Q. What is a Ground (Earth) Loop?

By Hugh Robjohns

 
Lifting the mains earth like this could lead to serious injury.

Years ago I touched a guitar that was plugged into an amp against a radiator, and it blew the amp and melted the guitar string. Afterwards someone told me that this was caused by a ground loop, but I've never actually understood what that means.

SOS Forum Post

Technical Editor Hugh Robjohns replies: That wasn't a ground loop — that was a faulty amp with a missing safety earth. There was no loop, because there was no earth at all until touching the radiator provided the missing link. Had you sat on the radiator and then picked up the guitar, you might well have been playing Emaj7 on a harp on a fluffy white cloud by now!

This is a distressingly common and life-threatening situation often caused by guitarists (or their so called 'technicians') in a futile effort to stop audible hums.

A ground loop is different — it occurs when there is more than one ground path between two items of equipment. Usually, one path is the screen of an audio cable connecting the two pieces of equipment and the other path is via their chassis safety earths in the mains plugs. Inside the equipment, the audio screen earth is often linked directly to the chassis earth, hence the possibility of a loop. If the two bits of equipment are plugged into the same mains socket, their chassis safety earths are effectively tied together at the same potential, and so there is unlikely to be any circulating ground current, despite the apparent ground loop. However, if one item is plugged into a different mains socket, its chassis safety earth might be grounded some way away from the other equipment's earth, and there can be a small difference in potential voltage between them. Silly as it sounds, earth is not the same everywhere. The potential voltage difference between their two chassis earths can cause a small current to flow, and since the earth provides a reference for the audio electronics, that flowing current causes the earth reference voltage point to vary slightly. This can be heard, usually as a low-level hum or buzz.

Ideally, the solution is to make sure that everything is earthed at one central point, so that everything shares the same common earth reference point. The easiest way to do that is to plug everything into a star arrangement of plug-boards fed from a single socket (assuming suitable power capacity). If that can't be done, the safest solution is to break the loop by isolating the audio cable screens at one end. The cable is still screened, but there is no longer any possibility of a loop, so the hum currents can't flow around it. Inserting transformers in the signal path can also break the loop, and this solution is common in outside broadcast and live sound rigs. DI boxes feature transformers for this purpose too.

Problems arise when uninformed people decide to break the loop by removing the safety earth in the mains plug instead.

This does break the loop, obviously, so any related hum will disappear. But it also means that the equipment is no longer earthed, and thus any fault that occurs in the equipment is now life-threatening! Sometimes you don't even need a fault to cause dangerous problems, though. Most equipment has filtering on the mains input to stop mains-borne noise getting in (or out). If you disconnect the mains earth in the plug, the nature of that filtering is such that the (previously earthed) chassis — and everything connected to it — 'floats' up to half mains voltage (making it about 115V in the UK). This means everything that should be safely earthed — all the exposed metalwork, including guitar strings — now carries a life-threatening voltage.

Going back to your guitar incident, the strings on the guitar are supposed to be earthed through the guitar lead to the socket on the amp. That, in turn, is usually connected to the amp's chassis earth, and thus through the mains plug to the mains safety earth. Metal radiators are connected to the mains safety earth point too, as is all house plumbing.

So if you have a guitar amp with the safety earth disconnected in the plug, the chassis is likely to rise to 115V, and everything that should be at 0V (earth) is now cooking with gas! Rest the guitar strings on anything that really is earthed (the radiator in your example, but a mic stand holding a mic that is earthed through its cable to a mixer is another very common alternative), and you have mains power now flowing directly through the equipment to find a real ground. This is almost certainly going to severely damage or destroy the amp, the guitar and — if you get yourself between the two — you as well! The classic way for budding pop stars to die is because of an amp with the mains safety earth disconnected. They rest one hand on their guitar strings and reach up with the other to hold a properly earthed mic. That leaves them with 115V effectively straight across their heart. Sweaty hands make very good conductors and it only takes a few milliamps of current flow to stop the heart. Musicians are electrocuted in this way every year, and while I'm all in favour of Darwinian evolution, sometimes the good guys get caught out too.

The lesson is that you should never remove the earth from a mains plug. If you have hum problems, break the screens on the audio cables or use isolating transformers in DI boxes.

Published August 2006

OKGO Contest Announcement

Flightcasing And Protection

By Mike Crofts

 
A metal 19-inch rack case safely housing a set of radio-mic receivers.
Photo: Mike Crofts

Stage gear takes a lot more punishment than studio equipment, so investing a few quid in suitable protection for it has to be worth considering. But what do you need to know before buying?

Once you've spent hard-earned cash on equipment for live sound use, it's the generally accepted wisdom that it will need protecting against bumps and scrapes along the way. Captain Kirk could look after the Enterprise by simply saying 'raise shields' but for the likes of you and me, it's 'flightcase'. You might think that the cost of flightcases to properly protect your valuable gear against damage will be quite high (although cases don't actually cost as much as you might expect). However, investing in suitable cases not only offers protection against damage, but has other 'on road' benefits too, in terms of added convenience and flexibility, not to mention the professional look it gives you.

If you try a Google search on 'flightcase', about two million results will be returned, the first page or so of which are likely to refer to flightcases for the entertainment and corporate advertising industries. No matter what the object is, the chances are that someone, somewhere makes a flightcase for it, or is quite willing to do so if asked. In the field of live sound, we need cases in which to hold, operate, store and transport our equipment. The degree of protection required and the overall design and quality of the case will depend not only on the gear itself but also on the intended or potential use. When I started out I couldn't afford much in the way of decent sound gear, let alone cases to put it in, but as I've built up and improved my PA inventory I've tried to ensure that it is packed and protected as well as possible. In choosing which cases are the best for your needs, don't forget that they will add to the overall size and weight of the gear you need to move around; many full flightcases are heavier than their contents, so you may need to consider the individual weight of each item, and perhaps even the load capacity of the vehicle you use to transport your gear.
 
This rack case lid incorporates a weatherproof sealing strip.Let's have a brief look at the cases commonly used for live sound gear, and consider one or two factors along the way.

Equipment Mounting Cases

Generally known as 'rack cases' or 'rackmount cases', these are usually made to accommodate standard 19-inch gear, which is permanently fixed into the case and is stored, transported and operated in situ. Cases of this type usually have removable front and rear doors, with a centre section that contains the rack gear itself. A few different types are available: you can obtain the traditional plywood cases with aluminium edging and heavy-duty butterfly catches, or there are lighter, moulded cases that are suitable for outboard processors as well as heavier items such as amplifiers. One issue I have with moulded cases is that they often have the rackmounting strips at the front edge of the case, which means that control knobs will protrude and will be exposed when the front cover is removed. The more traditional cases usually mount the gear a little further back inside the casing.

