if the monitoring system isn't giving a good account of the truth, which is why we always try to optimise this at the outset. This visit was also unique in that the word duvet was never mentioned! Damn, I've blown it now... Published in SOS May 2004
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Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Todd Rundgren
In this article:
Inter Action Todd Rundgren, Record Producer Shapes And Sounds A Certain Sound Todd Rundgren The Engineer The Virtual Studio No Secret Any More
Todd Rundgren Recording Liars Published in SOS May 2004 Print article : Close window
People : Artists/Engineers/Producers/Programmers
As a musician, engineer, producer and multimedia artist, Todd Rundgren has been a pioneering figure in rock music since the late 1960s. His first album of new material for 10 years was, like many of its predecessors, almost entirely played, produced and recorded by Rundgren alone. Paul Tingen
More than three decades ago a lanky youth with eccentric, gaunt features and large eyes sprang onto the international music stage with a remarkable double album called Something/ Anything? The young man had written and produced all the songs, and on three of the four vinyl sides, had played all the instruments and sung all the vocals. The 1972 double album contained such a richness of topdrawer songs, nifty arrangements, and virtuoso playing and singing that it's still regarded as a timeless classic. The youth's next album, released in 1973, was called A Wizard, A True Star, and contained a brilliant sonic collage full of studio trickery, which was not only produced but also engineered by the man himself. His name was, of course, Todd Rundgren.
Photo: Danny O'Conner
Inter Action It took 10 more years for Rundgren to acquire a status close to that of living legend. In addition to his thriving solo career, he formed, in 1974, a band called file:///H|/SOS%2004-05/Todd%20Rundgren.htm (1 of 11)9/22/2005 6:10:45 PM
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Utopia, which recorded an array of albums during the 1970s and 1980s while moving stylistically from lengthy synth-driven instrumentals, via progressive metal rock, to power pop. In addition, Rundgren applied his knowledge of the recording process as a wildly successful producer for others. Often also functioning as engineer and player he oversaw the creation of classic albums like Grand Funk Railroad's We're An American Band (1973), the New York Dolls' self-titled 1973 debut, the Tubes' Remote Control (1976), Meatloaf's Bat Out Of Hell (1977), Patti Smith's Wave (1979), and XTC's Skylarking (1986). However, the Philadelphia-born American refused to be a 'true star' in the traditional sense. Instead, he frequently took bizarre leftfield musical directions, dressed in weird camp clothing and make-up, and wrote lyrics that addressed much deeper issues than boymeets-girl. He also became a pioneer of multimedia and computer technology. For instance, Rundgren organised the world's first interactive television concert in 1978, during which home audiences could chose what songs would be played; in 1980 he created the first colour graphics tablet, which was licensed to Apple as the Utopia Graphics Tablet; and in 1982 he made the first music videos to be Todd Rundgren in more familiar commercially released, one of which was guitar-hero pose. nominated for a Grammy. He founded Secret Sound Studios in 1972 and opened Utopia Video Studios in 1978, exemplifying a keen interest in audio-visual art. His 1992 solo album No World Order was not only issued as a standard audio CD, but also on CD-I as the world's first interactive album, with the songs chopped up in four-bar segments, allowing the listener to construct his or her own song structures. In 1994, The Individualist was the world's first full-length enhanced CD, and by the late 1990s most of Rundgren's creative energies were focused on pioneering the use of the Internet as a vehicle for music delivery. Among other things, Rundgren founded two web sites, www.tr-i.com, and www. patronet.com, through which listeners can act as patrons of the arts by paying the artist upfront to have music delivered directly to them. Since 1998 Rundgren issued his music song by song via Patronet, and some of this material, including some reworkings of older songs, was compiled on One Long Year, released as a traditional CD in 2000. In several recent interviews, Rundgren has argued that CDs and record companies are outmoded routes for getting music from artist to audience. In his view, younger music lovers especially prefer to search the Internet, download what they like, and make their own MP3, MD or CD compilations. Rundgren's solution is to treat music delivery as the equivalent of cable television, with users paying a monthly fee of, say, $20, to gain access to all the music they want. Presumably, the proceeds are then divided between all those who supply music,
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according to some unspecified formula that may prove to be the tricky part to work out.
Todd Rundgren, Record Producer During the 1970s and 1980s Todd Rundgren was a big-name producer. He's produced artists like Janis Joplin, the New York Dolls, Hall & Oates, Alice Cooper, Steve Hillage, Tom Robinson, the Tubes, Patti Smith and Grand Funk, not to mention Meatloaf's multi-million-selling blockbuster Bat Out Of Hell. "The most difficult thing of all in the studio, as a producer," explains Rundgren, "is not technical. It's to get the performer in the proper frame of mind to perform as if he or she's in front of 1000 adoring fans. That's the hardest thing. For a lot of people the studio is a foreign place, because you're isolating everyone, and they don't think of themselves in the moment any more. They're all thinking, 'Oh my God, I'm making a record, I'm making a record.' "I've been involved in great records that have been artistic successes and that have been incredible commercial successes. It's hard to imagine something more commercially successful than Meatloaf, and at the same time that's musically so signature that it's also a classic. It was fun making that album. Meatloaf and Steinberg [the songwriter] are both fairly colourful personalities. It was also a tortuous record to make, because during recording Meatloaf wanted to be released by his label, and they let him off, and nobody else signed him. So I was underwriting the whole record, and it was a big gamble and not something I'd ever done before. I wasn't sure it was going to happen... even Bearsville [Rundgren's label at the time] turned him down! "The work I did with Grand Funk Railroad [on We're An American Band] was fun, because they were a completely underestimated band. They had a big audience, but their records sounded like crap, because they had been produced by their manager, who knew nothing about making records. They were actually remarkably good players and singers, which their producer had managed to disguise from everyone! The gift in doing that record was instantaneous. Success happened like the speed of light, as opposed to Meatloaf, which took two years to break. "[XTC's] Skylarking album is legendary in many respects... because of what happened between me and Andy [Partridge]. It was a little bit of a surprise for him to save all his vitriol until he was well away from me, and do it in the press, which I thought was shooting himself in the foot. It was a great album, and my attempts at making it accessible infuriated Andy. He thought it was another Andy Partridge solo album, with the help of the two other guys in the band. The problems probably started when I insisted that Colin [Moulding]'s material was as important as his... "I've done a few productions in recent years, but nothing that has jived with the commercial market. Popular music is going through an abysmal phase right now."
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Much of what Rundgren has done since the late 1980s has pushed the boundaries of the artist-audience relationship, in the search for greater interactivity. It therefore comes as a bit of a surprise to find that this April a traditional CD by him was released, called Liars, without interactive components, and put in traditional record shops by a traditional record company (Sanctuary). What's more, the synth-dominated work is hyped as a major event, because it's Rundgren's first CD release of all-new material for 10 years. "Many people think of a record as a certain collection of songs put onto a disc-shaped object," begins Rundgren. "But in the past few years we've begun to realise that it isn't the shape of the object that matters. Instead sound can be virtualised in any number of forms. It may at one point be a link on your computer screen, and then it's a thing that appears in the window of your MP3 player, and it never takes physical form. At that point the experience of the music is more and more defined by the listener."
Photo courtesy ©1980 Danny O'Conner.
Todd Rundgren's Utopia, 1980: Todd Rundgren, Willie Wilcox, Kasim Sulton and Roger Powell.
