Havian30 Video Manual: SongBook

Q. Does freezing really improve the quality of valves?

By Hugh Robjohns

I've heard that some companies routinely freeze valves and certain electrical components to improve their performance. Would this technique actually have any effect on quality?

Clive Edmondson, via email

SOS Technical Editor Hugh Robjohns replies: There are certainly some people who claim that it can, but I've not experienced it personally and my knowledge of materials science isn't up to giving a definitive view, I'm afraid.

However, the basic idea of 'deep cryogenic treatment' is to freeze the valve to an extremely cold temperature (well below the capability of any domestic freezer: typically, the valve is submerged in liquid nitrogen at about ‑195 degrees Celsius and stored like that for a day or so before gradually returning it to room temperature. This cryogenic process is claimed to allow the crystalline structure of the metals used in the valve plates to realign in a way that allows electrons to flow more easily (ie. resistance is reduced), and that's what brings the claimed benefits to sound quality.

 
Assuming that it's done slowly and in a controlled environment, valves can be frozen in liquid nitrogen for a while without being damaged. Some companies offer this as an option on the valves they sell, as it is said to improve long‑term reliability.

 
Certainly, strange things can happen to metals at very low temperatures — like superconductivity — but these effects don't usually last when the metal is returned to room temperature. NASA also apparently use cryogenic treatments to prepare materials for use in space. But whether the science behind their applications extends to the use of audio valves at room temperature, I'm not so sure. Allegedly, the benefits of the cryogenic process remain throughout the life of the valve, despite the heating and cooling cycles it will go through in normal use.

Given the natural variability in valve sets anyway, I think it likely that differences will be heard between normal and cryogenically treated tubes. However, whether those differences are really better or just different — and whether the cost is justified — I suspect comes down to personal choice.


Published December 2010

Havian30 Video Manual: Syles

Q. Why does using a high-pass filter make things seem louder?

By Hugh Robjohns

Why does applying a high‑pass filter to a sound sometimes result in the output being noticeably higher than it was before? Today I have been working on a sound that peaks at 0dBFS. It has a lot of low‑frequency content. I am applying a high‑pass filter at around 100Hz and the output from the EQ is peaking at around +4dBFS. Why should this happen? Most of the power in this sound is in the low frequencies, and it has little going on above 2kHz, so surely with the high‑pass filter most of the energy from the sound has gone!

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: This is a very common effect and there are several possible reasons for it. Fundamentally, the filtering process changes the shape of the waveform, so although there may be less total energy in the signal, the peak amplitude may well increase.

If you think about a bunch of different‑frequency tones all playing at the same time, their phase relationships vary continuously and add to or cancel each other to create the total waveform. Remove some of those tones and some of those cancellations won't occur. That can result in the waveform becoming bigger.

 
There are various reasons why using a high-pass filter on a signal can make it sound louder. Some equalisers, for example, actually boost the region just above the turnover frequency, which can produce an increase in peak level.

Most equalisers also introduce significant phase shifts and that, again, will change the way different frequencies combine and cancel. It can also happen because some equalisers actually boost the region just above the turnover point below which they are attenuating, potentially increasing peak level.

Published December 2010

Havian30 Video Manual: Recording a Song

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

Announcing KORG Gadget for Mac 1.5 | for iOS 3.5

Q. What should I do about the dent in my tweeter?

I noticed a dent in the tweeter of one of my Yamaha MSP10 monitors this morning (the kids, I'm sure). Is this something to be concerned with or will it be OK? It isn't ripped, but a third of the silver cone is dented.

SOS Forum Post

Technical Editor Hugh Robjohns replies: This is a problem which we have encountered before in these pages and which comes up regularly in the SOS Forum.

 
A dented tweeter can be repaired but its performance will be permanently compromised.

It's usually possible to suck dents back out (of which more later), but the bottom line is that once dented, the tweeter will be damaged and never perform exactly as intended. I'm afraid the only way to regain perfect performance is to replace the damaged tweeter. To maintain stereo imaging precision, ideally you should change the other speaker's tweeter too. That way, both drivers will be the same age and, hopefully, close to the same specifications.

However, how audible the damage is and how much it bothers you will depend on how good the complete monitor is in the first place and how good your ears are! With a relatively affordable monitor like the MSP10, there is a good chance that you will not be able to hear a great deal of difference in the tweeter if you 'repair' the dent, so it is worth attempting to fix yourself.