For especially delicate gear, 'sleeved' rack cases are available. The 19-inch rack holding the equipment fits inside an outer casing, or sleeve, and between the two parts is a layer of foam to cushion against impact and vibration (ever sat on the floor in the back of a Transit van when it's negotiating a bumpy road?). Any equipment containing moving parts, such as CD and Minidisc players, or anything containing a hard drive, would undoubtedly benefit from this type of protection.
 
A cable trunk. It's amazing what you can fit in when you try...

Prices for rack cases range from around £90 for a budget 4U traditional board case, through standard board and lightweight polyethylene 4U cases at between £90 and £125, to high-protection sleeved 4U cases at around £150. Bear in mind that the larger the case gets, the cheaper it will usually be per 'U'.

Keeping The Gremlins Out

Although rack cases are usually built with a sort of aluminium tongue-and-groove arrangement around the lid edges, this is primarily to locate the lid securely and not for environmental protection, unless the case is specifically designed to provide this. Most cases will prevent a small amount of moisture from getting inside, meaning that you can carry them (or, much better, get your roadie to carry them) through the rain, but if you want to keep the elements out completely, the case will have to have a proper environmental rating, such as IP44 — as applied to outdoor mains connectors, for example. Such specialised cases — designed mainly for transporting items like laboratory instruments — tend to be expensive and are not always very robust, but are worth considering for very sensitive gear. I use such cases for my recording equipment, which tends to be carried in cars rather than vans and is only handled by me!
 
This flightcase was designed to hold audio-visual display equipment, not PA gear. The damage has been caused by a heavy object moving around inside.

Rack cases are available in different depths (front-to-back), and experience has taught me to use the correct depth if at all possible: too shallow and your equipment pokes out from the back (and is therefore at risk and looks naff); too deep and you'll be forever shining a torch inside and struggling to plug things into the proper holes. A good rack depth will comfortably accommodate the equipment, allow good access to the rear-panel connections and provide enough space to store or permanently install things like power cables. Having said that, if you do keep cables inside the rack (which saves time during setup), make sure that the metal plugs can't roll around inside and damage anything. A power distribution panel is a very useful thing in an effects rack, as it provides a neat and safe solution. Most of these can be mounted on the rear rack strips (if you have them), which means that you're not sacrificing an 'operational' slot in your rack.

One final thought: it's a good idea to label the outside of the case to identify the front and the top, so that it can be transported the right way up, and placed in situ the right way around. Saves time every time!

Trunk Call

 
This trunk holds 15 mic stands and also makes a useful trolley.

However useful road trunks are, there's a compromise to be made when deciding what goes into them and therefore what size you need. It's very convenient and very fast to have all manner of bits and pieces in one or two large trunks — just wheel 'em in and away you go — but consider the weight and size of large cases, and the difficulty of handling them. It doesn't take many cables or mic stands to make a road trunk into a heavy and unwieldy object, and you may then need a second person to help get them in and out of the vehicle or into the venue. One way around this is to use what I think of as the Russian doll approach: various bits of gear (for example, microphones, adapters, small signal cables, and so on) can be kept in small cases, and then several of these cases can be transported inside one larger trunk, depending on how much you need to take to the gig, and how many helpers you have. This gives the best of both worlds and also provides two layers of protection. A fully-loaded road trunk can be a difficult beast to control, especially if all four wheels are swivelling castors. I've lost count of the times a slight sideways gradient has given the trunk a mind of its own (the 'Shopping Trolley Effect') and then there's Postlethwaite's Theorem, which states that if one end of a laden castor-equipped trunk is lifted by a person, the opposite end will tend to describe an arc which terminates against an adjacent vehicle.

Mic stands are a real pain to carry when you've got more than two in each hand, and you can get neat little road trunks specially for them. Do watch out when emptying these long, thin cases though. Due to their tall, narrow shape and the weight of the lid acting on one side when open, some of them can be prone to tipping over as you take the last stand out. If possible, stand them up against a wall so that they can't do anyone any harm.

Pack Your Trunk

Trunk cases can save loads of time and leg-work when you're loading, unloading and setting up. Road trunks come in all shapes and sizes, can be used to transport and protect virtually anything, and cost as little as £150 or so new. The most obvious uses are for cables, microphone stands and the like — in fact, anything which otherwise would have to be carried in small numbers and in lots of trips between van and venue. It's great to roll the trunk right up to the stage area and simply pull out all the cables you need, in the correct order and neatly coiled; I reckon this is the biggest time-saver of all during setup. When packing up at the end of the night you can again save time by just throwing everything in, provided you remember to sort it all out before the next gig.

Case contacts

• Flightcase Warehouse

+44 (0)1827 60009.

www.flightcasewarehouse.co.uk

• R&J Flytes

+44 (0)1536 723451.

www.rjflytes.co.uk

• The Noizeworks

0870 240 3119.

www.thenoizeworks.co.uk

Equipment Transport Cases

For live sound equipment, rack cases and road trunks will cover most of the basic requirement, but some kit will require special attention. Up-market backline will often need specially-sized cases for life on the road. For example, a guitar 4x12 could be transported in a bespoke case where most of the height consists of a very deep lift-off lid, so that the cabinet can be left on the shallow base (and the castors) during the gig, if need be. For main PA speakers, very large flight cases would be needed, so unless you're on the road all the time, touring far afield or engaged in the hire business, you can generally get away with padded bags (available for as little as £80 a pair for popular compact PA speakers such as the Mackie SRM450) and a bit of careful packing and handling. Many items of professional audio-visual display equipment are housed in lightweight transit cases, which offer convenience of mobility and enough protection and look like 'real' flight cases, but are not designed to withstand roadie rage. Beware of re-using these cases for heavy items such as amplifiers, because the side panels may not be strong enough to withstand much of an impact.

Speaking of using and re-using, there are a lot of second-hand cases available, some at very good prices. It's worth taking a close look at older ones before purchasing, because it can be very frustrating and time-consuming to repair or replace things like aluminium edging strip, distorted hinges or seized-up butterfly catches. I must admit to having two fairly large and currently unusable cases in my gear graveyard because I just haven't got the time to repair them properly, and they're a waste of space and money if they're not working for a living!

Being Creative

This second-hand case was 're-foamed' to fit a Spirit FX16 mixer, and there's plenty of room for the power supply too.Because of the relatively simple construction of flightcases, they are quite easy to adapt; if you find one with a lift-off lid and shallow base, it can be converted into a cable trunk by turning it upside-down and putting the wheels on what was originally the lid.