No problem here, one would say, but according to Rundgren, this does make it harder for musicians to realise their intentions. "You used to make a record and would say, 'OK, this record is 12 songs in this specific order and in this package with this cover and with these lyrics, blah, blah, blah. But now, at a certain point it gets completely redefined. Someone takes one song off this record and one song off that record, and burns themselves a CD, and this may be like nothing you imagined. That makes it very hard to say whether you're satisfying the original goals you set yourself. You have to be aware that you may have a concept about your songs appearing in a specific order, and that you may have to do something tricky to enforce that. Like on Liars, I had the songs overlap. I'm thinking here in terms of 'OK, I made a distribution deal with the record label, how do they hold on to the value of that format as long as possible?' You do that by trying to make the format unique in some sense. That's why all the songs flow together. No-one is going to download an entire album as one giant MP3 file. If you want the real experience of Liars, you're more or less forced to buy the CD. "In addition to this, I'm delivering unlocked MP3s, with fade-outs in them, to my on-line subscribers, for them to do with what they want. What I know, and what I believe the record company knows, is that those listeners are going to buy both the CD and the MP3 files. They're going to buy everything there is. That's the whole idea of patronage, to get the most dedicated listeners, the ones who make a point of supporting you, and they will get a certain amount of attention and levity and special things that other people find difficult to get. And in that sense they give you the kind of support you used to get from the record label, like underwriting making records and financing your tours."
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A Certain Sound Todd Rundgren has something of a reputation as a guitar hero, but he has always explored other instruments, and on Liars, synths and other keyboard sounds take the spotlight. Once again, Rundgren plays all the instruments (barring a couple of solo spots by others). "I was going for a certain sound, " he explains. "I gravitated toward a B3 and Fender Rhodes for the central element. To me it's old-fashioned, and I wanted it to resemble all my influences... some Mose Allison, Beatle-esque things. And there's always a bit of Marvin Gaye in there somewhere." Much of Liars therefore has a retro feel to it, dominated by influences of 1970s soul, and, rather surprisingly, four-to-the-floor 1980s house. According to Rundgren, this musical approach was shaped by the subject matter of the album. "It's a challenge to do an album called Liars, and not put yourself in a compromised position," explained Rundgren, quite understandably, and apparently not referring to the cover image of himself as a crazed Easter Bunny. "The album isn't about finger-pointing. It's about the phenomenon of dishonesty and non-reality and the many flavours there are, from the most innocent to the most venal. Not one of us is truly honest, and therefore the album is left open to accusations of fakery, which is why I'm, like always, reluctant to do any kind of breakdown of exactly how the album was constructed." This kind of reluctance usually spells trouble for interviews of a technical nature, but luckily it was possible to persuade Rundgren to elaborate on the nuts and bolts of Liars. "Part of the reason there are so many keyboards on this album is compositional," he reveals. "I can do a lot more on a keyboard than on a guitar. There are just some things you can convey with keyboards that can't be done with guitars. The other reason is that certain keyboard sounds take me to a certain time. Like the Wurlitzer piano, which nobody plays any more because they're too much of a maintenance problem. They were a giant pain in the ass, but they had a characteristic sound. When you hear the Wurlitzer on 'Get Back', it takes you immediately back to the late 1960s. The same goes for the sound of the Fender Rhodes, and the B3, and some of the old, mono, analogue synths."
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Todd Rundgren The Engineer Throughout his long career, Todd Rundgren has engineered and produced many of his solo albums, as well as often playing all the instruments. Among them are Something/Anything? (where he left the engineering to others), Hermit Of Mink Hollow (1978), A Capella (1985), No World Order (1993), and now Liars. "There's certainly been a whole evolution in working," recalls Rundgren. "After Something/Anything? I got into engineering as well. I built a studio [Secret Sound] for the recording of A Wizard, A True Star, because my approach to music was so unrestricted that it was impractical to pay a studio by the hour. I engineered almost everything that happened there. In most cases this involved pushing a button and running out into the studio to participate in the performance. I also was punching in and out for myself when recording vocals. Initiation (1975) was also done that way. "Recording Hermit Of Mink Hollow was a real hardship, because the control room was upstairs and the drums downstairs, so when trying to record drums, if I made a mistake, I had to run up and down the stairs just to rewind the machine. I didn't have a remote with a lead than ran long enough! For a long time I had a degree of aversion towards using synthetic or sequenced elements for basics like drums and bass, but eventually decided that the ends are always worth the means. My resistance came to an end on No World Order, because the music was delivered in such a radically different way, all chopped up in fourbar segments and later resequenced, that I couldn't consider another way of recording it. It's really difficult to get musicians to play with that degree of precision, so you can splice it every four bars. So it was a technical necessity to perform it substantially electronically. "For me the challenge in the studio has never been the technology. I absorbed everything I possibly could from doing my own recordings, and so when I got to do recording for other people, I had a fair amount of knowledge. You learn things like that the most important thing is mic choice and placement. Boards have nearly nothing to do with the end sound. People get fixated about signal paths and other things, but it all starts from the point where the sound is picked up. If you don't have the right mics and right placement, there's no way you can recover what you actually want.
Photo: Bob Gruen / Star File
John Seigler (bass), Ralph Schuckett (electric piano), Todd Rundgren (guitar), Moogy Klingman (organ) and John Siomos (drums) rehearse at Secret Sound Studios for Todd's album A Wizard, A True Star in 1972.
"I was never a big splicer. For a long time I did not want to have to stop and start a mix and then splice it. For me the process of mixing was the discipline of learning exactly where the faders were supposed to be at every point in the song, and riding the vocals 12-20 times until you knew exactly where the peaks and drops were. And then being able at one glorious moment to take a big breath, push the Record button and file:///H|/SOS%2004-05/Todd%20Rundgren.htm (6 of 11)9/22/2005 6:10:45 PM
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do the mix from beginning to end and have it perfect! But at a certain point I started to work more with splicing, because I said, 'Yes, the little mistakes become part of the character of the song. But if they're going to bother you at all, why put up with them if it's easy just to stop the machine, roll back a couple of feet and do the splice? "When I began doing digital records, it got much more difficult. It was easy at first because the only digital equipment that I used was a two-track, and you could still splice the early two-tracks with a razor blade. The first album I did like that was XTC's Skylarking. The first album I recorded entirely on digital was Nearly Human [1989], the live-in-the-studio one. Although everything was live, we would do many performances and take the best performances and splice them together to get the ultimate performance. That turned out to be such a timeconsuming nightmare, it put me back to a place where I thought 'Why can't I do it all in one take?' "Later on I was using ADATs for a while. I got into them not because they were digital, but because of the great convenience they offered. You had as many tracks as you needed, you just kept popping more tapes in. And I was happy with that, but then when Alesis tried to move up to higher bit density and I upgraded, I went through a nightmare project, with drop-outs and trying to get three machines to sync up and other stuff. The response I got from the company when I complained wasn't satisfactory. So I decided that this was the end of my relationship with tape. At the time I had all my multitracks baked and transferred to ADAT. Now, if I really intend to preserve the music, I have to transfer them all to hard drive. "The other thing about the way that I mixed is that it was very visual. Most studios that I have owned have been equipped with a graphic equaliser on every single channel, usually a 22-band graphic equaliser. This became a characteristic of my sound. The reason I did that was so I could visualise the sound. I could see all the places where the sound was either too harsh or inadequate like a graph. I could see whether the 22 bands represented an ideal wave shape or the opposite. It gave me a picture of how well or how poorly the sound has been recorded. That has been my approach to recording and to a certain degree it still is today. I still use equalisation, but it becomes less and less necessary because of the sophistication of modern tools. Now I'll use something like the Channel Strip plug-in. Although it's not a graphic equaliser, it has a six-band equaliser set up like it's a graphic EQ, and I create the same look and curves like I do on a graphic equaliser."