A commonly suggested technique is to use sticky tape to pull dents in the cone back out. This can work, but I have seen more tweeters further damaged (by ripping off the coating or else leaving a residue) than I have seen fixed this way. So I would strongly advise against this technique.

Instead, I would recommend trying to gently suck the dent back out. I know lots of people who have been successful sucking the tweeter back out with nothing more than their mouths — the minimal amount of moisture the tweeter is exposed to during this process doesn't seem to cause any problems.

If you don't like the idea of getting up close and personal with the speaker, I have had many successes using a vacuum cleaner. The trick is to hold the nozzle very, very steadily, and to start with it a couple of inches away from the tweeter before drawing it up slowly towards the tweeter. Do not allow the vacuum cleaner nozzle to touch or fully enclose the tweeter as you are likely to damage it further, but as you get close the tweeter will probably pop back into the right shape.


Published May 2006

Creativity Unleashed | KORG Gadget Plug-in Collection supports NKS

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 TM-60 - Advanced Combo Tuner and Metronome; a must-have item for practicing

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

 

Published March 2006

 

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

KORG Collection for Mac/Win | SOFTWARE INSTRUMENTS

Q. How do I get rid of the buzzing noises in my home studio?

I'm emailing you in the hope that you can help me with some problems I am having in my home studio. Each time I record audio I get a continuous buzz and high-frequency noise recorded along with any music. It is clearly audible during intros, quieter passages and outros. However, when my screensaver comes on, this noise drops considerably. I have an Inta Audio AMD 64 system with a 19-inch TFT monitor, an Emu 1820M interface, and I'm running Cakewalk Sonar 3 Producer Edition.

The other problem which I cannot trace is that my speakers quite frequently crackle and cut out, especially if I happen to play a harder or slightly louder note or sound. My suspicions are that the problem might lie in the main output area of my ageing Studiomaster Mixer. The monitors I have are passive Tannoy Reveals driven by a Samson Servo 170 amp.
 
If all else fails, using a DI box will break a ground loop.

Mel Hayler

SOS Contributor Martin Walker replies: Your buzz troubles sound suspiciously like a ground loop problem, especially since the noises change with computer monitor activity. It's always difficult to diagnose the culprit from a distance, but I wrote a step-by-step guide to tracking down such problems back in SOS July 2005 (www.soundonsound.com/sos/ jul05/articles/pcnotes.htm). Essentially, you have to temporarily unplug all the cables between your power amp and mixer, and whatever gear is plugged into the mixer. If the buzzing and high-frequency noises go away, this probably confirms a ground loop problem.

First, reconnect the cables between your mixer and amp/speakers and listen for the noises. If they have returned, you need to switch to using balanced or pseudo-balanced cables between the two. Next, connect the output of your Emu 1820M to your mixer and listen again. Again, if the noise returns, you either need balanced or pseudo-balanced cables between the two, although many mixers provide balanced inputs, in which case balanced cables are the better option. A search on www.soundonsound.com will provide you with details of cable wiring.

Continue to reconnect all your other gear, and you should end up with a hum-, buzz- and whistle-free system. If at any stage you can't get rid of a noise, try using a DI (Direct Injection) box between the two items to break the ground loop.

As to the crackling, it could well be your mixer. In my experience, the push-button switches used for routing are prime culprits, so try operating all such switches on your mixer that affect the main output, including any that affect the control room output, such as monitoring options. Often a few repeated push-on, push-off actions will clean the contacts sufficiently to cure the problem, temporarily at least. Other possible causes of intermittent audio are dirty pots or faders — again, moving them back and forth may clear the problem, although a quick spray of a suitable fader lubricant is a slightly more long-term solution.


Published April 2006

KORG volca mix - ANALOGUE PERFORMANCE MIXER

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

KORG prologue | A new-generation of flagship analog synthesizer (Teaser Trailer)

KORG AW-OT Series Tuners - The first ever tuner with OLED technology

Q. Why do my mixes clip when I apply a high-pass filter?

By Sam Inglis

 
Here we see the characteristic 'hump' just above the turnover frequency of a shelving filter.

There's something happening when I master my mixes that I can't find an explanation for. This has been bothering me for nearly two years, so I thought it was time to call the experts! After maximising my mix, so that the level of the audio is just below the point of clipping, if I insert a high-pass filter at, say, 40Hz, suddenly the audio starts to clip. I thought that after inserting a high-pass filter the level should drop, but this is not the case. I've tried it with several EQs but always with the same result. What is happening? I've been using digital EQ, but will this still happen with analogue EQ?