You will need to turn any flip-up handles the other way round too, because they are only designed to take the load in one direction, and if you use them the wrong way around they can trap your fingers against the case. If not already fitted, wheel brakes are a good idea too, especially for heavy cases that have to remain upright in transit; having at least one locking castor should prevent too much moving around or the possibility of the case rolling off the tail lift.

How It's All Done

On my last trip to Flightcase Warehouse in Tamworth, I took my camera along and had a chat with Jason Furneaux, FW's general manager, and the company's owner and director, Steve Austin, about how the cases are made. Jason talked and then walked me through the whole process of turning raw materials and boxes of fittings into finished flightcases. They're all made from birch plywood, either 7mm or 9mm thick, which is supplied in large sheets and has a coloured (usually black) phenolic surface layer ready-bonded on both sides (they call it 'Hexaboard', but I'm not sure if this is a trade name or a generic name).
 
A cutting machine used on the boards that form the panels of the flightcases.

The case edging is aluminium extrusion in 7mm and 9mm sizes, depending on the board being used, and the rest of the 'raw' stock consists mainly of fittings (ball corners, castors, handles, hinges and butterfly catches) and the foam used to line the cases, which is cut to exactly fit the equipment going into the case. One case size, for example, can — if 'foamed' to suit — accommodate several different but similar items of equipment.

Flightcase Warehouse make around 120-150 cases every month, nearly all based on specific customer orders (more than half via their web site), so the manufacturing process has had to be streamlined and automated as far as possible, to keep prices down and production up. The various pieces of machinery in the workshop areas are set up to produce whatever model of flightcase is required, and a production run can be for a single case or as many as required to meet a customer's needs. All the specifications are maintained on a specialist piece of design software, which means that the tooling-up and identification of the component parts needed to build a particular size and shape of case is quick and straightforward. An order can literally be in production within minutes of being received. Not all the cases are for the music industry: clients include motor-racing teams, specialist equipment manufacturers (for example, drinks vending machines) and promotional display companies.

Stencil Case!

Don't forget the potential advertising value of your flightcases. They're often quite big, and they'll often be the first thing anyone sees when you roll up to a new venue, so a logo or name stencilled on the outside can give a good impression from the start. Stencilling your cases also adds a degree of security, and it can save time at a gig if the cases are labelled with their contents. However, I tend to use coded language for this, because you don't necessarily want casual observers watching you putting a box labelled 'little, very expensive microphones' into your vehicle. Something like '12 vox h/held' does it for me.

Making A Flightcase

When an order is received, a job sheet will be raised detailing the model and specification of the case required. The panels and aluminium extrusion are cut to the correct size and the necessary holes and recesses are routed and cut to accommodate the fittings at a later stage.
 
A case awaiting fittings...

The individual panels are then riveted together to form the basic shell of the flight case. This has been made into a much more efficient process by the introduction of specialised machines, such as the one I'd seen being delivered that very day. Jason explained that it was a 'long-arm riveter', which can punch rivets straight through aluminium extrusion and side panels without the need for pre-drilling. The 'long-arm' part means that the riveter can reach across larger panels and fasten the case together without stopping to re-position the work.

Once the case has been assembled, complete with all the correct openings and recesses, the 'hybrid' sections are attached. These are the aluminium 'mating' edges that fit around any openings, such as the edges of lids and doors, and must be properly aligned to maintain the case's structural integrity. After this, various other pieces of hardware, such as the metal ball corners, hinges, catches and handles can be added. This job must be done by hand, but because the routing and cutting is done according to the case spec in advance, before assembly, all the openings are exactly the right size for the fittings being used. When the case is complete with fittings, the castors — if required — are attached, either direct to the case, or mounted on an 18mm plywood wheel-board for larger versions. 
Internal foam having been glued into place, this case is almost ready to go.

Foam If You Want To

The final stage is to cut and fix the internal foam lining. Once again, the exact dimensions are taken from the design database and the foam (firm packaging foam called Jiffy foam) is cut to size with an electric knife, then fixed in place with a compressed-air spray-glue gun. This looks like brilliant fun, and you get to wear a cool mask!

The finished cases are checked over and sent to the despatch area for shrink-wrapping or bubble-wrapping, before the firm's two courier services come to collect them every afternoon.

Before leaving, I asked the owner, Steve Austin, whether the company had ever been asked to make anything out of the ordinary — and apparently they have. A gentleman once ordered a flightcase made to fit himself, to be used as his coffin, wheels and all. Now that really is rock and roll!

Published March 2006

Introducing KORG volca fm [with CC]

Q. Is it safe to run everything off one power outlet?

By Hugh Robjohns

I'm about to move house and set up a studio in the spare bedroom with all my stuff, and I'm just interested in what the best use of power points is. Is it acceptable (and safe!) to run many plug boards (between 24 and 28 at least) from one outlet in the wall, or is it better to use all the available outlets around the room? I have quite a lot of stuff to power in one small room, and I am thinking about perhaps getting a power conditioner as well — I don't want to be responsible for burning down our landlord's house!
 
If you want to avoid ground loops, it's best to run all your music-making gear from a single mains socket (like this UK one, shown).

SOS Forum Post

Technical Editor Hugh Robjohns replies: From the point of view of avoiding ground loops, it is best to run everything from a single socket, or from adjacent sockets if you have a double-socket outlet. This is a much better approach than running some gear from a socket on one side of the room, and other gear from a socket on the other side — a practice almost guaranteed to produce ground-loop problems!

If you are concerned about the total power you will be drawing from a single socket, you can reduce the load by plugging non-audio equipment into another socket in the room — desk lamps, phone chargers, kettles and so on.

If you're using plug boards to increase the number of available sockets, connect them in a star arrangement rather than serially. By that I mean you should connect one board to the wall socket, then plug the other boards into that first one, and then plug the equipment into this second 'layer' of boards. That way, the earth paths are kept as short as possible and in a star arrangement.

A single socket is able to supply around 3kW (230V x 13 Amps) and it is very unlikely that your domestic recording equipment will draw that much power — but every piece of equipment will have a label on it near the power connection that says what power (or current) it draws. If possible, try to balance the power demands on each plug board and make sure that all the fuses (in the plug boards and in individual plugs) are sensibly rated. Also, bear in mind that the first plug board has to carry the entire current load.

Published October 2005

Q. Should I apply bus processing while I am mixing?

By Mike Senior

 
Amongst other things, the Drawmer DC2476 provides high-quality compression and EQ.

I've seen it suggested that compressing the stereo bus is the key to getting a mix to come together and sound 'like a record'. Is this really the case, and if so, at what point in the mixing process should I be adding bus processing?