The Virtual Studio Hearing Rundgren speak about the musical influences that went into Liars, one imagines he recorded it in a wood-panelled recording studio full of valve gear. But in fact, Liars is as modern as they come — almost the entire album was recorded using only virtual instruments. "I've also recorded albums completely live in the studio," Rundgren asserts, "which is the other extreme. I know the difference, and I know what the priority is, which is to express yourself and get your message across. The music would be a complete failure if it didn't convey this. And while on this album I wanted to get a retro keyboard bass sound and B3
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and Fender Rhodes and Wurlitzer tonalities, I've never been someone who is anal about getting the original equipment when something else will do. When I got my first Line 6 equipment, I was so happy that I couldn't imagine ever buying another guitar amp, and I haven't. Some people get hooked on a certain sound and they truly believe the only way to get it is to have the original Minimoog. There's a whole market for them, and that's cool. But I'm a means-to-an-end guy. For me the end is: does the music deliver, or doesn't it? And whatever the path of least resistance is, that's the path I'm taking. "In this case it meant using a MIDI master keyboard and lots of virtual sound sources. I also used an old blue and white Mac G3 with OS 9.2, which is not fast, but it's fast enough to run Reason — my principal composition and performance tool on this record — and it's stable. I also used Pro Tools v5.1, and the Pro Tools hardware was doing most of the work anyway. I didn't have the Rewire version of Reason, so I had to go through a fairly convoluted process of printing all instruments one at a time after I had recorded everything I wanted. But Reason is great. You can output at 24-bit, which I did, and you can get a nice, rich sound. There's a huge user community out there producing sounds, which I took advantage of. Reason has so many different ways in which you can approach a composition, and the virtualisation of instruments and effects is something I can't go back from now. My virtual instruments were for the most part Reason's take on the classics. Also, I got used to the idea of not having to own Urei 1176 compressors any more. Instead I can have 20 of them virtually, if I want." The drum programming on Liars varies a lot, from subtle and convincingly real to baldly mechanical. "I was forced to go this way because of the subject matter," says Rundgren, "which involved ideas that were initially somewhat amorphous. This makes it difficult to involve other musicians, because you're not entirely sure yet what you're doing. Normally you'd bring one song to full completion, so you can teach it, but I was more or less working on all the material at once and moving it along a little bit at a time. By the time I had made a demonstration of it, I was essentially done. "Even though I'm using all these automated instruments, I'm not satisfied with an automated performance. So it was a constant process of going back, almost like a painter adding layer upon layer and then details until the crude broad strokes that the automation represents are obscured. The drums were also done in Reason. I used a combination of sequencing and loops combined with other things, and sometimes there's a real attempt to make them sound like real drums, in which case you can get into an excruciating degree of detail. There's no plausible software at the moment that will automatically reproduce what a player does, because in the end it involves musical sensibility. You simply have to imagine what a real drummer would play here. But equally, using loops of real drummers means you have this embarrassment of riches — you can spend
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hours going through sounds and samples and beat, looking for the one that's right, and at some point you realise 'Hey, I'm making music here.'" The only non-virtual instruments Rundgren used on Liars were his guitar, the overdubs done by the two soloists, and his voice. For the guitar he used "some pedal boxes and software guitar emulation", whereas he recorded his voice with a Beyer MCD100 digital dynamic microphone. "It gave me a really good basic sound, and in addition I used the Antares microphone modelling software to emulate a Neumann 87, which is my preferred mic for my voice. In some cases I also used a 1176 plug-in, but no EQ. Ironically I couldn't get my authorisation to work on my Line 6, so I didn't use it at all."
Todd Rundgren created the world's first colour graphics tablet in 1980, and is still exploring the possibilities today, thanks to the Tablet PC.
Rundgren adds that he didn't use Reason to record audio, because "it's not a practical environment for recording. Instead I used Pro Tools and a combination of plug-ins. I also used Pro Tools for mixing, with the help of a Pro Control unit. I couldn't do it without a control surface. I need the feedback that a control surface gives you. Faders on a board give you a picture of the mix, without even hearing it. But I don't use any outboard gear any longer. All the outboard gear was in the computer, so the material never left the computer. I bounced it all down to one file. It's great to work like this, because it allowed me to put all my equipment in the front seat of a truck, and it was a couple of loads for one guy to carry and I'm set up."
No Secret Any More Given Rundgren's well-documented eagerness to be at the technological forefront, it appears obvious why he chose this way of recording, which is, after all, regarded as the future. But it turns out that there was also a very practical reason — for the first time since he founded Secret Sound, Rundgren has been without his own recording studio for a few years. After he moved from Woodstock, NY, to Hawaii in May 1996, he gave up his studio in Woodstock, but has so far been unable to find the ideal location for a new studio in Hawaii. "I haven't had my own studio for a while now, but we'll start building one this year. It'll be the first studio that I designed from the ground up, from an architectural point of view. Acoustics are a consideration, and also size and application. For instance, the studio will be able to do mixdown for film, and so it will have that slight wedge shape that a lot of movie theatres have, where it's larger at the back than at the front. I'll be able to set up orchestras on the steps as well. I've never
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had a studio that large before. It'll be completed in a year or two. It will be for my own projects, but it will also have some apartments attached to it so people will be able to come and do their projects there." The lack of a private studio space, as well as the experience of delivering his music song by song via the Internet, had some unexpected consequences for the way Liars was recorded. "It taught me something about the way I record most effectively," Rundgren concedes. "The loss of isolation in Hawaii made it more difficult for me to create the state of mind to be prolific. I've discovered that I'm at my most effective when I'm able to filter out the Since moving to Hawaii several years ago, noise of the world, so I can hear what I Todd Rundgren has not had a permanent studio, and has worked by setting up his actually think. My own thoughts will computer-based recording rig in various come to the fore, rather than my rented houses and holiday homes. forebrain constantly addressing distractions all the time. It's like I have to get in a larger sense into a meditative state so my subconscious can go to work. To create this album I ended up renting local vacation cottages and setting up my gear in those, so that I could have the peace and quite and isolation I needed. And when writing music for the Internet, I found that writing a song once a month isn't practical for me. Some months you're hot and some months you're not." It appears that, despite concerted efforts to get away from the conventional artistaudience relationship and the traditional music delivery format, Rundgren chose to return to these traditions, because he found that songs arrive in groups during creative peaks and that these songs tend to belong together. However, he says that he never longed for a return to a traditional 'real' studio: "Looking at a computer screen that has a picture of rack of equipment isn't any different than looking at a real rack of equipment," he laughs. "I always said that I consider the studio to be an instrument, a thing that requires techniques that you must learn, and that involves creativity in applying those techniques. That's no different than playing the guitar, or using a computer. You have to spend time learning it. In other words, in the argument as to whether it's valid or not to use virtual gear to make a record that's supposed to sound like live musicians... the only argument is: if you can do it, go ahead and do it! I'm still the one who figured out how to do it, and that's the same thing a musician has always had to figure out. "In any case, I've long been sick of tape-related disasters, waiting for tape machines to rewind, or the sticky shed. This is why I love the idea of hard disk recording so much. To be able to press a button and be immediately at your cue point... I just got tired of tape. After a while it made no sense to go back to the old way, unless you're some sort of religious fanatic. I think the argument has been around long enough to move on to the next argument, which is that it's no
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different for a guy programming something in Pro Tools than have an engineer record it 20 times. It's no different or less musical than, for instance, Mutt Lange making a rhythm guitar player sit in the studio for three days straight and play the same part over and over and over again until it sounds like a machine! Obviously, you can bless or curse all these new technologies, but in the end none of it matters, unless your music is affected in an incredibly negative way. In that case you have to find out why that is, so you don't do it again. Or you may find that it has affected your music in an incredibly positive way, and you want to find out how it's done, so you can repeat it." Which is, it appears, what Todd Rundgren has been doing for the last three and a half decades... Liars was released on Sanctuary Records on April 5, while BMG release The Todd Rundgren Collector's Deluxe Edition DVD on May 3. Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Your Letters
In this article:
Sounding Offended Kitten Conspiracy?