Johan De Visser

Features Editor Sam Inglis replies: It's surprising, but true, that using EQ can cause clipping whether you are cutting or boosting. There are two reasons for this.

One is that EQ changes the phase relationships between the different frequencies that make up a complex signal, such as a full mix. The result of this is that even though you're cutting the low frequencies, you could be shifting other frequencies around in such a way that they reinforce one another at a point where they had not previously done so. It's also possible that whatever signal element you removed was actually serving to 'cancel out' another element at some points, so removing it has created larger peaks at these points.

The other reason is that some EQ designs actually have a resonant peak at the corner frequency. If this is the case then applying a low-pass filter at, say, 40Hz might actually introduce more energy than it removes, assuming there was nothing below 40Hz in the signal to start with.

It's possible to experience gain increases when cutting with both analogue and digital EQ — it's in the nature of equalisation that this will happen. The answer is to always do loudness maximising as the last process in the mastering chain (apart from dithering) — always use EQ before the maximiser, not after.

Published December 2005

KORG KR-55 Pro: a Comprehensive Rhythm and Drum Machine

Q. What is the difference between mono with one speaker and mono with two?

By Hugh Robjohns

I read recently that when top engineers check their mixes in mono, they don't just hit a mono switch, but instead route the mix through a single speaker to hear it in true mono. What's the difference between the two?
 
A single speaker in a sealed enclosure is the classic means of monitoring in mono.

SOS Forum Post

Technical Editor Hugh Robjohns replies: It's important to check the derived mono signal from a stereo mix to ensure that nothing unexpected or unacceptable will be heard by anyone listening in mono, as could be the case in poor FM radio reception areas, on portable radios, in clubs, on the Internet and so on. Mono compatibility, as it's called, is very important for commercial releases — the artist, producer and record company want the record to sound as good as possible in these less-than-ideal circumstances.

In addition to simply checking the finished product, mixing in mono — or regularly switching the monitoring to mono while mixing — is very useful and a good habit to get into. Summing to mono removes any misleading phasing between the left and right signals that can make a stereo mix sound artificially 'big'.

The crucial difference between auditioning the summed mono signal on a single speaker, as compared to a 'phantom' mono image between two speakers, relates to the perceived balance of the bass end of the frequency spectrum. When you listen to a mono signal on two speakers, you hear a false or 'phantom' image which seems to float midway between the speakers, but because both speakers are contributing to the sound, the impression is of a slightly over-inflated level of bass. Listening to mono via one speaker — the way everyone else will hear it — reveals the material in its true form!

Checking the derived mono is always best done in the monitoring section of the mixer or with a dedicated monitor controller. Although a mono signal can be derived in the output sections of a mixer (real or virtual), this is potentially dangerous — if you should forget to cancel the mono mixing, you'll end up with a very mono final mix. It does happen, believe me! Sadly, very few monitor controllers outside of broadcast desks and related equipment provide facilities to check mono on a single speaker. Most provide a phantom mono image, which is fine for checking imaging accuracy and phasing issues, but no good for checking the mono balance.

Published November 2005

KORG prologue; Analog is only the beginning

Q. How do I lower the latency on my laptop?

I have been experiencing some big problems with latency whilst trying to use Cubase SX. I would be grateful for any help or advice you can offer me. I'm using a Sony Vaio laptop with a 1.4GHz Intel Celeron M processor, 512MB of RAM, a 60GB hard drive, and a Realtek High Definition Audio sound chip. I've tried reducing the buffer size on this driver and upping the sample rate to 96kHz, with no effect on latency. Could the cause be my hardware?



Carol Robinson

Features Editor Sam Inglis replies: The latency is almost certainly caused by the hardware — most built-in laptop sound chips only have Direct X and MME drivers, and these can suffer latencies of half a second or more. Ideally, you'd be better off with a specialist audio device for music with proper ASIO drivers: upgrading your sound hardware will improve both audio quality and driver performance. Either a PCMCIA or USB device should be OK, or a Firewire one if your computer has a Firewire port. However, you could also investigate third-party ASIO drivers such as ASIO4ALL (www.tippach.net/asio4all) which are designed to work with any hardware.

Published February 2006

KORG prologue; Analog is only the beginning

Q. What should be the next step for my studio?

By Mike Senior

I'm looking for advice on what's the best next step for improving my signal chain. I've got a budget of about £1000. I'm writing music to picture (so mainly instrumental, not too 'in your face'), mainly using sample libraries and the odd MIDI sound piped in from my Roland XV5080. I'll occasionally record live sources, but have so far never needed more than two inputs to do so.