SOS Forum Post

Reviews Editor Mike Senior replies: I find that bus processing of various kinds does wonders in pulling together a mix, and I do usually mix through a selection of bus processors for that reason. But let's not get carried away — a great mix will benefit as much (or even more) from bus processing as a mediocre one, so having access to decent bus processors doesn't really let you off doing a decent mix, because these days most serious engineers have access to decent bus processing!

If you start with all your 'polishing' bus processes in place at the start of the mix, you're likely to work less hard at getting the basic mix right in the first place. I think this was one of the lessons to be learnt from the On-line Mastering Shootout listening tests we did here in the SOS office for our March 2006 issue. Trying to fix mix problems with bus processing is fantastically difficult — improving any element of the mix usually involves compromising some other part. So, if you don't do the mix properly to start with, you'll find it very difficult to sort out any problems later using mastering-style processing. It's important to do the very best you can with your mix before you switch in any bus 'polishing', otherwise your final results will suffer.

My general advice would therefore be to avoid bus processing for as long as you can with a mix, so that you get it sounding as good as possible without any extra help. However, there are a couple of exceptions I would make to this.

The first is that, speaking personally, I usually patch in a full-band bus compressor over the mix while I'm creating my opening balances. Whether you would find this suitable as well will depend on whether you use a pumping compression sound as I tend to. If you plan to, then I'd suggest mixing with the compressor switched in — your balance decisions have to be different if you are intending to hit a bus compressor in this way, so you need to be able to hear what you're doing.

The second exception concerns using EQ rather than compression on the mix bus. It's a little trick I learned from our interview with Spike Stent, one of my mixing heroes, in SOS January 1999 (www.soundonsound.com/sos/jan99/articles/spike366.htm). He patches in a really high-quality EQ over the whole mix, boosting the 'air' frequencies so that he doesn't need to do this using lots of individual lesser-quality channel EQs. I use my Drawmer DC2476 mastering processor for this, so that I can stay in the digital domain. Spike was using a Massenburg EQ in preference to his 'low-quality' G Series SSL channel EQ, so the quality difference for him is probably less than for the rest of us — the difference between a really nice EQ processor like the Drawmer and the built-in digital channel EQs in a digital multitracker or software sequencer is much bigger.

What I would say though, even with regard to both these exceptions, is that you should always make sure to record completely unprocessed versions of mixes along with the processed versions. That way, if anything goes wrong with the bus processing, you don't need to completely redo the mix — you can just reprocess the unprocessed versions.

Published May 2006

The Korg minilogue with Lendrum & Forte

Q. Is a matched pair of mics necessary for stereo recording?

By Hugh Robjohns

 
Even small differences in the polar pattern and frequency response of the two mics in an X-Y pair will have an adverse effects on stereo image. Using a matched pair — these are Rode NT5s — is recommended.

Do I really need to use a 'matched pair' of microphones, or even two mics of the same make and model, for stereo recording? I guess professionals need the best quality possible, but for the rest of us surely it doesn't make that much difference. Can't you make good stereo recordings, even if the mics are not of the same make?

SOS Forum Post

Technical Editor Hugh Robjohns replies: Different microphones will sound... well, different! They will have different frequency responses, and more importantly, they will have different polar pickup patterns. What's more, the way in which the two microphones' polar responses vary according to the frequency of the signal will also be different.

If you try to use two different mics for X-Y coincident stereo recording, the inevitable result will be an unstable and ill-defined stereo image that appears to wander about, with different instruments affected in different ways. This isn't a subtle effect, either — it takes very little difference between the two mics to make this a serious problem. This is precisely why the high-end mic manufacturers go to such trouble to maintain tight tolerances in producing matched pairs, and why companies producing more affordable mics offer hand-selected matched pairs.

If you want to check the compatibility of two cardioid (or any other directional pattern) mics, try this simple experiment. You'll need stereo monitoring, a mixer and somewhere to set the mics up where an assistant can walk around them while you listen to the monitoring output in isolation — a studio with a separate control room would be ideal.

Place your two mics above one another with the two capsules facing forward along the same axis and as close together as you can get them. Plug one into the left channel of your mixer and the other into the right channel, pan them hard left and right and match their gains exactly.
 
High-end mic manufacturers like Microtech Gefell test matched pairs to very tight tolerances.

The easiest way to do this is to get someone to stand in front of the mics and talk. Meanwhile, you select the mono button on your monitoring and reverse the polarity of one of the mics. Fade up the first mic and set the gain to a sensible level to hear the speech. Then as you fade up the second mic some cancellation should occur as the polarity is reversed. Adjust the second mic's gain to achieve the deepest cancellation null. Now you can remove the polarity inversion and cancel the mono monitoring. You should now hear the speech coming from mid-way between the two speakers as a phantom centre image.

Now ask your assistant to walk around the mics, in a circle, keeping roughly the same distance from the capsules and facing them all the time as he or she walks around. If the mics are perfectly matched, the image will not move from the centre line, although the level will obviously vary as the assistant moves around to the rear null of the cardioids, and then back towards the front around the other side.
With two dissimilar (or poorly matched) mics, what you will hear instead is the image wander or flick about, often with different frequency components appearing to come from different places — sibilants from the left and fundamentals from the right, for example. The image will sound unstable and poorly defined.

Exactly the same kind of imaging inaccuracies will occur when the mics are rotated to face left and right as in a conventional X-Y stereo pair, resulting in a poorly defined, blurry image — definitely not a good stereo recording!

Spaced mic arrays aren't quite as critical in terms of mic pattern because level differences caused by the polar pattern don't play as large a role in defining the stereo image. But you will still suffer from strange and unstable imaging if they have different frequency responses.

The only stereo technique that deliberately uses different mics is the coincident M&S array — but that requires that one mic have a figure-of-eight pickup pattern. Even so, it is important that both mics have similar frequency responses if the imaging is to be accurate. In this case, though, poor matching results in unstable width problems, rather than image shifts, which is probably less audible.

Published May 2006

Korg Minilogue: Motion Sequencing (Part 8 of 8) [with CC]

Q. What kind of bass trap do I need?

By Paul White

We have set up a new recording studio at our school. It's not huge, but we do have a separate (very small) control room as well as a live room. We were expecting to need acoustic treatment and have used bass corner traps from Studiospares as well as some general treatment in the control room to reduce reflections from the walls at the mix position. But, although most of the room modes have been cured, we have a humdinger of a resonance at about 58Hz — using the test oscillator in Logic Pro shows it up nicely. I'm at a loss to know what to do next but wondered if a tuned bass trap was the answer, as it is very specific to that frequency. I think the problem is that the height and length of the room are about the same, even though the room is not rectangular (one end wall is angled).