Your Letters Crosstalk Published in SOS May 2004 Print article : Close window
People
Sounding Offended Andy Searle-Barnes' Sounding Off piece in SOS March 2004 (www. soundonsound.com/sos/mar04/articles/soundingoff.htm) seems to have touched a nerve. In it, he argued that the 'DIY' recording boom is harming the industry, and that, for a serious project, a professional studio will always be able to do better job. Here are just two of the letters we received. Aah! Don't you just adore the voice of youth? Especially when the owner of that voice is possessed of a degree — a device which, experience has shown me, allows the owner to be wrong with absolute confidence! My own youth was spent in a time and place in which everything seemed to fight to suppress the desire to make music. There were no local studios and if you wanted to consider further or higher education in music you had the choice of classical or... forget it! For a number of years (I exaggerate not) I saved to gather enough money to pay to have some of my music recorded. On the eve of my first-ever recording session, I was offered a second-hand Tascam Portastudio for slightly less than I was about to pay to have a 'professional' job done. Thinking that 'pro is best' I declined the offer and paid for my studio time. My experiences of professional recording then and since have amounted to a lengthy and expensive catalogue of errors, including tapes wiped by accident, insufficient monitoring setups, mixes ruined by engineers deciding to 'try a few things' and duplicated tapes which ran nearly three semitones sharp.
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Your Letters
Though these guys probably had 'the best tools available and years of experience', not once have I enjoyed a satisfactory service in a studio so now, in my retirement, I am setting about learning to 'do it myself'. I am insulted by Andy Searle-Barnes' suggestion that because a job is not done by professionals then it is 'shoddy work'. I, like a great many other 'DIY' musicians, take great pride in everything I do and would never be satisfied with a 'shoddy' job. I think the main (though hidden) thrust of Andy's article is given away by his use of the phrase 'black art' to describe recording. This calls to mind what I call 'Witch Doctor syndrome'. I coined this phrase for the guys in the IT department where I used to work. They, like the witch doctor, knew that their power — their only power — came from the ignorance of those they serve/rule. Those IT guys were openly hostile to anyone with the temerity to question their 'black art' and were always keen to ridicule those seeking knowledge. I think the difference between a job that I do for myself and a job done by 'professionals' is that I am keen to complete the task to my own high standards, while my experience of professionals is that their only interest appears to be separating me from my money. Chris Marsh
Andy Searle-Barnes is critical of the current trend towards 'DIY' studios and home recording for two reasons: the disappearance of studio apprenticeships and secondly, inferior equipment and inexperience. These arguments do have their merits, but they aren't new. Similar arguments have been made in the face of changing technologies in many other creative disciplines. For example, the golden age of graphic design began in the '60s, and by the '70s typography had come of age. Professional typographers were specialists with years of experience behind them. But then came the personal computer. It was affordable and with software like Photoshop, the individual graphics designer didn't need a typographer. The lone graduate could produce flyers in his bedroom using professional-quality photographic effects without having to go to a typesetting bureau. Undoubtedly there are downsides too. In those early days, every Photoshop picture featured the 'Lens Flare' filter. It was graphic design's equivalent to overprocessing with EQ or reverb in music production. With time however, we grew up and adapted, just like we will grow up and adapt to all this marvellous, liberating 'DIY' technology.
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Your Letters
Many 'DIY' creators master in a pro studio (which is exactly what I do) with the aid of an experienced engineer. Pro studios have adapted to this function everywhere, and it is often more rewarding for the artist. My experiences in a band and in professional studios have shown me how difficult it is to accurately convey a concept to others. Most of the time you end up with a compromise, the incommunicable going by the wayside. Andy's fear of the watering-down of skills is well-founded, but the watering-down of skills has been in evidence since the invention of the drum machine, to a greater extent since the arrival of MIDI, and to an even greater extent since digital recording arrived. The loss however, is balanced by the invention of new techniques and musical forms, which in turn create new experts. Bayo Akinsiku
Kitten Conspiracy? Paul White's Leader columns in SOS February and March 2003 (www. soundonsound.com/sos/feb04/articles/leader.htm and www.soundonsound.com/ sos/mar04/articles/leader.htm), which revealed his personal battle with a pet cat bent on sabotage, also struck a chord... Sean Vincent from Computer Music here. I've just read Paul's editorial in the latest SOS and had to write in — for the second time in a month my cat has decided to piss on my toaster! Some kind of conspiracy maybe? Still, it made me chuckle, even if I do have to buy a second toaster in as many weeks. Sean Vincent
Electrical equipment beware — for danger lurks in the shadows...
I was interested to read about Paul White's problems with cat pee. I have suffered for many years with similar problems: smoking wall sockets and toasters stinking the whole house out. In fact, I do not have a toaster any more — I got through three of them and gave up. Citronella oil and a few drops of peppermint oil in a foil container on the exact spot — recycle your takeaway trays! — can help to break the cycle. I also have a spray comprising citronella and washing-up liquid for clearing up after a 'present' has been left. A nice piece of cultivated garden not too far from the house which they find comfortable to 'use' can be of great help too. Ken Hill file:///H|/SOS%2004-05/Your%20Letters.htm (3 of 4)9/22/2005 6:10:50 PM
Your Letters
Published in SOS May 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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Automatic Beat-slicing
In this article:
Automatic Beat-slicing
Logic Tips Preparation For Beat Slicing Logic Notes Cutting & Inserting Sections Published in SOS May 2004 In The Arrange Window Print article : Close window Using Strip Silence Technique : Logic Notes Have Your Say! Sorting Out Gaps & Overlaps
Current Versions Mac OS X: Logic Pro v6.4 Mac OS 9: Logic Pro v6.4 PC: Logic Audio Platinum v5.5.1
Although beat slicing is something more immediately associated with programs like Propellerhead Recycle or Bitshift Audio Phatmatik Pro, Logic also has facilities to chop up your loops automatically. Len Sasso
Unless you've been on holiday for the past decade, you're undoubtedly aware of the technique of slicing an audio file into segments in order to change the tempo of the audio without changing its pitch. Programs such as Propellerhead's Recycle, Bitshift Audio's Phatmatik Pro, Sonic Foundry's Acid, and Ableton's Live accomplish this with varying degrees of user intervention. With a little effort, you can accomplish the same thing completely within Logic using the Strip Silence function. There are several digital signal processing (DSP) techniques for changing the length of an audio file while mostly preserving its pitch and timbre. Granular synthesis and analysis/resynthesis are two common methods. If time-stretching is your goal, you can use Logic's Time The Strip Silence window's Threshold and Machine, either in the Sample Editor Minimum Time To Accept As Silence window or directly in the Arrange parameters have been adjusted so that most window, more easily than the eventof the events are caught in individual slices, slicing method I'll describe here. but no single event is sliced in two. However, audio-file slicing allows you to do a number of things beyond simply changing tempo. Those include rearranging the order of events, applying different DSP processing to individual events, and replacing some events with other recorded or synthesized sounds. You can also load the individual slices into a sampler such as Logic's EXS24 and play them from a MIDI keyboard or
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Automatic Beat-slicing
MIDI sequence. Although there are several ways to approach slicing up an audio file, the underlying concept is the same in all cases. The duration of an audio file, and therefore its tempo if it is a rhythm-based file such as a drum loop, is fixed. If you slice the file into separate regions, usually corresponding to individual sound events (for example, kick drum hits) and leave the regions at their original beatpositions on Logic's bars/beats Time Ruler, you can freely change Logic's tempo without affecting the rhythmic relationship between the slices.