My setup currently includes a Yamaha 03D mixer and a Focusrite Twin Trak Pro preamp, both with word clock, AES-EBU and S/PDIF I/O), and an M-Audio Delta 1010 audio interface with S/PDIF and word clock I/O. I never use more than two inputs on the 1010, but do use all eight outputs into the 03D for mixing down to a stereo pair. My sequencer is Apple Logic Pro 7 on a Mac G5, and I'm using Adam S3A monitors. I spent a large part of the summer knocking together traps and absorbers for my room, which now sounds pretty good.

So where would I get most bang for my buck? Should I get a word clock generator to help tighten things up? Something to replace the M-Audio interface? A friend suggested a summing mixer, but there seems to be so much discussion about whether these are voodoo or not that I wonder if I'd be capable of hearing the differences through my tin ears! I also appreciate that it might be worth going for better preamps, but I do so little live recording that I can't help but wonder if there's something else that would make more of an impact on my day-to-day studio work. Any thoughts?

SOS Forum Post

Reviews Editor Mike Senior replies: There are a few different possibilities here, I'd say. I can understand that you're a little reluctant to shell out masses of money on a new preamp, but have you thought of looking at a decent A-D converter instead? Something like the RME ADI2 would be a neat product for your purposes and price range, as it offers both A-D and D-A conversion, so it would not only improve the quality of all input signals, but also increase the resolution of the digital output from the Yamaha desk that feeds your main monitors.

Further up the scale are Apogee's Rosetta 200 and RME's ADI96 Pro, and although these are a little over the price range you stated, you should still give them some consideration, as they both have word clock outputs from their high-resolution internal crystals. Clocking your entire system from the converter's word clock output would then upgrade the sound of all the converters in your soundcard and mixer — in the case of the original 0-series mixers in particular, I've heard that this can make a big difference to the sound.

Another possibility to consider is a dedicated monitor controller and a second pair of monitors to go with it. This would allow you to stay objective about your mixes by flipping between the two sets of monitors during the mixdown process. The second pair of monitors doesn't need to be as good as your main pair, and could even function as the 'grot boxes' — monitors that will give you some idea of how your material will sound on lower-quality domestic systems.

The other advantage of a monitor controller is that it would allow you to quickly check the mix in mono (still important for broadcast work), and would also allow you to quickly and easily audition external sound sources on your main monitors — perhaps the output of a CD player or television, again for referencing purposes during mixing. Towards the lower end of your price range is the Mackie Big Knob, which is very flexible, but you might also want to have a look at the SPL Model 2381 or Presonus Central Station, the former for its more 'audiophile' bias, and the latter for its built-in digital source monitoring facilities and remote controller.

 
A monitor controller such as the Mackie Big Knob, a quality A-D/D-A converter such as the RME ADI2 or an external plug-in processor such as the TC Electronic Powercore Compact or Waves APA44M would be welcome additions to just about any studio.

A final suggestion would be to look at some of the add-on DSP processors currently available, in order to increase the number and quality of plug-ins and virtual instruments that you can run using your single G5 Mac. There's lots of choice here within your price range, such as the Universal Audio UAD1 processor card bundles, several varieties of TC Electronic Powercore (both PCI and external Firewire), and the new Waves APA series. Which one you go for will depend on what plug-ins you're most likely to use, but you should see a significant increase in audio processing power in all cases. I imagine you already know how useful it is to have everything running live when the director changes the brief at the last moment!


Forum member Tomás Mulcahy also suggests exchanging the G5's audio interface for one with ADAT I/O, and then also buying the ADAT-equipped mini-YGDAI expansion board for the 03D. This would remove unnecessary extra stages of A-D and D-A conversion when transferring audio from the computer to the mixer, and would certainly improve the sound quality of the whole system. RME's Multiface and Digiface are both possibilities well within your price range, and as RME also have an excellent reputation for digital clocking you should still see some improvement in the Focusrite channels' A-D conversion and the Yamaha mixer's D-A conversion. Whether you want the analogue I/O provided on the Multiface will depend on your future expansion plans, but by the sound of things there's not that much need for it in your system, so the Digiface might be the better bet, as well as the cheaper one!

Published January 2006

KORG D1 - The stylish digital piano in slim, compact design