Tim Morris
 
Editor In Chief Paul White admires his handiwork. 

Technical Editor Hugh Robjohns replies: Small rooms are always difficult to treat well, especially if some of the dimensions are similar, because the modes tend to pile up very close together. A tuned trap might well help, but I would suspect you would actually be better off extending the existing bass trapping.

Commercial foam traps are OK, but you'll get better performance if you build some DIY corner traps using slabs of Rockwool two to four inches thick in simple timber frames covered with fabric.
 
This diagram shows a cross-section of the bass trap we built in the Studio SOS article in the March 2006 issue.

Suitable examples of this approach — as well as more complex limp membrane absorbers — have been illustrated in the pages of SOS recently. In the March 2006 issue, for example, we explained in detail how to build just such a trap in the course of a Studio SOS visit. We've included the diagram (right) again for reference, but I'd definitely advise reading the whole article. There's lots of other information on acoustics available on our web site, and there's always plenty of debate on the subject in our DIY Studio Acoustics forum, too.

In a small room you can't really put in too much trapping, so I would suggest treating all four vertical corners, and if possible, go for the wall-ceiling corners too. It might also pay to place Rockwool absorber panels behind the speakers to help reduce any back wall reflections at the bass end.

Published May 2006

Korg Minilogue: Clicking (Part 7 of 8) [with CC]

Korg Minilogue: Slider (Part 6 of 8) [with CC]

Q. Should I use my mixer's group outputs or its direct outs for recording?

By Mike Senior

 
Like other mixers, this Allen Heath GL2400-424 offers both direct outs on channels and group outs. But which should be used and when?

I recently started teaching music technology in a college and was asked to rebuild one of the studios. It uses a 32-channel mixing desk, patchbay and Alesis HD24 hard disk recorder to record to, as well as outboard gear. The desk has eight group busses arranged in four stereo pairs. There are 24 mono group output sockets, three per group buss, so that group 1 goes to outputs 1, 9 and 17, group 2 goes to 2, 10 and 18, and so on. The way it was set up previously was that these 24 group outputs were normalled through the patchbay to the 24 inputs on the HD24. The students were being taught that the signal should come into the desk and then be routed through the relevant group to get to the HD24. For instance, if your mic is plugged into channel 3 and you want to go to track 5, you have to route it to group 5-6, pan it hard left and bring up the channel fader and group fader. However, I changed it so that the direct outs of the first 24 channels are normalled through to the 24 inputs of the HD24, which seems to make more sense. One of the lecturers is kicking up a fuss, so my question is: which practice is most common in professional studios?

Thom Corah

Reviews Editor Mike Senior replies: You're both right after a fashion, but I'm afraid that I think the lecturer is probably more right in this case, as you appear to be using a group desk, rather than an in-line one. Your approach has two main limitations. Firstly, you can only route channel 1 on the desk to channel 1 on the recorder. This is admittedly less of a limitation with a digital recorder, where you can swap tracks digitally, but it's still quicker to do this from the desk than from the recorder.

The second (and more serious) limitation is that you can't record a mix of several channels to the same track on the recorder. Although 24 tracks is quite a lot to work with, you might need to submix a number of microphones to, say, a stereo pair of tracks — for example, when layering up a string quartet a few times to make a composite string sound for a pop production. Another problem is that you can't use the mixer's EQ on the way to the recorder, as direct outputs are often taken from before the EQ circuitry. Also, you couldn't bounce down a group of tracks through the desk in this way without sending them all to a group first, and then patching from the group output to a further channel. So you'll have more flexibility if you do things the lecturer's way.

One reason that you're not completely wrong is that you're implementing a kind of in-line methodology, treating the input stage up to the direct output as the input path and the rest of the channel as the monitor path. However, a group desk isn't really sufficiently well equipped to do this properly, most notably because there is no routing matrix between the input channels and the recorder inputs, as there would be on an SSL desk or similar. There's only one routing matrix per channel on a group desk, and that is situated after the channel fader. There's no real alternative, given the facilities, but to have separate channels for the input and monitor paths. In your case, as you have only 32 mixer channels, this means repatching for mixdown and monitoring purposes, I imagine, but I don't know all the details of your setup.

One situation where you can get away with using an in-line configuration on a group desk, exactly as you have, is where the recorder is actually a computer system. In this case, given the powerful processing facilities a computer offers, there's little advantage these days in pre-processing audio before it reaches the computer, so the lack of input EQ would not really be a problem. Also, there are comprehensive input routing and mixing facilities built into most modern audio-recording packages, so a hardware routing matrix would also be unnecessary. Perhaps you could justify your routing scheme as just being a little ahead of its time? You are simply anticipating the happy day when the college moves to a more flexible computerised system!

At the end of the day, which is the more appropriate arrangement depends on how many tracks you plan to record at one time. The group routing approach is more flexible when it comes to being able to do track bounces and partial submixes, and it is an important way of working to teach students. However, the down side is that you can record no more than eight (different) tracks at a time because there are only eight groups on your mixer.

Taking the direct outs approach allows up to 24 different tracks to be recorded at the same time and is ideal in areas designed purely for tracking, but you are then in for lots of replugging when it's time to mix. In any case, students should definitely be made aware of both techniques and configurations.
One possible solution that you could consider is using the patchbay to normal the group outputs to the recorder inputs, as before, but also send all of the desk's direct outs to patchbays on the row above, so that when you need to patch direct outs straight into recorder tracks it's just a case of plugging in some patch cords.

Published January 2006

Korg Minilogue: Voice Modes (Part 5 of 8) [with CC]

Q. What is 'aliasing' and what causes it?

By Hugh Robjohns

 
Figure 1: The D-A converter's low-pass filter, set at half the sample rate, removes the upper and lower images while keeping the wanted audio.

With reference to A-D/D-A converters, what exactly is an 'alias'? How and when do they occur and what causes it?

SOS Forum Post

Technical Editor Hugh Robjohns replies: An alias occurs when a signal above half the sample rate is allowed into, or created within, a digital system. It's the anti-aliasing filter's job to limit the frequency range of the analogue signal prior to A-D conversion, so that the maximum frequency does not exceed half the sampling rate — the so-called Nyquist limit.
 
Figure 2: When the 10kHz signal overloads the A-D converter, the resulting third harmonic at 30kHz creates an alias at 18kHz which will be allowed through by the low-pass filter.