Logic Tips The channels displayed in the Track Mixer window reflect the arrangement of tracks in the Arrange window. You can therefore use empty Folders with different Track Lists to access different Track Mixer views. If you put the Track Mixer in Contents Link mode, simply selecting the Folder will change the Track Mixer view. Len Sasso When you're using multi-channel audio plug-ins, you need to set the Audio Instrument object's MIDI Cha parameter to All in order to play the instrument's different channels with MIDI data on different MIDI channels. Len Sasso
Preparation For Beat Slicing Several preliminary steps are helpful in getting the slices automatically placed at the correct Time Ruler positions in Logic's Arrange window. In the Sample Editor window, create a region within the audio file that starts and ends on a bar line and includes all the material you wish to slice. Then drag that region to the desired starting bar on an audio track in the Arrange window. Next, set Logic's left locator to the beginning of the region and set the right locator to define a bars/ beats Time Ruler selection which matches the region's length in bars. (Note that, unless Logic's tempo already matches the audio, this will not correspond to the actual end of the region.) If you ultimately wish to preserve the Song's tempo, select an unused Tempo Alternative using the Options menu in Logic's Tempo List window. Finally, select the region and choose Adjust Tempo Using Object Length & Locators from the Tempo submenu of Logic's Options menu. Your region will now fit perfectly between the locators, and will occupy the correct number of bars. The quickest, though not necessarily the most useful, way to slice up the region is to use the Scissors tool while holding the Alt key. Clicking at the desired slice-length from the left region boundary (one beat for example) will slice the region into equal-sized pieces. That may work well for precisely-played or quantised material, but in general the slices will bear no relation to the musical events within the audio file. More intelligent slicing, similar to that done in Recycle and Phatmatik Pro, can be accomplished with Strip Silence.
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As its name suggests, the purpose of Strip Silence is to break an audio file into regions by removing the silence between them. It does that by searching for segments within an audio file that are below a specified threshold level for at least a specified minimum amount of time. It then splits the file into the regions between the 'silent' areas. That process is very handy for extracting loops and hits from audio sampling CDs, for example, where a track might contain several loops or many individual drum hits separated by silence. Beat-slicing programs work on a slightly different principle, detecting individual events by their attack transients and slicing the file up into the detected events. You can use Strip Silence to do much the same thing.
Cutting & Inserting Sections In The Arrange Window One job that's particularly tedious when using Logic's basic edit commands is the process of deleting a few bars from the middle of a Song and then closing up the gap. You can audition with a skipped section easily enough by marking a Skip Cycle in the Time Ruler at the top of the window (this shows up as a narrow white bar when you drag from right to left), but what's the fastest way to make this permanent? It turns out that the exact command you need is available under the Arrange window's Functions menu. The Cut/Insert Time submenu has a Snip: Cut Time & Move By Locators option which works on the area of the Song between the two locators, though it also seems to work fine if you've set the locators up to do a Skip Cycle for auditioning purposes. However, you do have to remember to Select All before doing the operation, otherwise only the selected tracks will be affected, and most of the time you'll want to do it to all the tracks. Once you've carried out the Snip command, everything between the locators is removed and copied to the clipboard, and the end of the Song is moved back to fill the gap — easy! Furthermore, if you need to take the clipboard contents and insert them elsewhere in the Song, all you need to do is put the Song Position Line where you want the insert to start, Select All again, then use the Splice: Insert Snipped Part At Song Position option from the same submenu. When carrying out these operations, it is worth knowing that any tempo changes, time-signature changes, and global Score-window symbols are also moved. If you don't select all objects at the edit point, Logic will open a dialogue box asking whether tempo changes and global score symbols should be affected or not. Any looped objects affecting the edit will be turned into real objects as required, but be aware that any frozen tracks will remain exactly as they were, and no warning dialogue will be issued. Paul White
Using Strip Silence You invoke Strip Silence by selecting the unsliced audio file and choosing Strip Silence from the Arrange window's Audio menu. (You can also invoke Strip Silence in the Sample Editor window, but for our purposes the Arrange window is file:///H|/SOS%2004-05/Automatic%20Beat-slicing.htm (3 of 6)9/22/2005 6:11:18 PM
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the place to be.) For practice, start with audio material which has clearly defined events — a basic drum loop, for example. The Threshold and Minimum Time To Accept As Silence settings are the most critical, so start by setting the Pre Attack Time and Post Release Time to zero. You'll notice that the number of regions varies inversely with Minimum Time To Accept As Silence (ie. a higher setting results in fewer regions) — a few 100ths of a second is usually a good starting point. You'll also notice that the number of regions at first increases as you raise the Threshold setting, then begins to decrease as the threshold gets higher. Depending on the material, settings between five and 15 percent tend to produce the best results. The goal in adjusting the Threshold and Minimum Time To Accept As Silence settings is twofold. You want to get enough slices to separate most of the events without also slicing up individual events. Having some slices capture more than one event is tolerable, because you can fix those manually or by invoking Strip Silence again on the errant slices. Over-slicing is more of a nuisance, so err on the side of fewer slices.
Here the tempo has been increased, resulting in overlapping of the events produced by Strip Silence. Alternate events have been moved to a second track to make setting the fade-out times visually easier.
The Pre Attack Time and Post Release Time settings allow you to extend all of the slices to the left and right respectively. Leave Pre Attack Time at zero and adjust Post Release Time until there are no gaps between the slices. That gives you back all the 'silence' that would otherwise have been left on the cutting-room floor, without changing the number of slices or their positions. After you click the OK button, Strip Silence will slice the file into individual regions residing at the Time Ruler positions they originally occupied in the file. As previously mentioned, you will need to apply Strip Silence again or manually split those regions (if any) that contain multiple events. If you open the Audio window, you will see that the regions exist there as well, and you can use the Audio window's Audio menu to save the regions as individual audio files if so desired. (You might want to do that in order to use the regions in a sampler, for example.) If your goal is simply changing tempo, you can now return to the Song's original Tempo Alternative and play the track to see how it sounds at the original tempo. If the difference in tempo is small, you may not need to make any adjustments, but the chances are that some tweaking will be in order. Since the regions are tied to the bars/beats Time Ruler, increasing the tempo reduces the time between regions, whereas lowering the tempo increases it. Put another way, increasing the tempo causes the regions to overlap whereas decreasing it produces gaps between the regions.