Aliasing can occur either because the anti-alias filter in the A-D converter (or in a sample-rate converter) isn't very good, or because the system has been overloaded. The latter case is the most common source of aliasing, because overloads result in the generation of high-frequency harmonics within the digital system itself (and after the anti-aliasing filter).

The sampling process is a form of amplitude modulation in which the input signal frequencies are added to and subtracted from the sample-rate frequency. In radio terms, the sum products are called the upper sideband and the subtracted products are called the lower sideband. In digital circles they are just referred to as the 'images'.

These images play no part in the digital audio process — they are essentially just a side-effect of sampling — but they must be kept well above the wanted audio frequencies so that they can be removed easily without affecting the wanted audio signal. This is where all the trouble starts. The upper image isn't really a problem, but if the lower one is allowed too low, it will overlap the wanted audio band and create 'aliases' that cannot be removed.

Let's consider what occurs if we put a 10kHz sine-wave tone into a 48kHz sampled digital system. The sampling process will generate additional signal frequencies at 58kHz (48 + 10) and 38kHz (48 - 10). Both of these images are clearly far above half the sample rate (24kHz), so can be easily removed with a low-pass filter, which is the reconstruction filter on the output of the D-A converter, leaving the wanted audio (the 10kHz tone) perfectly intact. See Figure 1, above.

However, consider what happens if our 10kHz tone is cranked up too loud and overloads the A-D converter's quantising stage. If you clip a sine wave, you end up with something approximating a square wave, and the resulting distortion means that a chain of odd harmonics will be generated above the fundamental. So our original 10kHz sine wave has now acquired an unwanted series of strong harmonics at 30kHz, 50kHz and so on.

Note that these harmonics were generated in the overloaded quantiser and after the input anti-aliasing filter that was put there to stop anything above half the sample rate getting in to the system. By overloading the converter, we have generated 'illegal' high-frequency signals inside the system itself and, clearly, overloading the quantiser breaks the Nyquist rule of not allowing anything over half the sample rate into the system.

Considering just the third harmonic at 30kHz for the moment, the sampling modulation process means that this will 'mirror' around the sample rate just as before, generating additional signal frequencies at 78kHz (48 + 30) and 18kHz (48 - 30). The 18kHz product is clearly below half the sample rate, and so will be allowed through by the reconstruction filter. This is the 'alias'. We started with a 10kHz signal, and have ended up with both 10kHz and 18kHz (see Figure 2, above). Similarly, the 50kHz harmonic will produce a 2kHz frequency, resulting in another alias.

Note that, unlike an analogue system, in which the distortion products caused by overloads always follow a normal harmonic series, in a digital system aliasing results in the harmonic series being 'folded back' on itself to produce audible signals that are no longer harmonically related to the source.
In the simplistic example I've explained, we have ended up with aliases at 2kHz and 18kHz that have no obvious musical relationship to the 10kHz source. This is why overloading a digital system sounds so nasty in comparison to overloading an analogue system.

I hope this brief explanation helps to clear up the topic of aliasing for you.

Published January 2006

Korg Minilogue: Modulation (Part 4 of 8) [with CC]

Q. How can I achieve phase inversion?

By Hugh Robjohns

In the October 2000 issue of Sound On Sound there was an article called Improving Your Stereo Mix. In example three ('Ye olde phase trick'), you explained a technique that increases the perceived width of the stereo mix using phase inversion. I want to use this technique, but as my mixer doesn't have phase-invert buttons I'm not sure I can?

I have a Signex CP44 unbalanced patchbay in which I've wired some sockets in parallel, so splitting the signal is no problem. On the returns, could I use balanced cable and swap the hot and cold pins to create the phase inversion, or would this not work, as the original signal and patchbay are unbalanced?

Dez Ford

Technical Editor Hugh Robjohns replies: The technique you're referring to works by adding some of the right-hand signal to the left-hand channel out of phase and some of the left-hand signal to the right-hand channel out of phase. You can read the original article on-line at www.soundonsound.com/sos/oct00/articles/stereomix.htm, but I've included the original diagram here for easy reference. Many phase-inversion tricks aren't mono-compatible, but one advantage of this widening technique is that when the left and right channels of the final mix are summed to mono, the effect disappears without causing any problems.

Swapping the hot and cold pins at one end of the cable feeding a balanced input will indeed invert the phase of the signal. Another way you can achieve the same thing is to use a pair of aux outputs from the main channels (one for the left signal, the other for the right) instead of splitting the signal at the patchbay. The advantages are, firstly, that the aux outputs are properly buffered and, secondly, if you make these post-fade sends, the inverted return channel levels will follow the input-signal level, making mixing easier. You may also benefit from the fact that in many budget consoles the aux outputs are phase-inverted relative to the main inputs, which would avoid the need to make up inverting cables! You'd have to experiment to find out if your mixer provides inverted aux outs.

Published January 2006

Q. Should my Valve Mic be this noisy?

By Hugh Robjohns

I have just got hold of a CAD M9 valve mic and am concerned about the noise level from it. When I switch it on it chuffs and farts a bit, much as my Fender valve amp does, then settles down. That seems OK. But once it has warmed up, the background noise level seems high compared to my other condensers. If I have a vocal take at normal levels being recorded at about -6dB peak, then the noise level is registering at -38dB. The vocal sounds fine, but the noise seems high. Is this what I should expect from a valve mic? Should I try another valve in it?

 

A faulty valve can easily be replaced; more serious repairs should be carried out by an experienced technician.

SOS Forum Post

Technical Editor Hugh Robjohns replies: Valve mics are generally more noisy than solid-state condensers. The M9 is specified with a self-noise figure of 15dBA, which is roughly 8dB higher than the best of the large-diaphragm solid-state designs — the Neumann TLM103 has a self-noise figure of 7dBA, for example.

However, while it is possible that your mic is faulty or requires a new valve, the high noise floor you describe could also be down to poor mic technique.

With vocals peaking at -6dBfs, a noise floor of -38dBfs does seem poor. The question is, how much of that noise floor is due to the mic, and how much is due to the recording environment? Are you recording a low-volume source at a considerable distance, or in a noisy room, or with a poor-quality mic preamp, for example?

If you have access to another large-diaphragm mic, I would suggest you rig that alongside the M9 and adjust the gain to get the signal peaking at the same level for both mics, and then compare the background noise floors. If both mics deliver similar noise levels, then the room or your technique are at fault. If the M9 is more than a few dBs noisier than a solid-state large-diaphragm mic, then the M9 is in need of repair.

It could be that the valve is faulty or worn out, and certainly that's the easiest thing to replace yourself. However, there could also be a problem in the power supply or elsewhere in the mic's output circuitry, which would require a return to the supplier to be fixed.

Published August 2005

Korg Minilogue: Filters (Part 3 of 8) [with CC]

Q. Is it worth isolating my speakers and other equipment?

By Hugh Robjohns

 
With properly designed and constructed monitors, whether active or passive, you needn't worry about internal vibrations damaging electrical components.

I would like to know how much benefit can be gained by isolating my speakers and other gear from their supports — so-called 'seismic isolation'. I recently saw a forum posting suggesting that passive monitors have an advantage over active ones as, in the case of the latter, vibrations from the speaker can affect the components of the built-in amp. The benefits of decoupling equipment using springs that have a very low resonance frequency so that it 'floats' was also discussed. Apparently many things can benefit from this technique — not just speakers but CD players and studio gear also. Improvements to the stereo field and depth are said to be quite noticeable. I would like to know if the £1300 I spent on an active Blue Sky System One (which I like very much) would have been better spent on passive speakers and an external amplifier. Can you shed any light on all of this for me please?

SOS Forum Post

Technical Editor Hugh Robjohns replies: With regards to what you read about internal vibrations in monitors, in general neither the active or passive form has an advantage in this regard, and both potentially suffer exactly the same problem.

Clearly, there is a lot of sound energy inside most loudspeaker cabinets, and if that energy is allowed to impact on electronic circuit boards it is possible that some components might resonate and vibrate, eventually resulting in damage to the solder joints or the components themselves, and possibly such mechanical resonances might affect the electrical signal passing through the components. However, this would apply equally to passive crossover boards as much as active amplifiers.

In 30-odd years of playing around with loudspeakers in many and various forms, I can't say I have ever found this to be a real problem. I have occasionally come across speakers that have suffered component or solder joint failures, but in all cases the causes have been traced to faulty production or failures in quality control. When the faults were fixed properly, none recurred as far as I am aware — even though you would expect them to if the sole cause was sound vibrations within the cabinet. So I am confident that this argument can be set aside as a popular but completely unfounded myth.

Mechanical isolation of speakers or other devices from their supports can be used to advantage in certain situations, but it is a complex subject and it is easy for the inexperienced to make the situation worse with inappropriate decoupling systems. In my experience, most equipment works best when mounted on solid, heavy supports — there is nothing as effective at controlling vibrations as a lot of mass.

However, sometimes it is necessary to come up with some form of decoupling to prevent vibrations generated in one source from entering an adjacent surface. The classic example is that of placing nearfield speakers on a desktop, when the inherent speaker cabinet vibrations will often cause the desktop to vibrate and resonate, resulting in unwanted rattles. In this situation, placing the speakers on some form of decoupling medium can improve matters — something like the Auralex Mo-Pads, for example, are very effective. However, far better results can be obtained by removing the speakers away from the table top completely and mounting them properly on solid, heavy stands placed directly on the floor.
 
Auralex Mopads can be used to isolate monitors placed on a desktop, but heavy-duty floor stands are best of all.

As far as equipment is concerned, I don't subscribe to the view that properly designed and manufactured amplifiers and other electronics should be decoupled to improve stereo imaging or anything else. However, when it comes to systems involving some mechanical element — like record players, CD players and so forth — unwanted vibrations entering the mechanical system certainly can cause problems.

Most people are very well aware of the susceptibility of record players to external mechanical or acoustic vibration. The required tracking precision in CD players and DVD players is many orders higher, and mechanical vibrations that reach the mechanism will affect the accuracy of the tracking. Potentially, this will cause the tracking and focus servos to work harder, forcing greater current flows at higher frequencies through the motors. In cheaper designs, this may well affect the power supply's stability and result in noise currents reaching other parts of the circuitry. Reduced tracking precision can also potentially result in a greater uncorrected error rate and far more jitter. Cheap and poorly designed players are likely to suffer these effects to a much higher degree than properly engineered equipment, which will usually incorporate properly decoupled drives, effective de-jittering circuitry, and so on.

It's a familiar scenario in the hi-fi world — people discover that badly engineered equipment reacts 'unexpectedly' to different cables, mechanical decoupling, or painting with a green pen — all of which bestow a 'miraculous' benefit to the sound... and then declare (from no scientific basis whatever) that all vaguely similar equipment will behave the same. It's just not the case.

As to whether you would have been better off buying passive monitors and an amp, the answer is probably not. I can think of some excellent passive monitor and amp combinations for the rough cost you mention, and in direct comparisons I dare say some people would prefer a passive speaker and amp configuration over your Blue Sky System One. But it comes down to personal preferences regarding sound, convenience and styling, and how the system works in a given room. I think you can continue to enjoy your Blue Sky system and completely disregard any faux concerns raised by the technical myth-spreaders!

Published August 2005

Korg Minilogue: Oscillators (Part 2 of 8) [with CC]

Q. Why don't my mixes sound good on a subwoofer system?

By Hugh Robjohns

 
Steinberg Wavelab's real-time audio analyser can help to give a clearer picture of the frequency content of a mix.

I've just finished mixing a couple of tracks and I'm checking out how they sound on different systems. So far I've managed to get them to sound OK (well, acceptable anyway) on my Genelec 1030s, different sets of hi-fi speakers, a boombox, TV speakers and so on. I thought I was on my way to engineering stardom until I tried the tracks on a friend's system which has a subwoofer. Even with the subwoofer turned down a bit, the amount of boomy low bass was incredible and pushed the other frequencies to the back. Do you have any tips on how to prevent this from happening? I guess EQing out the sub-bass frequencies is the trick, but how do you know what you're doing if all these other sets of speakers don't give you a clue as to what's going on?

SOS Forum Post

Technical Editor Hugh Robjohns replies: Do not despair! I strongly suspect the problem is with your friend's monitoring system rather than your mix, especially as it sounds OK everywhere else. Genelec 1030s are pretty good at telling you what is going on at the bottom end, even if you are doing daft things with subsonic rumbles, and if nothing stands out as silly on them, then it tends to point to a bad subwoofer setup.

So, assuming your track doesn't contain stupid amounts of sub-40Hz bass rumbles, you need to help your friend to sort out the standing-wave problems in his room, and to set up his subwoofer properly.
If you can, check your mix on a real-time audio analyser (as found in Steinberg's Wavelab, above) to see what is going on below 80Hz or so. Make sure there are no excessive peaks in the low frequency area — compare your mix against commercial tracks in a similar music style to get a feel for what is 'normal'. I suspect you'll find your track is fine and it is the combination of the room and the subwoofer setup that is the real problem.