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Have Your Say! If you want to suggest changes or improvements to Logic, then here's your chance! The Emagic development team are inviting SOS readers to send in their suggestions of what they'd most like added or changed in Logic. Email your top five suggestions (in order of preference) to
[email protected], and we'll forward your lists on to the Logic team. We'll be asking them for feedback on which changes users deem most important and how these might be addressed.
Sorting Out Gaps & Overlaps When increasing tempo, if the overlapping is not too severe, you can compensate in two ways. The simplest is to select all the regions and choose 'Tie Objects By Length Change' from the Object submenu of the Arrange window's Functions menu. That shortens the regions as necessary to make them fit end-toend in the allotted space. If cutting off the end of each region produces unsatisfactory results, try fading out the regions rather than resizing them. To do that it is convenient to move every other region to an adjacent audio track. That allows you to visually judge the amount of fade required. Next select all regions on both tracks and input a Fade Out value in the Parameters box so that you fade out most of the overlaps. If the regions vary greatly in size, you may need to adjust some individual fades as well. About the only way to adjust for gaps when reducing the tempo is to add a little reverb to the track. If that turns out to be unsatisfactory, you will need to use a program such as Recycle that will fill out the slices using more sophisticated DSP processing. There are a number of things you can do with the regions you've created beyond simply altering their timing. For one thing, you can allocate them among several audio tracks (retaining their time positions) in order to apply different plug-in processing. For another, you can replace some of the regions with other audio clips of similar or completely different sounds, for example replacing one kick drum with another, or replacing a hi-hat with a ride cymbal. You can also easily create a MIDI file corresponding to the timing of the slices, to be used as a Groove Template or as the rhythmic basis for another part played on a totally different synthesized or sampled instrument. Published in SOS May 2004
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Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email:
[email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895
All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved. The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the publishers. Web site designed & maintained by PB Associates | SOS | Relative Media
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
In this article:
First To 6000 Made In Britten Close & Distant Pad Power In-ear Monitoring One Drum At A Time Taking It All Off More Sex Cooking At Last
CLASSIC TRACKS: 'What's Love Got To Do With It?' Artist: Tina Turner; Producer: Terry Britten; Engineer: John Hudson Published in SOS May 2004 Print article : Close window
Technique : Recording/Mixing
In 1984, a dose of British soul resurrected Tina Turner's flagging career in spectacular style. For engineer John Hudson, the recording of 'What's Love Got To Do With It?' also provided a memorable example of the 'less is more' principle in action... Richard Buskin
In 1984, after appearing to be completely washed up as a mainstream recording artist, 45-year-old soul dynamo Tina Turner made one of the great comebacks in the history of the music business. Having signed a solo deal with Capitol and enjoyed a Top 30 US hit with a sultry cover of Al Green's 'Let's Stay Together', she then struck paydirt with her next single, 'What's Love Got To Do With It'. Written by Terry Britten and Graham Lyle, it topped the American charts for three weeks and, courtesy of Britten's slick production underpinning the singer's passionate and sometimes plaintive delivery, helped create what the New York Times described as "new blue-eyed soul from England". The accompanying album, Private Dancer, spawned two more hit singles: the title track, penned by Mark Knopfler, and 'Better Be Good To Me'. However, if any one song marked the turning point in Turner's career, it was 'What's Love', and for this she owed a lot to both engineer John Hudson and Terry Britten, the seasoned session guitarist who not only co-authored the number, but who also made all the right moves when crafting both her vocal performance and the arrangement.
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
First To 6000 Commencing in 1972, John Hudson's career has seen him record, produce and/ or mix projects by artists ranging from Rod Stewart, Jeff Beck, Elvis Costello and Judy Tzuke to A-ha, Ultravox, Cliff Richard, Gary Glitter, Wynonna Judd, Kylie Minogue, the Moody Blues and, most recently, Herbert Gronemeyer. In 1980, John and wife Kate launched Mayfair Studios, which moved to its current Primrose Hill location in North-West London the following year. Since then it has evolved from a single SSL room to a six-studio state-of-the-art complex that, in addition to some of the aforementioned acts, has hosted Coldplay, Robbie Williams, Tom Jones, Blur, Bon Jovi, Radiohead, Suede, Manic Street Preachers, Bryan Adams, Annie Lennox, David Bowie, George Michael, Travis, Pink Floyd, the Pretenders, the Jesus & Mary Chain, Tears For Fears, Kate Bush and Björk, among many, many others. Hudson was in the middle of engineering some Ultravox sessions when the call came to work on a trio of Tina Turner tracks over the course of just a few weekends: 'What's Love Got To Do With It', 'I Can't Stand The Rain' and 'Show Some Respect'. Working with a 48-channel SSL 6000E console, Urei 813A main monitors and a pair of Studer A80 MkIV 24-track machines, he and his studio were more than up to the challenge. "That desk was the first 6000 made by SSL," Hudson recalls, "and the brilliant thing about it was that it housed the integral Adams Smith synchronizer control panel. It was fantastic, and we were also one of the first studios to have the tape machines in a separate room. That threw a lot of people. In those days it was a bit of a revelation for the guy who sat at the desk to also be operating the machines. If you had two 24-tracks running in sync — and sometimes there were three machines, as we had set up the console with three lots of 24-track ties to the machine room — you could disconnect either one, make either one the master or put in offsets, all from the console. You didn't have to go into the machine room at all. Of course, there were a lot of bugs with that system, and we had quite a few teething troubles, but they didn't affect the sessions."
Made In Britten Terry Britten had got the writing assignment after sending demos of the three songs to Tina Turner's manager, Roger Davies. These had been recorded to a two-inch machine in his small home studio, and featured guitars, Linn 2 drums and his own voice, together with keyboards played by Billy Livsey. "The demos were very well recorded and mixed by Terry," says Hudson, "and so the idea was to replicate many of the parts. For 'What's Love', Terry set the Linn 2 to '95' — to use it as a 'click' for building the track until we put on real drums —
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
and we ran the machine and recorded it onto three tracks: the kick onto one track, the snare onto a second, and the rest onto the third. You couldn't overdub the Linn, because back then you couldn't run it in sync. Terry then recorded a guide bass guitar and guitar parts sitting in the control room, and he also played a recurring harmonica-type pattern on guitar and laid down a guide vocal. "I remember for that bass part Terry didn't even tune the guitar. It was a DI'd Fender Jazz. I ran the tape, he played along, he thought it was good enough for now, and we ended up using it. There were a couple of times when he decided to redo it, and he'd spend four or five hours just playing away — seeing that we had a drum slave and a guitar slave, we had loads of tracks to play with and we could therefore keep everything. However, when it came to the mix, I told Terry that, in my opinion, his guide bass was the one to use, and after listening to it, he agreed." Tina Turner and John Hudson at
Playing a Fender Telecaster that was Mayfair Studios. alternately DI'd or fed through a Vox amp and Fender Champ miked with a combination of Shure SM57 and SM58 and AKG C12A mics, Terry Britten tracked quite a few guitar parts for 'What's Love' — varying the style, content and tuning — before choosing what worked best. And what did work included a reggae-flavoured part that came to the fore during the chorus, providing the song with its rhythmic backbone. "Although, at this point, Terry regarded the bass as a guide track, he wanted his plucky and damped guitar parts to be the finished thing, and to get them spot-on with the drum machine," Hudson says. "That way, once the real drums were added, the whole thing would be really tight. I mean, the bass was pretty tight as well — we were just worried about the tuning and whether or not he had played the right notes. Nothing he ever played was scrappy. "Since we wanted everything stereo, we recorded all the guitars in stereo, and a double-tracked, damped guitar might use anywhere up to four tracks for a single part. Then, if we tracked it, that would take up eight tracks, and at some point I'd have to bounce it down to a couple of tracks, but that didn't happen until the mix. Instead, to keep my options open, I would record a chorusy-type effect and keep that separate. In the '80s, because we had plenty of tracks, we'd usually record the effects on separate tracks."