A handy way to check for standing waves and the resulting uneven bass response in a given listening setup is to record a simple sine-wave signal from a synth or sound module playing each note in turn over the bottom two octaves. Make sure that all the notes have the same velocity, and ideally, make each note a 'ping' rather than a constant drone.

Play that back over each system in turn and you'll be able to assess their ability to reproduce low bass evenly, and the effect of the room's standing waves. I expect that in your friend's room you'll find some notes set off huge resonant peaks and others disappear completely.

Standing waves are a common problem, and one we come across all the time on our Studio SOS visits, in letters and emails to Q&A and on the SOS Forum. The solution is to set up some proper acoustic treatment in the room and to adjust the positioning of the subwoofer. As a starting point, I suggest you read 'Monitoring & Acoustic Treatment' and 'Choosing & Installing a Subwoofer' by Mallory Nicholls. You also might want to read through some previous Studio SOS articles for some practical examples of room treatment, and searching the SOS web site for terms like 'acoustic treatment' and 'subwoofer' will turn up lots of useful information.

Published July 2005

Korg Minilogue: Oscillators (Part 1 of 8) [with CC]

Korg GEC5 Software Tutorial – Initial App Setup

Q. Should I buy a vintage analogue synth or a modern modelling synth?

By Steve Howell

I want to buy a 'knobby' synth because I am fed up of setting up sounds with a data wheel. I also want a very analogue sound. I am thinking of buying a modelled synth but, at the same time, I would really like a genuine vintage synth to get a 'real' analogue sound (and because they look so cool!). Any pointers would be appreciated.

Ben Slater

SOS contributor Steve Howell replies: As was pointed out in Sounding Off in SOS February 2004 (www.soundonsound.com/sos/feb05/articles/soundingoff.htm), analogue synths are not without their pitfalls. Firstly, assuming you can actually find a good example of the synth you favour, they can be costly to buy but, more importantly, they can also be costly to maintain.

When buying a vintage synth, you should check for noisy pots and switches. Whilst these can often be fixed with a squirt of an appropriate contact cleaner, replacing them can be expensive, especially if the pots are surface-mounted to the PCB and/or the switches aren't now available. You might think that noisy pots aren't really a problem, but a large part of the appeal of a knobby synth is the ability to tweak controls during a performance — there's nothing worse than your ripping solo being spoilt by the intrusive sound of crackles!

You must also check out the keyboard. Often, the keyboard mechanism on these old synths is very simple and it is all too easy for the contacts to break (or become bent or twisted so that they don't make contact). You can sometimes fix these yourself if you're handy with a soldering iron, but getting them repaired or replaced by a specialist is likely to set you back a few bob! And what about MIDI? Most old synths don't have it, although they can usually be triggered by control-voltage (CV) and Gate signals. So if you want to integrate the vintage synth into an existing sequencing setup, you're going to have to seek out an example that has a MIDI retrofit, or budget for some kind of MIDI-to-CV converter.
 
Korg's analogue-modelling MS2000B synth is more flexible than the vintage MS20 (top) and still offers plenty of 'tweakability'... but is it as desirable?

Then there's the sound-generating circuitry itself. By definition, it's going to be old, and components may be failing, leading to tuning and other instabilities as well as noisy outputs — I once tried an ARP Axxe that sounded as though someone was frying bacon in the background! Not only are these problems potentially costly to repair but it could well be that some components are simply not available any more, especially if the manufacturer used any integrated circuits that are now in short supply, or worse, custom components.

Of course, if you buy the synth from a reputable dealer who specialises in vintage synths, a lot of these issues can be avoided, as the stuff they sell will invariably be refurbished (or at least serviced prior to sale) and will often carry some form of warranty. You will pay a bit more for that peace of mind, understandably, but it can be worth it.

You should also listen carefully to anything you are thinking about buying — or even do a blindfold test — and ask yourself, "Does it actually sound good?". Do not allow yourself to be deluded by the attractive retro looks or the allure of owning a genuine analogue. Due to component tolerances (and failing components), not every analogue synth sounds good (or even the same as another identical model). And just because it has a Moog badge on it (or whatever), don't consider that a guarantee of 'fatness', 'warmth' or any other adjectives that are applied with dewy-eyed nostalgia to anything vintage.
 
If you have to have a true analogue synth, the Voyager by Bob Moog (above) might be expensive, but you won't find a MIDI-equipped original Minimoog (top) in pristine condition and perfect working order for less money, if at all.

I had lusted after an ARP Odyssey since the time I tried one as a teenager in Rod Argent's Keyboards back in the mid-'70s, and when one was offered to me many, many years later for a very silly price, I bought it on spec without checking it out first — bad move! When it arrived, it looked gorgeous — a prime example of a white-faced original, with all its sliders intact — but it was a totally underwhelming example of the instrument, and not at all what I had been remembering so fondly. I guess what I am saying is, don't buy an old synth wearing rose-tinted spectacles. If you do, you may well be in for a disappointment!

Modern, modelled synths are often a much better bet as a long-term investment. To all intents and purposes, and perhaps contentiously, they sound equally as good as the majority of vintage synths, if not better in some respects. They are inherently more flexible, are usually polyphonic, and are often more versatile, with sound-shaping facilities that the originals could only have dreamt of. They are also usually multitimbral, come with effects to polish the sound built in, and may have sophisticated (and often programmable) arpeggiators. They might not sound exactly like a vintage Moog, ARP, or Roland, but they're pretty close, and (unless you're very unlucky) won't spend much time being serviced.

I guess the only slight downside to these modern, modelled synths is that whilst many have plenty of knobs, they don't always have a control or switch for every parameter, unlike original analogue synths. Often, the less frequently used parameters on the modelled versions are accessed via an LCD and menus. However, it's perfectly possible to create very vibrant and convincing analogue synth sounds without ever having to delve into the more obscure aspects of the synth's programmability.

No-one has a greater respect for old synths than I do — after all, they paved the way to the technology we enjoy today. But just because a synth is old and carries a badge doesn't make it good. Witness the Polymoog — what a weak-sounding, unreliable crock! There are some great old synths out there if you can find a good example of one that satisfies your requirements and budget, but don't dismiss the more recent modelled hardware synths.

If you're still in the market for analogue, check out Gordon Reid's guide to buying a vintage keyboard from SOS September 1994 — see www.soundonsound.com/sos/1994_articles/sep94/vintagesynths.html. And for a more detailed idea of some of the things that can go wrong with vintage gear, check out the two-part feature on equipment servicing that appeared in SOS March and April 1996 (see www.soundonsound.com/sos/1996_articles/mar96/servicing.html and www.soundonsound.com/sos/1996_articles/apr96/servicing2.html).

Published May 2005