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
Close & Distant To commit Tina Turner's voice to tape, Hudson had settled on a two-mic setup, one positioned close to the singer, and the other about four inches further away. "Years before I had done a couple of live recordings with Gary Glitter, who had a very dynamic voice," he explains. "He'd really let it rip and give a huge yell, and the way I coped with that was to have one mic really close and another further away. I mean, the way that a cabaret singer moves the mic away from his mouth by several inches whenever he sings a loud note is the worst thing you can possibly have in a studio environment, because the whole sound changes. Instead, you should set up two mics the same few inches apart and tell him to not move. "For some of the low notes in the verses, I already knew that when you sing softly, close on the mic, and you hit a note that's going towards the bottom of your register, it's going to get boomy, a bit chesty. However, if you put up a second mic slightly further away, you can crank that up when the person sings low notes softly. That brings the presence back to the low note, and you can either roll off the bottom on the close mic or you can pull it down a bit. Usually you can do that just by having a couple of compressors correctly adjusted and automating that effect — the other mic comes up and it sounds like the person's right in your face. Then, when he or she lets rip, the close mic compresses out like crazy and, of course, you don't get the woofy sound that usually occurs when someone's singing quietly. "That's what I did with Tina, close-miking her with a valve Neumann U67 while the distant mic, an AKG C12A, was there to pick up the loud parts. However, the C12 was overloading. She was so loud, it was unreal — we had the doors closed and they could hear her in the reception area! I had never experienced anything like it. I was absolutely staggered, and I could tell that Terry was in a bit of a panic. He said, 'It's distorting! It's distorting! Quick, quick! Fix it! Fix it!' Any valve mic that didn't have an attenuator on it couldn't cope, so what I did was put up a PZM, because I knew you could stick one of those on the side of a bass drum and it wouldn't distort. In this case it sounded a bit thin, but at least it was able to cope, and on the loud bits it was great when combined with the 67." Thanks to the double-miking technique, very little vocal treatment was required at the mix. "I never compress a vocal very much during the recording, because I'll be knackered when I'm mixing it," says Hudson. "I won't have any scope. At the same time, you can't experiment on the singer, because he or she might run out of steam before you've even got the sound. With 12 years of recording experience already behind me, I knew that my first priority was to get Tina's performance on tape with an OK sound and leave as much undone until the mix, when I could sit there all night without taking up her time. "I used a [Teletronix] LA2A for a little compression on the 67, but I didn't compress the loud [ie. distant] mic at all. If someone's got a really good voice and shouts something out, it shouldn't be compressed. I don't care what anybody says, even a Fairchild will change the sound and pull it in like a clenched fist. It won't have the freedom and the air that you want. That's why it was important to set up the PZM a few inches further away and not compress it, but just record it at a lower level. Turn the mic gain down so that most of the time it isn't peaking, and then when there's a shout it goes up to zero. Mixed in with the other mic, it sounds fantastic."
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
Pad Power Meanwhile, for the keyboard parts, Nick Glennie-Smith used a rig consisting of an Emulator II, a Roland MKS80 Super Jupiter, the new-fangled Yamaha DX7, and Terry Britten's Oberheim OB8. With Tina Turner scheduled to record her vocals the following Sunday, Britten wanted to complete the basic backing tracks for all three songs during that first weekend and the Saturday that followed, so all of the parts were laid down fairly quickly. "When there's only a basic rhythm track, you normally put a pad on before someone does a vocal, even if you remove it afterwards," John Hudson states. "It makes the track more comfortable for the singer to pitch to than if there are just these guitars with plenty of space in between. What's more, Terry had never worked with Tina before, and so he didn't have a clue as to whether she was going to come in and just rattle off each song, or whether she was going to do things verse by verse or line by line. That's not how he intended it to be, but he had to wait and see. "As it turned out, when Tina came in and did her vocals on that second Sunday, I was blown away. Beforehand I thought 'OK, here's this artist who had some hits years ago and we've never heard from her since. I hope she can bloody sing or we'll be here all bloody night!' I was doing another session with Ultravox at 10 o'clock on Monday morning, so the last thing I needed was an all-nighter before that. We'd have to complete three songs, and at that point 'What's Love' hadn't even been selected as a single. Tina didn't like it, and she, like the rest of us, was probably assuming 'I Can't Stand The Rain' would be released, because it was a really rocking track. So, Terry decided to start things off with 'What's Love' as that was the one which was going to be the least demanding on her voice. He'd get it out of the way while hearing how she could sing. "When we first put the track up, Tina sang like she was on stage in front of 25,000 people, belting it for all she was worth, and Terry was sitting there shaking his head, clearly concerned that this was going to be a disaster. He said 'Tina, you just need to back it off a bit,' and we spent ages trying to get her to do that, but she didn't appear to know what 'back it off' meant. She was in the studio, Terry, myself and the assistant engineer were in the control room, and for a time it was very, very tense. She began getting annoyed, saying 'I sing everything like this!' to which Terry said 'But this is a love song,' and he went into the studio and sang along with her to illustrate how it should be done. She liked how he'd sung on the demo, and he said 'Well, if you want it to sound like the demo, you can't belt it.'"
In-ear Monitoring
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
With his back virtually pinned against the wall, the producer had to think fast about how to encourage a little light and shade. "Nothing sounds good cranked full up," asserts John Hudson. "If you want something to sound really loud and powerful it's got to be below 90 percent, because otherwise you've got no leeway for those extra little spikes, whether they're vocal or guitar. In the end Terry told Tina 'Forget about the microphone. Imagine you're singing only to me, not an audience. Sing into my ear.' And when she did that, she sang quietly and I was going 'Yes!' I pushed the talkback and I said, 'Terry, that sounds great,' and Tina said 'Oh, whaddya mean, whaddya mean?' She didn't really understand what was going on. Terry just stood there with his ear by the microphone and said 'Right, sing the first verse,' and by the time we'd recorded the verses with him standing there with his ear in her face, she had totally got the hang of it. At that point, she was so pleased. She thought it was great, and I think it was really good that we did 'What's Love' first, because it put her in the whole mode of not giving 110 percent for every bloody note. "As soon as she got a handle on it, she was brilliant. I mean, she locked into it within a couple of hours. It wasn't a case of playing it back and saying 'Right Tina, that was great. Do it like that.' There wouldn't be any point. She's a performer. She was relying totally on Terry and me, and once he'd sussed the way to do it, Terry was blown away by the results. He was going 'Oh, this is fantastic.' Her vocal John Hudson (left) and Terry Britten share a was so smooth and powerful, and joke during the 'What's Love' sessions. Terry was absolutely brilliant to get that performance out of her. Still, doing the ad libs was really funny, because she'd never do them in the right place. "A lot of great singers aren't arrangers. Again, they're performers, so you don't want to stop the tape or say 'I'll give you a couple of bars to run up and do a "whoa-whoa" after the chorus.' You can't do that. Instead, Terry had to say 'Don't sing the verses, just sing the chorus, and then do some ad libs when you get to the end.' That way, although some of the ad libs were in the wrong place, we'd get a fantastic performance and so we could just move them. As we didn't have samplers in those days, we'd move the ad libs by offsetting the two tape machines. We'd have a click on both tapes, the guide mix on the second tape, and if, say, she did an ad lib two bars late, I'd just trim the offset until the clicks were in sync and then bounce the ad lib across. That's how we did all the ad libs."
One Drum At A Time After Tina Turner had recorded her vocals for 'What's Love' — later compiled by Hudson from a half-dozen takes — and the other two songs, drummer Graham
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CLASSIC TRACKS: 'What's Love Got To Do With It?'
Jarvis entered the picture. He listened to a drum-machine reference track on the drum slave to get the timing and then set about trying to recreate what he heard on the 'What's Love' demo. However, in those pre-Pro Tools days, this was no easy task. "Doing drums, you could drop in, but of course, because of the separation, you couldn't drop in just on the snare track," says Hudson. "Thinking ahead, Graham agreed to record everything separately so we'd have total separation. Well, we did record the cymbals in one pass, the bass drum and snare in another, the hihat in another, and also the tom-toms in a separate pass. You see, we wanted to get a very close snare sound, so we didn't want a lot of mics open. I've never used gates, because to my mind you cannot gate a drum kit and get it sounding fantastic, and you also might miss a soft hit and not notice this until the mix." Once the drums had been recorded, Fairlight strings were then added, Simon Morton played percussion, Tessa Niles contributed backing vocals, and several hours were also spent superseding the harmonica-like guitar pattern that Britten had recorded on the demo with real harmonica. This was played by session man Nicky Payne, who had previously fulfilled a similar role for the likes of Culture Club and Paul Young.
John Hudson and Mayfair Studios today.
"He performed it in different pitches," recalls John Hudson. "Up the octave, down the octave, tracking it and trying every combination he could think of, and it sounded fantastic. We then did a rough mix of the whole thing — because we only ever did one mix, we never did a remix — and gave it to Roger Davies, but when it came back there was a note saying 'We don't like the harmonica, we like that sound on the demo.' Terry was going 'Oh, bloody hell,' and we got Nick back in to replicate that sound, but he couldn't do it, so Terry then said 'I'm going to phone Billy Livsey and see if there's any chance he can come in for a couple of hours and do it.'" And that's what happened. Initially trying to recreate the pseudo-harmonica part by utilising the breath controller on Nick Glennie-Smith's DX7 — a mouthpiece that plugged in and controlled the volume of each note when blown into — Livsey subsequently achieved the desired result by using the DX7's harmonica patch and tweaking it to sound like something more closely resembling a flute.
Taking It All Off Thereafter, it was time for the mix, during which the modus operandi would consist of John Hudson being left to his own devices and Terry Britten file:///H|/SOS%2004-05/CLASSIC%20TRACKS%20%20...%20Love%20Got%20To%20Do%20With%20It%20%27.htm (7 of 11)9/22/2005 6:11:26 PM
CLASSIC TRACKS: 'What's Love Got To Do With It?'
periodically gauging the results and giving his feedback. "I must have spent an entire Saturday morning bouncing things across from all the slave tapes, getting everything onto two 24-track tapes for 'What's Love'," Hudson remarks. "Then I started to get some sounds up; a basic rhythm-section sound as well as the vocal, really low just to hear the tune and see where it was going. Once I'd set up the mix, Terry then phoned late on that Saturday to ask how things were going and I told him 'It actually sounds pretty good,' which was Hudson-speak for 'incredible'. I said 'It's really rocking. In fact, when you get to the chorus, it's like "wow"! It's just brilliant.' He said 'Well, do you want to spend a bit longer on it?' and I said 'Yeah, I wouldn't mind that. If I work until midnight you can then come in tomorrow morning and have a listen.' "Everything was coming together, there were all these effects, and I was thinking 'Ohhh, this is the best thing I've ever done! It's bloody amazing!' I was so excited about it, I could hardly sleep. I checked it before Terry arrived in the morning and it was all happening, and when he came in he said 'Don't play it too loud. I don't want to hear it blasting. I want to get a perspective on it.' So, I played it while he stood in front of the console, and as A wide shot of the Mayfair control room, taken during the recording of Tina Turner's it was automated I just went and follow-up album Break Every Rule. leaned against a wall in a corner of the room, which is what I normally do in order to get my own different perspective. Anyway, there I was, thinking 'This is pretty bloody good,' and Terry meanwhile was standing completely motionless and without any expression on his face. Then, when we got to the end and I stopped the tape, he still didn't say a thing, and what's more, he wasn't smiling. "I was thinking 'Bloody hell, what's wrong? It sounds amazing,' and after several seconds he finally said 'Ah, that's not right.' Well, I nearly murdered him — I honestly could have grabbed him around the throat — and I felt like I was going to explode. I said 'What?' and again he said 'It's not right.' Outwardly I was quite polite, but I was absolutely seething. I was trembling, I could hardly contain myself. As far as I was concerned, he had given me all these options, allowing me to make the decisions, and now, after hours of bouncing things down, level tweaking, comping the congas, comping the backing vocals, setting up loads of effects and so on, he was suddenly saying, 'It's not right!' "Obviously, Terry could see I was extremely upset, and he's such a nice guy; he's not the sort of guy who'd get a kick out of putting someone down. However, he pulled a cassette out of his pocket and he said 'Just play this.'"
More Sex file:///H|/SOS%2004-05/CLASSIC%20TRACKS%20%20...%20Love%20Got%20To%20Do%20With%20It%20%27.htm (8 of 11)9/22/2005 6:11:26 PM
CLASSIC TRACKS: 'What's Love Got To Do With It?'
What Britten handed Hudson was a cassette of the monitor mix from the day they'd recorded Tina Turner's vocal onto a basic track consisting of the Linn Drum, guide bass, some picky rhythm guitars and a rough synth version of the harmonica part. No backing vocals, no synthesized strings, nothing else. "We put that on and listened to it all the way through," Hudson recalls, "and afterwards Terry said 'That's it. That is it.' It was such a shock to listen to a monitor mix with virtually nothing on it after we'd just spent four weeks burning this huge track with everything under the sun — the change was so drastic that I was having trouble getting my head around it. When you work with producers, the main thing is trying to figure out what they're talking about. Like when I was working with Imagination and the guy was going, 'More sex! More sex!' Which knob do you turn to achieve that?
The SSL desk at Mayfair Studios was the first 6000 made, and the sessions for 'What's Love' exploited its features to the full. This photo shows John Hudson working on the mix.
"Therefore, when Terry played the cassette and said 'That's it,' I was wondering if he meant those precise instruments. I mean, he didn't really know either, but after I'd calmed down he said 'It's just too much. Play the vocal and the rhythm section.' By then I had cancelled the mix and I'd kind of recovered. I'd stopped trembling. So, we pulled most of the faders right down, including most of the overdub tracks — all of the strings and backing vocals — and retained just the rhythm section, more or less recreating what Tina had sung to on that first day. She hadn't sung to all this other stuff. And that's another thing I've learned since then: if you add or change things after the main vocal
